diff options
| -rw-r--r-- | camera/libcameraservice/Android.mk | 66 | ||||
| -rw-r--r-- | camera/libcameraservice/CameraHardwareStub.cpp | 410 | ||||
| -rw-r--r-- | camera/libcameraservice/CameraHardwareStub.h | 133 | ||||
| -rw-r--r-- | camera/libcameraservice/CameraService.cpp | 1273 | ||||
| -rw-r--r-- | camera/libcameraservice/CameraService.h | 194 | ||||
| -rw-r--r-- | camera/libcameraservice/CannedJpeg.h | 734 | ||||
| -rw-r--r-- | camera/libcameraservice/FakeCamera.cpp | 433 | ||||
| -rw-r--r-- | camera/libcameraservice/FakeCamera.h | 67 | ||||
| -rw-r--r-- | camera/tests/CameraServiceTest/Android.mk | 26 | ||||
| -rw-r--r-- | camera/tests/CameraServiceTest/CameraServiceTest.cpp | 919 | ||||
| -rw-r--r-- | cmds/surfaceflinger/Android.mk | 2 | ||||
| -rw-r--r-- | libs/audioflinger/A2dpAudioInterface.cpp | 466 | ||||
| -rw-r--r-- | libs/audioflinger/A2dpAudioInterface.h | 135 | ||||
| -rw-r--r-- | libs/audioflinger/Android.mk | 131 | ||||
| -rw-r--r-- | libs/audioflinger/AudioBufferProvider.h | 49 | ||||
| -rw-r--r-- | libs/audioflinger/AudioDumpInterface.cpp | 573 | ||||
| -rw-r--r-- | libs/audioflinger/AudioDumpInterface.h | 170 | ||||
| -rw-r--r-- | libs/audioflinger/AudioFlinger.cpp | 6078 | ||||
| -rw-r--r-- | libs/audioflinger/AudioFlinger.h | 1148 | ||||
| -rw-r--r-- | libs/audioflinger/AudioHardwareGeneric.cpp | 411 | ||||
| -rw-r--r-- | libs/audioflinger/AudioHardwareGeneric.h | 151 | ||||
| -rw-r--r-- | libs/audioflinger/AudioHardwareInterface.cpp | 182 | ||||
| -rw-r--r-- | libs/audioflinger/AudioHardwareStub.cpp | 209 | ||||
| -rw-r--r-- | libs/audioflinger/AudioHardwareStub.h | 106 | ||||
| -rw-r--r-- | libs/audioflinger/AudioMixer.cpp | 1195 | ||||
| -rw-r--r-- | libs/audioflinger/AudioMixer.h | 207 | ||||
| -rw-r--r-- | libs/audioflinger/AudioPolicyManagerBase.cpp | 1973 | ||||
| -rw-r--r-- | libs/audioflinger/AudioPolicyService.cpp | 924 | ||||
| -rw-r--r-- | libs/audioflinger/AudioPolicyService.h | 223 | ||||
| -rw-r--r-- | libs/audioflinger/AudioResampler.cpp | 595 | ||||
| -rw-r--r-- | libs/audioflinger/AudioResampler.h | 93 | ||||
| -rw-r--r-- | libs/audioflinger/AudioResamplerCubic.cpp | 184 | ||||
| -rw-r--r-- | libs/audioflinger/AudioResamplerCubic.h | 68 | ||||
| -rw-r--r-- | libs/audioflinger/AudioResamplerSinc.cpp | 358 | ||||
| -rw-r--r-- | libs/audioflinger/AudioResamplerSinc.h | 88 | ||||
| -rw-r--r-- | services/surfaceflinger/Android.mk (renamed from libs/surfaceflinger/Android.mk) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/Barrier.h (renamed from libs/surfaceflinger/Barrier.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/BlurFilter.cpp (renamed from libs/surfaceflinger/BlurFilter.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/BlurFilter.h (renamed from libs/surfaceflinger/BlurFilter.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/DisplayHardware/DisplayHardware.cpp (renamed from libs/surfaceflinger/DisplayHardware/DisplayHardware.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/DisplayHardware/DisplayHardware.h (renamed from libs/surfaceflinger/DisplayHardware/DisplayHardware.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/DisplayHardware/DisplayHardwareBase.cpp (renamed from libs/surfaceflinger/DisplayHardware/DisplayHardwareBase.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/DisplayHardware/DisplayHardwareBase.h (renamed from libs/surfaceflinger/DisplayHardware/DisplayHardwareBase.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/GLExtensions.cpp (renamed from libs/surfaceflinger/GLExtensions.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/GLExtensions.h (renamed from libs/surfaceflinger/GLExtensions.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/Layer.cpp (renamed from libs/surfaceflinger/Layer.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/Layer.h (renamed from libs/surfaceflinger/Layer.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/LayerBase.cpp (renamed from libs/surfaceflinger/LayerBase.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/LayerBase.h (renamed from libs/surfaceflinger/LayerBase.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/LayerBlur.cpp (renamed from libs/surfaceflinger/LayerBlur.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/LayerBlur.h (renamed from libs/surfaceflinger/LayerBlur.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/LayerBuffer.cpp (renamed from libs/surfaceflinger/LayerBuffer.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/LayerBuffer.h (renamed from libs/surfaceflinger/LayerBuffer.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/LayerDim.cpp (renamed from libs/surfaceflinger/LayerDim.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/LayerDim.h (renamed from libs/surfaceflinger/LayerDim.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/MODULE_LICENSE_APACHE2 (renamed from libs/surfaceflinger/MODULE_LICENSE_APACHE2) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/MessageQueue.cpp (renamed from libs/surfaceflinger/MessageQueue.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/MessageQueue.h (renamed from libs/surfaceflinger/MessageQueue.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/SurfaceFlinger.cpp (renamed from libs/surfaceflinger/SurfaceFlinger.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/SurfaceFlinger.h (renamed from libs/surfaceflinger/SurfaceFlinger.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/TextureManager.cpp (renamed from libs/surfaceflinger/TextureManager.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/TextureManager.h (renamed from libs/surfaceflinger/TextureManager.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/Transform.cpp (renamed from libs/surfaceflinger/Transform.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/Transform.h (renamed from libs/surfaceflinger/Transform.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/clz.cpp (renamed from libs/surfaceflinger/clz.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/clz.h (renamed from libs/surfaceflinger/clz.h) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/tests/Android.mk (renamed from libs/surfaceflinger/tests/Android.mk) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/tests/overlays/Android.mk (renamed from libs/surfaceflinger/tests/overlays/Android.mk) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/tests/overlays/overlays.cpp (renamed from libs/surfaceflinger/tests/overlays/overlays.cpp) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/tests/resize/Android.mk (renamed from libs/surfaceflinger/tests/resize/Android.mk) | 0 | ||||
| -rw-r--r-- | services/surfaceflinger/tests/resize/resize.cpp (renamed from libs/surfaceflinger/tests/resize/resize.cpp) | 0 |
71 files changed, 1 insertions, 19973 deletions
diff --git a/camera/libcameraservice/Android.mk b/camera/libcameraservice/Android.mk deleted file mode 100644 index 87975af..0000000 --- a/camera/libcameraservice/Android.mk +++ /dev/null @@ -1,66 +0,0 @@ -LOCAL_PATH:= $(call my-dir) - -# Set USE_CAMERA_STUB if you don't want to use the hardware camera. - -# force these builds to use camera stub only -ifneq ($(filter sooner generic sim,$(TARGET_DEVICE)),) - USE_CAMERA_STUB:=true -endif - -ifeq ($(USE_CAMERA_STUB),) - USE_CAMERA_STUB:=false -endif - -ifeq ($(USE_CAMERA_STUB),true) -# -# libcamerastub -# - -include $(CLEAR_VARS) - -LOCAL_SRC_FILES:= \ - CameraHardwareStub.cpp \ - FakeCamera.cpp - -LOCAL_MODULE:= libcamerastub - -ifeq ($(TARGET_SIMULATOR),true) -LOCAL_CFLAGS += -DSINGLE_PROCESS -endif - -LOCAL_SHARED_LIBRARIES:= libui - -include $(BUILD_STATIC_LIBRARY) -endif # USE_CAMERA_STUB - -# -# libcameraservice -# - -include $(CLEAR_VARS) - -LOCAL_SRC_FILES:= \ - CameraService.cpp - -LOCAL_SHARED_LIBRARIES:= \ - libui \ - libutils \ - libbinder \ - libcutils \ - libmedia \ - libcamera_client \ - libsurfaceflinger_client - -LOCAL_MODULE:= libcameraservice - -ifeq ($(TARGET_SIMULATOR),true) -LOCAL_CFLAGS += -DSINGLE_PROCESS -endif - -ifeq ($(USE_CAMERA_STUB), true) -LOCAL_STATIC_LIBRARIES += libcamerastub -else -LOCAL_SHARED_LIBRARIES += libcamera -endif - -include $(BUILD_SHARED_LIBRARY) diff --git a/camera/libcameraservice/CameraHardwareStub.cpp b/camera/libcameraservice/CameraHardwareStub.cpp deleted file mode 100644 index b3e0ee6..0000000 --- a/camera/libcameraservice/CameraHardwareStub.cpp +++ /dev/null @@ -1,410 +0,0 @@ -/* -** -** Copyright 2008, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#define LOG_TAG "CameraHardwareStub" -#include <utils/Log.h> - -#include "CameraHardwareStub.h" -#include <utils/threads.h> -#include <fcntl.h> -#include <sys/mman.h> - -#include "CannedJpeg.h" - -namespace android { - -CameraHardwareStub::CameraHardwareStub() - : mParameters(), - mPreviewHeap(0), - mRawHeap(0), - mFakeCamera(0), - mPreviewFrameSize(0), - mNotifyCb(0), - mDataCb(0), - mDataCbTimestamp(0), - mCallbackCookie(0), - mMsgEnabled(0), - mCurrentPreviewFrame(0) -{ - initDefaultParameters(); -} - -void CameraHardwareStub::initDefaultParameters() -{ - CameraParameters p; - - p.set(CameraParameters::KEY_SUPPORTED_PREVIEW_SIZES, "320x240"); - p.setPreviewSize(320, 240); - p.setPreviewFrameRate(15); - p.setPreviewFormat(CameraParameters::PIXEL_FORMAT_YUV420SP); - - p.set(CameraParameters::KEY_SUPPORTED_PICTURE_SIZES, "320x240"); - p.setPictureSize(320, 240); - p.setPictureFormat(CameraParameters::PIXEL_FORMAT_JPEG); - - if (setParameters(p) != NO_ERROR) { - LOGE("Failed to set default parameters?!"); - } -} - -void CameraHardwareStub::initHeapLocked() -{ - // Create raw heap. - int picture_width, picture_height; - mParameters.getPictureSize(&picture_width, &picture_height); - mRawHeap = new MemoryHeapBase(picture_width * picture_height * 3 / 2); - - int preview_width, preview_height; - mParameters.getPreviewSize(&preview_width, &preview_height); - LOGD("initHeapLocked: preview size=%dx%d", preview_width, preview_height); - - // Note that we enforce yuv420sp in setParameters(). - int how_big = preview_width * preview_height * 3 / 2; - - // If we are being reinitialized to the same size as before, no - // work needs to be done. - if (how_big == mPreviewFrameSize) - return; - - mPreviewFrameSize = how_big; - - // Make a new mmap'ed heap that can be shared across processes. - // use code below to test with pmem - mPreviewHeap = new MemoryHeapBase(mPreviewFrameSize * kBufferCount); - // Make an IMemory for each frame so that we can reuse them in callbacks. - for (int i = 0; i < kBufferCount; i++) { - mBuffers[i] = new MemoryBase(mPreviewHeap, i * mPreviewFrameSize, mPreviewFrameSize); - } - - // Recreate the fake camera to reflect the current size. - delete mFakeCamera; - mFakeCamera = new FakeCamera(preview_width, preview_height); -} - -CameraHardwareStub::~CameraHardwareStub() -{ - delete mFakeCamera; - mFakeCamera = 0; // paranoia -} - -sp<IMemoryHeap> CameraHardwareStub::getPreviewHeap() const -{ - return mPreviewHeap; -} - -sp<IMemoryHeap> CameraHardwareStub::getRawHeap() const -{ - return mRawHeap; -} - -void CameraHardwareStub::setCallbacks(notify_callback notify_cb, - data_callback data_cb, - data_callback_timestamp data_cb_timestamp, - void* user) -{ - Mutex::Autolock lock(mLock); - mNotifyCb = notify_cb; - mDataCb = data_cb; - mDataCbTimestamp = data_cb_timestamp; - mCallbackCookie = user; -} - -void CameraHardwareStub::enableMsgType(int32_t msgType) -{ - Mutex::Autolock lock(mLock); - mMsgEnabled |= msgType; -} - -void CameraHardwareStub::disableMsgType(int32_t msgType) -{ - Mutex::Autolock lock(mLock); - mMsgEnabled &= ~msgType; -} - -bool CameraHardwareStub::msgTypeEnabled(int32_t msgType) -{ - Mutex::Autolock lock(mLock); - return (mMsgEnabled & msgType); -} - -// --------------------------------------------------------------------------- - -int CameraHardwareStub::previewThread() -{ - mLock.lock(); - // the attributes below can change under our feet... - - int previewFrameRate = mParameters.getPreviewFrameRate(); - - // Find the offset within the heap of the current buffer. - ssize_t offset = mCurrentPreviewFrame * mPreviewFrameSize; - - sp<MemoryHeapBase> heap = mPreviewHeap; - - // this assumes the internal state of fake camera doesn't change - // (or is thread safe) - FakeCamera* fakeCamera = mFakeCamera; - - sp<MemoryBase> buffer = mBuffers[mCurrentPreviewFrame]; - - mLock.unlock(); - - // TODO: here check all the conditions that could go wrong - if (buffer != 0) { - // Calculate how long to wait between frames. - int delay = (int)(1000000.0f / float(previewFrameRate)); - - // This is always valid, even if the client died -- the memory - // is still mapped in our process. - void *base = heap->base(); - - // Fill the current frame with the fake camera. - uint8_t *frame = ((uint8_t *)base) + offset; - fakeCamera->getNextFrameAsYuv420(frame); - - //LOGV("previewThread: generated frame to buffer %d", mCurrentPreviewFrame); - - // Notify the client of a new frame. - if (mMsgEnabled & CAMERA_MSG_PREVIEW_FRAME) - mDataCb(CAMERA_MSG_PREVIEW_FRAME, buffer, mCallbackCookie); - - // Advance the buffer pointer. - mCurrentPreviewFrame = (mCurrentPreviewFrame + 1) % kBufferCount; - - // Wait for it... - usleep(delay); - } - - return NO_ERROR; -} - -status_t CameraHardwareStub::startPreview() -{ - Mutex::Autolock lock(mLock); - if (mPreviewThread != 0) { - // already running - return INVALID_OPERATION; - } - mPreviewThread = new PreviewThread(this); - return NO_ERROR; -} - -void CameraHardwareStub::stopPreview() -{ - sp<PreviewThread> previewThread; - - { // scope for the lock - Mutex::Autolock lock(mLock); - previewThread = mPreviewThread; - } - - // don't hold the lock while waiting for the thread to quit - if (previewThread != 0) { - previewThread->requestExitAndWait(); - } - - Mutex::Autolock lock(mLock); - mPreviewThread.clear(); -} - -bool CameraHardwareStub::previewEnabled() { - return mPreviewThread != 0; -} - -status_t CameraHardwareStub::startRecording() -{ - return UNKNOWN_ERROR; -} - -void CameraHardwareStub::stopRecording() -{ -} - -bool CameraHardwareStub::recordingEnabled() -{ - return false; -} - -void CameraHardwareStub::releaseRecordingFrame(const sp<IMemory>& mem) -{ -} - -// --------------------------------------------------------------------------- - -int CameraHardwareStub::beginAutoFocusThread(void *cookie) -{ - CameraHardwareStub *c = (CameraHardwareStub *)cookie; - return c->autoFocusThread(); -} - -int CameraHardwareStub::autoFocusThread() -{ - if (mMsgEnabled & CAMERA_MSG_FOCUS) - mNotifyCb(CAMERA_MSG_FOCUS, true, 0, mCallbackCookie); - return NO_ERROR; -} - -status_t CameraHardwareStub::autoFocus() -{ - Mutex::Autolock lock(mLock); - if (createThread(beginAutoFocusThread, this) == false) - return UNKNOWN_ERROR; - return NO_ERROR; -} - -status_t CameraHardwareStub::cancelAutoFocus() -{ - return NO_ERROR; -} - -/*static*/ int CameraHardwareStub::beginPictureThread(void *cookie) -{ - CameraHardwareStub *c = (CameraHardwareStub *)cookie; - return c->pictureThread(); -} - -int CameraHardwareStub::pictureThread() -{ - if (mMsgEnabled & CAMERA_MSG_SHUTTER) - mNotifyCb(CAMERA_MSG_SHUTTER, 0, 0, mCallbackCookie); - - if (mMsgEnabled & CAMERA_MSG_RAW_IMAGE) { - //FIXME: use a canned YUV image! - // In the meantime just make another fake camera picture. - int w, h; - mParameters.getPictureSize(&w, &h); - sp<MemoryBase> mem = new MemoryBase(mRawHeap, 0, w * h * 3 / 2); - FakeCamera cam(w, h); - cam.getNextFrameAsYuv420((uint8_t *)mRawHeap->base()); - mDataCb(CAMERA_MSG_RAW_IMAGE, mem, mCallbackCookie); - } - - if (mMsgEnabled & CAMERA_MSG_COMPRESSED_IMAGE) { - sp<MemoryHeapBase> heap = new MemoryHeapBase(kCannedJpegSize); - sp<MemoryBase> mem = new MemoryBase(heap, 0, kCannedJpegSize); - memcpy(heap->base(), kCannedJpeg, kCannedJpegSize); - mDataCb(CAMERA_MSG_COMPRESSED_IMAGE, mem, mCallbackCookie); - } - return NO_ERROR; -} - -status_t CameraHardwareStub::takePicture() -{ - stopPreview(); - if (createThread(beginPictureThread, this) == false) - return UNKNOWN_ERROR; - return NO_ERROR; -} - -status_t CameraHardwareStub::cancelPicture() -{ - return NO_ERROR; -} - -status_t CameraHardwareStub::dump(int fd, const Vector<String16>& args) const -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - AutoMutex lock(&mLock); - if (mFakeCamera != 0) { - mFakeCamera->dump(fd); - mParameters.dump(fd, args); - snprintf(buffer, 255, " preview frame(%d), size (%d), running(%s)\n", mCurrentPreviewFrame, mPreviewFrameSize, mPreviewRunning?"true": "false"); - result.append(buffer); - } else { - result.append("No camera client yet.\n"); - } - write(fd, result.string(), result.size()); - return NO_ERROR; -} - -status_t CameraHardwareStub::setParameters(const CameraParameters& params) -{ - Mutex::Autolock lock(mLock); - // XXX verify params - - if (strcmp(params.getPreviewFormat(), - CameraParameters::PIXEL_FORMAT_YUV420SP) != 0) { - LOGE("Only yuv420sp preview is supported"); - return -1; - } - - if (strcmp(params.getPictureFormat(), - CameraParameters::PIXEL_FORMAT_JPEG) != 0) { - LOGE("Only jpeg still pictures are supported"); - return -1; - } - - int w, h; - params.getPictureSize(&w, &h); - if (w != kCannedJpegWidth && h != kCannedJpegHeight) { - LOGE("Still picture size must be size of canned JPEG (%dx%d)", - kCannedJpegWidth, kCannedJpegHeight); - return -1; - } - - mParameters = params; - initHeapLocked(); - - return NO_ERROR; -} - -CameraParameters CameraHardwareStub::getParameters() const -{ - Mutex::Autolock lock(mLock); - return mParameters; -} - -status_t CameraHardwareStub::sendCommand(int32_t command, int32_t arg1, - int32_t arg2) -{ - return BAD_VALUE; -} - -void CameraHardwareStub::release() -{ -} - -sp<CameraHardwareInterface> CameraHardwareStub::createInstance() -{ - return new CameraHardwareStub(); -} - -static CameraInfo sCameraInfo[] = { - { - CAMERA_FACING_BACK, - 90, /* orientation */ - } -}; - -extern "C" int HAL_getNumberOfCameras() -{ - return sizeof(sCameraInfo) / sizeof(sCameraInfo[0]); -} - -extern "C" void HAL_getCameraInfo(int cameraId, struct CameraInfo* cameraInfo) -{ - memcpy(cameraInfo, &sCameraInfo[cameraId], sizeof(CameraInfo)); -} - -extern "C" sp<CameraHardwareInterface> HAL_openCameraHardware(int cameraId) -{ - return CameraHardwareStub::createInstance(); -} - -}; // namespace android diff --git a/camera/libcameraservice/CameraHardwareStub.h b/camera/libcameraservice/CameraHardwareStub.h deleted file mode 100644 index d3427ba..0000000 --- a/camera/libcameraservice/CameraHardwareStub.h +++ /dev/null @@ -1,133 +0,0 @@ -/* -** -** Copyright 2008, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#ifndef ANDROID_HARDWARE_CAMERA_HARDWARE_STUB_H -#define ANDROID_HARDWARE_CAMERA_HARDWARE_STUB_H - -#include "FakeCamera.h" -#include <utils/threads.h> -#include <camera/CameraHardwareInterface.h> -#include <binder/MemoryBase.h> -#include <binder/MemoryHeapBase.h> -#include <utils/threads.h> - -namespace android { - -class CameraHardwareStub : public CameraHardwareInterface { -public: - virtual sp<IMemoryHeap> getPreviewHeap() const; - virtual sp<IMemoryHeap> getRawHeap() const; - - virtual void setCallbacks(notify_callback notify_cb, - data_callback data_cb, - data_callback_timestamp data_cb_timestamp, - void* user); - - virtual void enableMsgType(int32_t msgType); - virtual void disableMsgType(int32_t msgType); - virtual bool msgTypeEnabled(int32_t msgType); - - virtual status_t startPreview(); - virtual void stopPreview(); - virtual bool previewEnabled(); - - virtual status_t startRecording(); - virtual void stopRecording(); - virtual bool recordingEnabled(); - virtual void releaseRecordingFrame(const sp<IMemory>& mem); - - virtual status_t autoFocus(); - virtual status_t cancelAutoFocus(); - virtual status_t takePicture(); - virtual status_t cancelPicture(); - virtual status_t dump(int fd, const Vector<String16>& args) const; - virtual status_t setParameters(const CameraParameters& params); - virtual CameraParameters getParameters() const; - virtual status_t sendCommand(int32_t command, int32_t arg1, - int32_t arg2); - virtual void release(); - - static sp<CameraHardwareInterface> createInstance(); - -private: - CameraHardwareStub(); - virtual ~CameraHardwareStub(); - - static const int kBufferCount = 4; - - class PreviewThread : public Thread { - CameraHardwareStub* mHardware; - public: - PreviewThread(CameraHardwareStub* hw) : -#ifdef SINGLE_PROCESS - // In single process mode this thread needs to be a java thread, - // since we won't be calling through the binder. - Thread(true), -#else - Thread(false), -#endif - mHardware(hw) { } - virtual void onFirstRef() { - run("CameraPreviewThread", PRIORITY_URGENT_DISPLAY); - } - virtual bool threadLoop() { - mHardware->previewThread(); - // loop until we need to quit - return true; - } - }; - - void initDefaultParameters(); - void initHeapLocked(); - - int previewThread(); - - static int beginAutoFocusThread(void *cookie); - int autoFocusThread(); - - static int beginPictureThread(void *cookie); - int pictureThread(); - - mutable Mutex mLock; - - CameraParameters mParameters; - - sp<MemoryHeapBase> mPreviewHeap; - sp<MemoryHeapBase> mRawHeap; - sp<MemoryBase> mBuffers[kBufferCount]; - - FakeCamera *mFakeCamera; - bool mPreviewRunning; - int mPreviewFrameSize; - - // protected by mLock - sp<PreviewThread> mPreviewThread; - - notify_callback mNotifyCb; - data_callback mDataCb; - data_callback_timestamp mDataCbTimestamp; - void *mCallbackCookie; - - int32_t mMsgEnabled; - - // only used from PreviewThread - int mCurrentPreviewFrame; -}; - -}; // namespace android - -#endif diff --git a/camera/libcameraservice/CameraService.cpp b/camera/libcameraservice/CameraService.cpp deleted file mode 100644 index 10668a4..0000000 --- a/camera/libcameraservice/CameraService.cpp +++ /dev/null @@ -1,1273 +0,0 @@ -/* -** -** Copyright (C) 2008, The Android Open Source Project -** Copyright (C) 2008 HTC Inc. -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#define LOG_TAG "CameraService" - -#include <stdio.h> -#include <sys/types.h> -#include <pthread.h> - -#include <binder/IPCThreadState.h> -#include <binder/IServiceManager.h> -#include <binder/MemoryBase.h> -#include <binder/MemoryHeapBase.h> -#include <cutils/atomic.h> -#include <hardware/hardware.h> -#include <media/AudioSystem.h> -#include <media/mediaplayer.h> -#include <surfaceflinger/ISurface.h> -#include <ui/Overlay.h> -#include <utils/Errors.h> -#include <utils/Log.h> -#include <utils/String16.h> - -#include "CameraService.h" - -namespace android { - -// ---------------------------------------------------------------------------- -// Logging support -- this is for debugging only -// Use "adb shell dumpsys media.camera -v 1" to change it. -static volatile int32_t gLogLevel = 0; - -#define LOG1(...) LOGD_IF(gLogLevel >= 1, __VA_ARGS__); -#define LOG2(...) LOGD_IF(gLogLevel >= 2, __VA_ARGS__); - -static void setLogLevel(int level) { - android_atomic_write(level, &gLogLevel); -} - -// ---------------------------------------------------------------------------- - -static int getCallingPid() { - return IPCThreadState::self()->getCallingPid(); -} - -static int getCallingUid() { - return IPCThreadState::self()->getCallingUid(); -} - -// ---------------------------------------------------------------------------- - -// This is ugly and only safe if we never re-create the CameraService, but -// should be ok for now. -static CameraService *gCameraService; - -CameraService::CameraService() -:mSoundRef(0) -{ - LOGI("CameraService started (pid=%d)", getpid()); - - mNumberOfCameras = HAL_getNumberOfCameras(); - if (mNumberOfCameras > MAX_CAMERAS) { - LOGE("Number of cameras(%d) > MAX_CAMERAS(%d).", - mNumberOfCameras, MAX_CAMERAS); - mNumberOfCameras = MAX_CAMERAS; - } - - for (int i = 0; i < mNumberOfCameras; i++) { - setCameraFree(i); - } - - gCameraService = this; -} - -CameraService::~CameraService() { - for (int i = 0; i < mNumberOfCameras; i++) { - if (mBusy[i]) { - LOGE("camera %d is still in use in destructor!", i); - } - } - - gCameraService = NULL; -} - -int32_t CameraService::getNumberOfCameras() { - return mNumberOfCameras; -} - -status_t CameraService::getCameraInfo(int cameraId, - struct CameraInfo* cameraInfo) { - if (cameraId < 0 || cameraId >= mNumberOfCameras) { - return BAD_VALUE; - } - - HAL_getCameraInfo(cameraId, cameraInfo); - return OK; -} - -sp<ICamera> CameraService::connect( - const sp<ICameraClient>& cameraClient, int cameraId) { - int callingPid = getCallingPid(); - LOG1("CameraService::connect E (pid %d, id %d)", callingPid, cameraId); - - sp<Client> client; - if (cameraId < 0 || cameraId >= mNumberOfCameras) { - LOGE("CameraService::connect X (pid %d) rejected (invalid cameraId %d).", - callingPid, cameraId); - return NULL; - } - - Mutex::Autolock lock(mServiceLock); - if (mClient[cameraId] != 0) { - client = mClient[cameraId].promote(); - if (client != 0) { - if (cameraClient->asBinder() == client->getCameraClient()->asBinder()) { - LOG1("CameraService::connect X (pid %d) (the same client)", - callingPid); - return client; - } else { - LOGW("CameraService::connect X (pid %d) rejected (existing client).", - callingPid); - return NULL; - } - } - mClient[cameraId].clear(); - } - - if (mBusy[cameraId]) { - LOGW("CameraService::connect X (pid %d) rejected" - " (camera %d is still busy).", callingPid, cameraId); - return NULL; - } - - client = new Client(this, cameraClient, cameraId, callingPid); - mClient[cameraId] = client; - LOG1("CameraService::connect X"); - return client; -} - -void CameraService::removeClient(const sp<ICameraClient>& cameraClient) { - int callingPid = getCallingPid(); - LOG1("CameraService::removeClient E (pid %d)", callingPid); - - for (int i = 0; i < mNumberOfCameras; i++) { - // Declare this before the lock to make absolutely sure the - // destructor won't be called with the lock held. - sp<Client> client; - - Mutex::Autolock lock(mServiceLock); - - // This happens when we have already disconnected (or this is - // just another unused camera). - if (mClient[i] == 0) continue; - - // Promote mClient. It can fail if we are called from this path: - // Client::~Client() -> disconnect() -> removeClient(). - client = mClient[i].promote(); - - if (client == 0) { - mClient[i].clear(); - continue; - } - - if (cameraClient->asBinder() == client->getCameraClient()->asBinder()) { - // Found our camera, clear and leave. - LOG1("removeClient: clear camera %d", i); - mClient[i].clear(); - break; - } - } - - LOG1("CameraService::removeClient X (pid %d)", callingPid); -} - -sp<CameraService::Client> CameraService::getClientById(int cameraId) { - if (cameraId < 0 || cameraId >= mNumberOfCameras) return NULL; - return mClient[cameraId].promote(); -} - -void CameraService::instantiate() { - defaultServiceManager()->addService(String16("media.camera"), - new CameraService()); -} - -status_t CameraService::onTransact( - uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { - // Permission checks - switch (code) { - case BnCameraService::CONNECT: - const int pid = getCallingPid(); - const int self_pid = getpid(); - if (pid != self_pid) { - // we're called from a different process, do the real check - if (!checkCallingPermission( - String16("android.permission.CAMERA"))) { - const int uid = getCallingUid(); - LOGE("Permission Denial: " - "can't use the camera pid=%d, uid=%d", pid, uid); - return PERMISSION_DENIED; - } - } - break; - } - - return BnCameraService::onTransact(code, data, reply, flags); -} - -// The reason we need this busy bit is a new CameraService::connect() request -// may come in while the previous Client's destructor has not been run or is -// still running. If the last strong reference of the previous Client is gone -// but the destructor has not been finished, we should not allow the new Client -// to be created because we need to wait for the previous Client to tear down -// the hardware first. -void CameraService::setCameraBusy(int cameraId) { - android_atomic_write(1, &mBusy[cameraId]); -} - -void CameraService::setCameraFree(int cameraId) { - android_atomic_write(0, &mBusy[cameraId]); -} - -// We share the media players for shutter and recording sound for all clients. -// A reference count is kept to determine when we will actually release the -// media players. - -static MediaPlayer* newMediaPlayer(const char *file) { - MediaPlayer* mp = new MediaPlayer(); - if (mp->setDataSource(file, NULL) == NO_ERROR) { - mp->setAudioStreamType(AudioSystem::ENFORCED_AUDIBLE); - mp->prepare(); - } else { - LOGE("Failed to load CameraService sounds: %s", file); - return NULL; - } - return mp; -} - -void CameraService::loadSound() { - Mutex::Autolock lock(mSoundLock); - LOG1("CameraService::loadSound ref=%d", mSoundRef); - if (mSoundRef++) return; - - mSoundPlayer[SOUND_SHUTTER] = newMediaPlayer("/system/media/audio/ui/camera_click.ogg"); - mSoundPlayer[SOUND_RECORDING] = newMediaPlayer("/system/media/audio/ui/VideoRecord.ogg"); -} - -void CameraService::releaseSound() { - Mutex::Autolock lock(mSoundLock); - LOG1("CameraService::releaseSound ref=%d", mSoundRef); - if (--mSoundRef) return; - - for (int i = 0; i < NUM_SOUNDS; i++) { - if (mSoundPlayer[i] != 0) { - mSoundPlayer[i]->disconnect(); - mSoundPlayer[i].clear(); - } - } -} - -void CameraService::playSound(sound_kind kind) { - LOG1("playSound(%d)", kind); - Mutex::Autolock lock(mSoundLock); - sp<MediaPlayer> player = mSoundPlayer[kind]; - if (player != 0) { - // do not play the sound if stream volume is 0 - // (typically because ringer mode is silent). - int index; - AudioSystem::getStreamVolumeIndex(AudioSystem::ENFORCED_AUDIBLE, &index); - if (index != 0) { - player->seekTo(0); - player->start(); - } - } -} - -// ---------------------------------------------------------------------------- - -CameraService::Client::Client(const sp<CameraService>& cameraService, - const sp<ICameraClient>& cameraClient, int cameraId, int clientPid) { - int callingPid = getCallingPid(); - LOG1("Client::Client E (pid %d)", callingPid); - - mCameraService = cameraService; - mCameraClient = cameraClient; - mCameraId = cameraId; - mClientPid = clientPid; - - mHardware = HAL_openCameraHardware(cameraId); - mUseOverlay = mHardware->useOverlay(); - mMsgEnabled = 0; - - mHardware->setCallbacks(notifyCallback, - dataCallback, - dataCallbackTimestamp, - (void *)cameraId); - - // Enable zoom, error, and focus messages by default - enableMsgType(CAMERA_MSG_ERROR | - CAMERA_MSG_ZOOM | - CAMERA_MSG_FOCUS); - mOverlayW = 0; - mOverlayH = 0; - - // Callback is disabled by default - mPreviewCallbackFlag = FRAME_CALLBACK_FLAG_NOOP; - mOrientation = 0; - cameraService->setCameraBusy(cameraId); - cameraService->loadSound(); - LOG1("Client::Client X (pid %d)", callingPid); -} - -static void *unregister_surface(void *arg) { - ISurface *surface = (ISurface *)arg; - surface->unregisterBuffers(); - IPCThreadState::self()->flushCommands(); - return NULL; -} - -// tear down the client -CameraService::Client::~Client() { - int callingPid = getCallingPid(); - LOG1("Client::~Client E (pid %d, this %p)", callingPid, this); - - if (mSurface != 0 && !mUseOverlay) { - pthread_t thr; - // We unregister the buffers in a different thread because binder does - // not let us make sychronous transactions in a binder destructor (that - // is, upon our reaching a refcount of zero.) - pthread_create(&thr, - NULL, // attr - unregister_surface, - mSurface.get()); - pthread_join(thr, NULL); - } - - // set mClientPid to let disconnet() tear down the hardware - mClientPid = callingPid; - disconnect(); - mCameraService->releaseSound(); - LOG1("Client::~Client X (pid %d, this %p)", callingPid, this); -} - -// ---------------------------------------------------------------------------- - -status_t CameraService::Client::checkPid() const { - int callingPid = getCallingPid(); - if (callingPid == mClientPid) return NO_ERROR; - - LOGW("attempt to use a locked camera from a different process" - " (old pid %d, new pid %d)", mClientPid, callingPid); - return EBUSY; -} - -status_t CameraService::Client::checkPidAndHardware() const { - status_t result = checkPid(); - if (result != NO_ERROR) return result; - if (mHardware == 0) { - LOGE("attempt to use a camera after disconnect() (pid %d)", getCallingPid()); - return INVALID_OPERATION; - } - return NO_ERROR; -} - -status_t CameraService::Client::lock() { - int callingPid = getCallingPid(); - LOG1("lock (pid %d)", callingPid); - Mutex::Autolock lock(mLock); - - // lock camera to this client if the the camera is unlocked - if (mClientPid == 0) { - mClientPid = callingPid; - return NO_ERROR; - } - - // returns NO_ERROR if the client already owns the camera, EBUSY otherwise - return checkPid(); -} - -status_t CameraService::Client::unlock() { - int callingPid = getCallingPid(); - LOG1("unlock (pid %d)", callingPid); - Mutex::Autolock lock(mLock); - - // allow anyone to use camera (after they lock the camera) - status_t result = checkPid(); - if (result == NO_ERROR) { - mClientPid = 0; - LOG1("clear mCameraClient (pid %d)", callingPid); - // we need to remove the reference to ICameraClient so that when the app - // goes away, the reference count goes to 0. - mCameraClient.clear(); - } - return result; -} - -// connect a new client to the camera -status_t CameraService::Client::connect(const sp<ICameraClient>& client) { - int callingPid = getCallingPid(); - LOG1("connect E (pid %d)", callingPid); - Mutex::Autolock lock(mLock); - - if (mClientPid != 0 && checkPid() != NO_ERROR) { - LOGW("Tried to connect to a locked camera (old pid %d, new pid %d)", - mClientPid, callingPid); - return EBUSY; - } - - if (mCameraClient != 0 && (client->asBinder() == mCameraClient->asBinder())) { - LOG1("Connect to the same client"); - return NO_ERROR; - } - - mPreviewCallbackFlag = FRAME_CALLBACK_FLAG_NOOP; - mClientPid = callingPid; - mCameraClient = client; - - LOG1("connect X (pid %d)", callingPid); - return NO_ERROR; -} - -void CameraService::Client::disconnect() { - int callingPid = getCallingPid(); - LOG1("disconnect E (pid %d)", callingPid); - Mutex::Autolock lock(mLock); - - if (checkPid() != NO_ERROR) { - LOGW("different client - don't disconnect"); - return; - } - - if (mClientPid <= 0) { - LOG1("camera is unlocked (mClientPid = %d), don't tear down hardware", mClientPid); - return; - } - - // Make sure disconnect() is done once and once only, whether it is called - // from the user directly, or called by the destructor. - if (mHardware == 0) return; - - LOG1("hardware teardown"); - // Before destroying mHardware, we must make sure it's in the - // idle state. - // Turn off all messages. - disableMsgType(CAMERA_MSG_ALL_MSGS); - mHardware->stopPreview(); - mHardware->cancelPicture(); - // Release the hardware resources. - mHardware->release(); - // Release the held overlay resources. - if (mUseOverlay) { - mOverlayRef = 0; - } - mHardware.clear(); - - mCameraService->removeClient(mCameraClient); - mCameraService->setCameraFree(mCameraId); - - LOG1("disconnect X (pid %d)", callingPid); -} - -// ---------------------------------------------------------------------------- - -// set the ISurface that the preview will use -status_t CameraService::Client::setPreviewDisplay(const sp<ISurface>& surface) { - LOG1("setPreviewDisplay(%p) (pid %d)", surface.get(), getCallingPid()); - Mutex::Autolock lock(mLock); - status_t result = checkPidAndHardware(); - if (result != NO_ERROR) return result; - - result = NO_ERROR; - - // return if no change in surface. - // asBinder() is safe on NULL (returns NULL) - if (surface->asBinder() == mSurface->asBinder()) { - return result; - } - - if (mSurface != 0) { - LOG1("clearing old preview surface %p", mSurface.get()); - if (mUseOverlay) { - // Force the destruction of any previous overlay - sp<Overlay> dummy; - mHardware->setOverlay(dummy); - } else { - mSurface->unregisterBuffers(); - } - } - mSurface = surface; - mOverlayRef = 0; - // If preview has been already started, set overlay or register preview - // buffers now. - if (mHardware->previewEnabled()) { - if (mUseOverlay) { - result = setOverlay(); - } else if (mSurface != 0) { - result = registerPreviewBuffers(); - } - } - - return result; -} - -status_t CameraService::Client::registerPreviewBuffers() { - int w, h; - CameraParameters params(mHardware->getParameters()); - params.getPreviewSize(&w, &h); - - // FIXME: don't use a hardcoded format here. - ISurface::BufferHeap buffers(w, h, w, h, - HAL_PIXEL_FORMAT_YCrCb_420_SP, - mOrientation, - 0, - mHardware->getPreviewHeap()); - - status_t result = mSurface->registerBuffers(buffers); - if (result != NO_ERROR) { - LOGE("registerBuffers failed with status %d", result); - } - return result; -} - -status_t CameraService::Client::setOverlay() { - int w, h; - CameraParameters params(mHardware->getParameters()); - params.getPreviewSize(&w, &h); - - if (w != mOverlayW || h != mOverlayH) { - // Force the destruction of any previous overlay - sp<Overlay> dummy; - mHardware->setOverlay(dummy); - mOverlayRef = 0; - } - - status_t result = NO_ERROR; - if (mSurface == 0) { - result = mHardware->setOverlay(NULL); - } else { - if (mOverlayRef == 0) { - // FIXME: - // Surfaceflinger may hold onto the previous overlay reference for some - // time after we try to destroy it. retry a few times. In the future, we - // should make the destroy call block, or possibly specify that we can - // wait in the createOverlay call if the previous overlay is in the - // process of being destroyed. - for (int retry = 0; retry < 50; ++retry) { - mOverlayRef = mSurface->createOverlay(w, h, OVERLAY_FORMAT_DEFAULT, - mOrientation); - if (mOverlayRef != 0) break; - LOGW("Overlay create failed - retrying"); - usleep(20000); - } - if (mOverlayRef == 0) { - LOGE("Overlay Creation Failed!"); - return -EINVAL; - } - result = mHardware->setOverlay(new Overlay(mOverlayRef)); - } - } - if (result != NO_ERROR) { - LOGE("mHardware->setOverlay() failed with status %d\n", result); - return result; - } - - mOverlayW = w; - mOverlayH = h; - - return result; -} - -// set the preview callback flag to affect how the received frames from -// preview are handled. -void CameraService::Client::setPreviewCallbackFlag(int callback_flag) { - LOG1("setPreviewCallbackFlag(%d) (pid %d)", callback_flag, getCallingPid()); - Mutex::Autolock lock(mLock); - if (checkPidAndHardware() != NO_ERROR) return; - - mPreviewCallbackFlag = callback_flag; - - // If we don't use overlay, we always need the preview frame for display. - // If we do use overlay, we only need the preview frame if the user - // wants the data. - if (mUseOverlay) { - if(mPreviewCallbackFlag & FRAME_CALLBACK_FLAG_ENABLE_MASK) { - enableMsgType(CAMERA_MSG_PREVIEW_FRAME); - } else { - disableMsgType(CAMERA_MSG_PREVIEW_FRAME); - } - } -} - -// start preview mode -status_t CameraService::Client::startPreview() { - LOG1("startPreview (pid %d)", getCallingPid()); - return startCameraMode(CAMERA_PREVIEW_MODE); -} - -// start recording mode -status_t CameraService::Client::startRecording() { - LOG1("startRecording (pid %d)", getCallingPid()); - return startCameraMode(CAMERA_RECORDING_MODE); -} - -// start preview or recording -status_t CameraService::Client::startCameraMode(camera_mode mode) { - LOG1("startCameraMode(%d)", mode); - Mutex::Autolock lock(mLock); - status_t result = checkPidAndHardware(); - if (result != NO_ERROR) return result; - - switch(mode) { - case CAMERA_PREVIEW_MODE: - if (mSurface == 0) { - LOG1("mSurface is not set yet."); - // still able to start preview in this case. - } - return startPreviewMode(); - case CAMERA_RECORDING_MODE: - if (mSurface == 0) { - LOGE("mSurface must be set before startRecordingMode."); - return INVALID_OPERATION; - } - return startRecordingMode(); - default: - return UNKNOWN_ERROR; - } -} - -status_t CameraService::Client::startPreviewMode() { - LOG1("startPreviewMode"); - status_t result = NO_ERROR; - - // if preview has been enabled, nothing needs to be done - if (mHardware->previewEnabled()) { - return NO_ERROR; - } - - if (mUseOverlay) { - // If preview display has been set, set overlay now. - if (mSurface != 0) { - result = setOverlay(); - } - if (result != NO_ERROR) return result; - result = mHardware->startPreview(); - } else { - enableMsgType(CAMERA_MSG_PREVIEW_FRAME); - result = mHardware->startPreview(); - if (result != NO_ERROR) return result; - // If preview display has been set, register preview buffers now. - if (mSurface != 0) { - // Unregister here because the surface may be previously registered - // with the raw (snapshot) heap. - mSurface->unregisterBuffers(); - result = registerPreviewBuffers(); - } - } - return result; -} - -status_t CameraService::Client::startRecordingMode() { - LOG1("startRecordingMode"); - status_t result = NO_ERROR; - - // if recording has been enabled, nothing needs to be done - if (mHardware->recordingEnabled()) { - return NO_ERROR; - } - - // if preview has not been started, start preview first - if (!mHardware->previewEnabled()) { - result = startPreviewMode(); - if (result != NO_ERROR) { - return result; - } - } - - // start recording mode - enableMsgType(CAMERA_MSG_VIDEO_FRAME); - mCameraService->playSound(SOUND_RECORDING); - result = mHardware->startRecording(); - if (result != NO_ERROR) { - LOGE("mHardware->startRecording() failed with status %d", result); - } - return result; -} - -// stop preview mode -void CameraService::Client::stopPreview() { - LOG1("stopPreview (pid %d)", getCallingPid()); - Mutex::Autolock lock(mLock); - if (checkPidAndHardware() != NO_ERROR) return; - - disableMsgType(CAMERA_MSG_PREVIEW_FRAME); - mHardware->stopPreview(); - - if (mSurface != 0 && !mUseOverlay) { - mSurface->unregisterBuffers(); - } - - mPreviewBuffer.clear(); -} - -// stop recording mode -void CameraService::Client::stopRecording() { - LOG1("stopRecording (pid %d)", getCallingPid()); - Mutex::Autolock lock(mLock); - if (checkPidAndHardware() != NO_ERROR) return; - - mCameraService->playSound(SOUND_RECORDING); - disableMsgType(CAMERA_MSG_VIDEO_FRAME); - mHardware->stopRecording(); - - mPreviewBuffer.clear(); -} - -// release a recording frame -void CameraService::Client::releaseRecordingFrame(const sp<IMemory>& mem) { - Mutex::Autolock lock(mLock); - if (checkPidAndHardware() != NO_ERROR) return; - mHardware->releaseRecordingFrame(mem); -} - -bool CameraService::Client::previewEnabled() { - LOG1("previewEnabled (pid %d)", getCallingPid()); - - Mutex::Autolock lock(mLock); - if (checkPidAndHardware() != NO_ERROR) return false; - return mHardware->previewEnabled(); -} - -bool CameraService::Client::recordingEnabled() { - LOG1("recordingEnabled (pid %d)", getCallingPid()); - - Mutex::Autolock lock(mLock); - if (checkPidAndHardware() != NO_ERROR) return false; - return mHardware->recordingEnabled(); -} - -status_t CameraService::Client::autoFocus() { - LOG1("autoFocus (pid %d)", getCallingPid()); - - Mutex::Autolock lock(mLock); - status_t result = checkPidAndHardware(); - if (result != NO_ERROR) return result; - - return mHardware->autoFocus(); -} - -status_t CameraService::Client::cancelAutoFocus() { - LOG1("cancelAutoFocus (pid %d)", getCallingPid()); - - Mutex::Autolock lock(mLock); - status_t result = checkPidAndHardware(); - if (result != NO_ERROR) return result; - - return mHardware->cancelAutoFocus(); -} - -// take a picture - image is returned in callback -status_t CameraService::Client::takePicture() { - LOG1("takePicture (pid %d)", getCallingPid()); - - Mutex::Autolock lock(mLock); - status_t result = checkPidAndHardware(); - if (result != NO_ERROR) return result; - - enableMsgType(CAMERA_MSG_SHUTTER | - CAMERA_MSG_POSTVIEW_FRAME | - CAMERA_MSG_RAW_IMAGE | - CAMERA_MSG_COMPRESSED_IMAGE); - - return mHardware->takePicture(); -} - -// set preview/capture parameters - key/value pairs -status_t CameraService::Client::setParameters(const String8& params) { - LOG1("setParameters (pid %d) (%s)", getCallingPid(), params.string()); - - Mutex::Autolock lock(mLock); - status_t result = checkPidAndHardware(); - if (result != NO_ERROR) return result; - - CameraParameters p(params); - return mHardware->setParameters(p); -} - -// get preview/capture parameters - key/value pairs -String8 CameraService::Client::getParameters() const { - Mutex::Autolock lock(mLock); - if (checkPidAndHardware() != NO_ERROR) return String8(); - - String8 params(mHardware->getParameters().flatten()); - LOG1("getParameters (pid %d) (%s)", getCallingPid(), params.string()); - return params; -} - -status_t CameraService::Client::sendCommand(int32_t cmd, int32_t arg1, int32_t arg2) { - LOG1("sendCommand (pid %d)", getCallingPid()); - Mutex::Autolock lock(mLock); - status_t result = checkPidAndHardware(); - if (result != NO_ERROR) return result; - - if (cmd == CAMERA_CMD_SET_DISPLAY_ORIENTATION) { - // The orientation cannot be set during preview. - if (mHardware->previewEnabled()) { - return INVALID_OPERATION; - } - switch (arg1) { - case 0: - mOrientation = ISurface::BufferHeap::ROT_0; - break; - case 90: - mOrientation = ISurface::BufferHeap::ROT_90; - break; - case 180: - mOrientation = ISurface::BufferHeap::ROT_180; - break; - case 270: - mOrientation = ISurface::BufferHeap::ROT_270; - break; - default: - return BAD_VALUE; - } - return OK; - } - - return mHardware->sendCommand(cmd, arg1, arg2); -} - -// ---------------------------------------------------------------------------- - -void CameraService::Client::enableMsgType(int32_t msgType) { - android_atomic_or(msgType, &mMsgEnabled); - mHardware->enableMsgType(msgType); -} - -void CameraService::Client::disableMsgType(int32_t msgType) { - android_atomic_and(~msgType, &mMsgEnabled); - mHardware->disableMsgType(msgType); -} - -#define CHECK_MESSAGE_INTERVAL 10 // 10ms -bool CameraService::Client::lockIfMessageWanted(int32_t msgType) { - int sleepCount = 0; - while (mMsgEnabled & msgType) { - if (mLock.tryLock() == NO_ERROR) { - if (sleepCount > 0) { - LOG1("lockIfMessageWanted(%d): waited for %d ms", - msgType, sleepCount * CHECK_MESSAGE_INTERVAL); - } - return true; - } - if (sleepCount++ == 0) { - LOG1("lockIfMessageWanted(%d): enter sleep", msgType); - } - usleep(CHECK_MESSAGE_INTERVAL * 1000); - } - LOGW("lockIfMessageWanted(%d): dropped unwanted message", msgType); - return false; -} - -// ---------------------------------------------------------------------------- - -// Converts from a raw pointer to the client to a strong pointer during a -// hardware callback. This requires the callbacks only happen when the client -// is still alive. -sp<CameraService::Client> CameraService::Client::getClientFromCookie(void* user) { - sp<Client> client = gCameraService->getClientById((int) user); - - // This could happen if the Client is in the process of shutting down (the - // last strong reference is gone, but the destructor hasn't finished - // stopping the hardware). - if (client == 0) return NULL; - - // The checks below are not necessary and are for debugging only. - if (client->mCameraService.get() != gCameraService) { - LOGE("mismatch service!"); - return NULL; - } - - if (client->mHardware == 0) { - LOGE("mHardware == 0: callback after disconnect()?"); - return NULL; - } - - return client; -} - -// Callback messages can be dispatched to internal handlers or pass to our -// client's callback functions, depending on the message type. -// -// notifyCallback: -// CAMERA_MSG_SHUTTER handleShutter -// (others) c->notifyCallback -// dataCallback: -// CAMERA_MSG_PREVIEW_FRAME handlePreviewData -// CAMERA_MSG_POSTVIEW_FRAME handlePostview -// CAMERA_MSG_RAW_IMAGE handleRawPicture -// CAMERA_MSG_COMPRESSED_IMAGE handleCompressedPicture -// (others) c->dataCallback -// dataCallbackTimestamp -// (others) c->dataCallbackTimestamp -// -// NOTE: the *Callback functions grab mLock of the client before passing -// control to handle* functions. So the handle* functions must release the -// lock before calling the ICameraClient's callbacks, so those callbacks can -// invoke methods in the Client class again (For example, the preview frame -// callback may want to releaseRecordingFrame). The handle* functions must -// release the lock after all accesses to member variables, so it must be -// handled very carefully. - -void CameraService::Client::notifyCallback(int32_t msgType, int32_t ext1, - int32_t ext2, void* user) { - LOG2("notifyCallback(%d)", msgType); - - sp<Client> client = getClientFromCookie(user); - if (client == 0) return; - if (!client->lockIfMessageWanted(msgType)) return; - - switch (msgType) { - case CAMERA_MSG_SHUTTER: - // ext1 is the dimension of the yuv picture. - client->handleShutter((image_rect_type *)ext1); - break; - default: - client->handleGenericNotify(msgType, ext1, ext2); - break; - } -} - -void CameraService::Client::dataCallback(int32_t msgType, - const sp<IMemory>& dataPtr, void* user) { - LOG2("dataCallback(%d)", msgType); - - sp<Client> client = getClientFromCookie(user); - if (client == 0) return; - if (!client->lockIfMessageWanted(msgType)) return; - - if (dataPtr == 0) { - LOGE("Null data returned in data callback"); - client->handleGenericNotify(CAMERA_MSG_ERROR, UNKNOWN_ERROR, 0); - return; - } - - switch (msgType) { - case CAMERA_MSG_PREVIEW_FRAME: - client->handlePreviewData(dataPtr); - break; - case CAMERA_MSG_POSTVIEW_FRAME: - client->handlePostview(dataPtr); - break; - case CAMERA_MSG_RAW_IMAGE: - client->handleRawPicture(dataPtr); - break; - case CAMERA_MSG_COMPRESSED_IMAGE: - client->handleCompressedPicture(dataPtr); - break; - default: - client->handleGenericData(msgType, dataPtr); - break; - } -} - -void CameraService::Client::dataCallbackTimestamp(nsecs_t timestamp, - int32_t msgType, const sp<IMemory>& dataPtr, void* user) { - LOG2("dataCallbackTimestamp(%d)", msgType); - - sp<Client> client = getClientFromCookie(user); - if (client == 0) return; - if (!client->lockIfMessageWanted(msgType)) return; - - if (dataPtr == 0) { - LOGE("Null data returned in data with timestamp callback"); - client->handleGenericNotify(CAMERA_MSG_ERROR, UNKNOWN_ERROR, 0); - return; - } - - client->handleGenericDataTimestamp(timestamp, msgType, dataPtr); -} - -// snapshot taken callback -// "size" is the width and height of yuv picture for registerBuffer. -// If it is NULL, use the picture size from parameters. -void CameraService::Client::handleShutter(image_rect_type *size) { - mCameraService->playSound(SOUND_SHUTTER); - - // Screen goes black after the buffer is unregistered. - if (mSurface != 0 && !mUseOverlay) { - mSurface->unregisterBuffers(); - } - - sp<ICameraClient> c = mCameraClient; - if (c != 0) { - mLock.unlock(); - c->notifyCallback(CAMERA_MSG_SHUTTER, 0, 0); - if (!lockIfMessageWanted(CAMERA_MSG_SHUTTER)) return; - } - disableMsgType(CAMERA_MSG_SHUTTER); - - // It takes some time before yuvPicture callback to be called. - // Register the buffer for raw image here to reduce latency. - if (mSurface != 0 && !mUseOverlay) { - int w, h; - CameraParameters params(mHardware->getParameters()); - if (size == NULL) { - params.getPictureSize(&w, &h); - } else { - w = size->width; - h = size->height; - w &= ~1; - h &= ~1; - LOG1("Snapshot image width=%d, height=%d", w, h); - } - // FIXME: don't use hardcoded format constants here - ISurface::BufferHeap buffers(w, h, w, h, - HAL_PIXEL_FORMAT_YCrCb_420_SP, mOrientation, 0, - mHardware->getRawHeap()); - - mSurface->registerBuffers(buffers); - IPCThreadState::self()->flushCommands(); - } - - mLock.unlock(); -} - -// preview callback - frame buffer update -void CameraService::Client::handlePreviewData(const sp<IMemory>& mem) { - ssize_t offset; - size_t size; - sp<IMemoryHeap> heap = mem->getMemory(&offset, &size); - - if (!mUseOverlay) { - if (mSurface != 0) { - mSurface->postBuffer(offset); - } - } - - // local copy of the callback flags - int flags = mPreviewCallbackFlag; - - // is callback enabled? - if (!(flags & FRAME_CALLBACK_FLAG_ENABLE_MASK)) { - // If the enable bit is off, the copy-out and one-shot bits are ignored - LOG2("frame callback is disabled"); - mLock.unlock(); - return; - } - - // hold a strong pointer to the client - sp<ICameraClient> c = mCameraClient; - - // clear callback flags if no client or one-shot mode - if (c == 0 || (mPreviewCallbackFlag & FRAME_CALLBACK_FLAG_ONE_SHOT_MASK)) { - LOG2("Disable preview callback"); - mPreviewCallbackFlag &= ~(FRAME_CALLBACK_FLAG_ONE_SHOT_MASK | - FRAME_CALLBACK_FLAG_COPY_OUT_MASK | - FRAME_CALLBACK_FLAG_ENABLE_MASK); - if (mUseOverlay) { - disableMsgType(CAMERA_MSG_PREVIEW_FRAME); - } - } - - if (c != 0) { - // Is the received frame copied out or not? - if (flags & FRAME_CALLBACK_FLAG_COPY_OUT_MASK) { - LOG2("frame is copied"); - copyFrameAndPostCopiedFrame(c, heap, offset, size); - } else { - LOG2("frame is forwarded"); - mLock.unlock(); - c->dataCallback(CAMERA_MSG_PREVIEW_FRAME, mem); - } - } else { - mLock.unlock(); - } -} - -// picture callback - postview image ready -void CameraService::Client::handlePostview(const sp<IMemory>& mem) { - disableMsgType(CAMERA_MSG_POSTVIEW_FRAME); - - sp<ICameraClient> c = mCameraClient; - mLock.unlock(); - if (c != 0) { - c->dataCallback(CAMERA_MSG_POSTVIEW_FRAME, mem); - } -} - -// picture callback - raw image ready -void CameraService::Client::handleRawPicture(const sp<IMemory>& mem) { - disableMsgType(CAMERA_MSG_RAW_IMAGE); - - ssize_t offset; - size_t size; - sp<IMemoryHeap> heap = mem->getMemory(&offset, &size); - - // Put the YUV version of the snapshot in the preview display. - if (mSurface != 0 && !mUseOverlay) { - mSurface->postBuffer(offset); - } - - sp<ICameraClient> c = mCameraClient; - mLock.unlock(); - if (c != 0) { - c->dataCallback(CAMERA_MSG_RAW_IMAGE, mem); - } -} - -// picture callback - compressed picture ready -void CameraService::Client::handleCompressedPicture(const sp<IMemory>& mem) { - disableMsgType(CAMERA_MSG_COMPRESSED_IMAGE); - - sp<ICameraClient> c = mCameraClient; - mLock.unlock(); - if (c != 0) { - c->dataCallback(CAMERA_MSG_COMPRESSED_IMAGE, mem); - } -} - - -void CameraService::Client::handleGenericNotify(int32_t msgType, - int32_t ext1, int32_t ext2) { - sp<ICameraClient> c = mCameraClient; - mLock.unlock(); - if (c != 0) { - c->notifyCallback(msgType, ext1, ext2); - } -} - -void CameraService::Client::handleGenericData(int32_t msgType, - const sp<IMemory>& dataPtr) { - sp<ICameraClient> c = mCameraClient; - mLock.unlock(); - if (c != 0) { - c->dataCallback(msgType, dataPtr); - } -} - -void CameraService::Client::handleGenericDataTimestamp(nsecs_t timestamp, - int32_t msgType, const sp<IMemory>& dataPtr) { - sp<ICameraClient> c = mCameraClient; - mLock.unlock(); - if (c != 0) { - c->dataCallbackTimestamp(timestamp, msgType, dataPtr); - } -} - -void CameraService::Client::copyFrameAndPostCopiedFrame( - const sp<ICameraClient>& client, const sp<IMemoryHeap>& heap, - size_t offset, size_t size) { - LOG2("copyFrameAndPostCopiedFrame"); - // It is necessary to copy out of pmem before sending this to - // the callback. For efficiency, reuse the same MemoryHeapBase - // provided it's big enough. Don't allocate the memory or - // perform the copy if there's no callback. - // hold the preview lock while we grab a reference to the preview buffer - sp<MemoryHeapBase> previewBuffer; - - if (mPreviewBuffer == 0) { - mPreviewBuffer = new MemoryHeapBase(size, 0, NULL); - } else if (size > mPreviewBuffer->virtualSize()) { - mPreviewBuffer.clear(); - mPreviewBuffer = new MemoryHeapBase(size, 0, NULL); - } - if (mPreviewBuffer == 0) { - LOGE("failed to allocate space for preview buffer"); - mLock.unlock(); - return; - } - previewBuffer = mPreviewBuffer; - - memcpy(previewBuffer->base(), (uint8_t *)heap->base() + offset, size); - - sp<MemoryBase> frame = new MemoryBase(previewBuffer, 0, size); - if (frame == 0) { - LOGE("failed to allocate space for frame callback"); - mLock.unlock(); - return; - } - - mLock.unlock(); - client->dataCallback(CAMERA_MSG_PREVIEW_FRAME, frame); -} - -// ---------------------------------------------------------------------------- - -static const int kDumpLockRetries = 50; -static const int kDumpLockSleep = 60000; - -static bool tryLock(Mutex& mutex) -{ - bool locked = false; - for (int i = 0; i < kDumpLockRetries; ++i) { - if (mutex.tryLock() == NO_ERROR) { - locked = true; - break; - } - usleep(kDumpLockSleep); - } - return locked; -} - -status_t CameraService::dump(int fd, const Vector<String16>& args) { - static const char* kDeadlockedString = "CameraService may be deadlocked\n"; - - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - if (checkCallingPermission(String16("android.permission.DUMP")) == false) { - snprintf(buffer, SIZE, "Permission Denial: " - "can't dump CameraService from pid=%d, uid=%d\n", - getCallingPid(), - getCallingUid()); - result.append(buffer); - write(fd, result.string(), result.size()); - } else { - bool locked = tryLock(mServiceLock); - // failed to lock - CameraService is probably deadlocked - if (!locked) { - String8 result(kDeadlockedString); - write(fd, result.string(), result.size()); - } - - bool hasClient = false; - for (int i = 0; i < mNumberOfCameras; i++) { - sp<Client> client = mClient[i].promote(); - if (client == 0) continue; - hasClient = true; - sprintf(buffer, "Client[%d] (%p) PID: %d\n", - i, - client->getCameraClient()->asBinder().get(), - client->mClientPid); - result.append(buffer); - write(fd, result.string(), result.size()); - client->mHardware->dump(fd, args); - } - if (!hasClient) { - result.append("No camera client yet.\n"); - write(fd, result.string(), result.size()); - } - - if (locked) mServiceLock.unlock(); - - // change logging level - int n = args.size(); - for (int i = 0; i + 1 < n; i++) { - if (args[i] == String16("-v")) { - String8 levelStr(args[i+1]); - int level = atoi(levelStr.string()); - sprintf(buffer, "Set Log Level to %d", level); - result.append(buffer); - setLogLevel(level); - } - } - } - return NO_ERROR; -} - -}; // namespace android diff --git a/camera/libcameraservice/CameraService.h b/camera/libcameraservice/CameraService.h deleted file mode 100644 index 8193e77..0000000 --- a/camera/libcameraservice/CameraService.h +++ /dev/null @@ -1,194 +0,0 @@ -/* -** -** Copyright (C) 2008, The Android Open Source Project -** Copyright (C) 2008 HTC Inc. -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#ifndef ANDROID_SERVERS_CAMERA_CAMERASERVICE_H -#define ANDROID_SERVERS_CAMERA_CAMERASERVICE_H - -#include <camera/ICameraService.h> -#include <camera/CameraHardwareInterface.h> - -/* This needs to be increased if we can have more cameras */ -#define MAX_CAMERAS 2 - -namespace android { - -class MemoryHeapBase; -class MediaPlayer; - -class CameraService: public BnCameraService -{ - class Client; -public: - static void instantiate(); - - CameraService(); - virtual ~CameraService(); - - virtual int32_t getNumberOfCameras(); - virtual status_t getCameraInfo(int cameraId, - struct CameraInfo* cameraInfo); - virtual sp<ICamera> connect(const sp<ICameraClient>& cameraClient, int cameraId); - virtual void removeClient(const sp<ICameraClient>& cameraClient); - virtual sp<Client> getClientById(int cameraId); - - virtual status_t dump(int fd, const Vector<String16>& args); - virtual status_t onTransact(uint32_t code, const Parcel& data, - Parcel* reply, uint32_t flags); - - enum sound_kind { - SOUND_SHUTTER = 0, - SOUND_RECORDING = 1, - NUM_SOUNDS - }; - - void loadSound(); - void playSound(sound_kind kind); - void releaseSound(); - -private: - Mutex mServiceLock; - wp<Client> mClient[MAX_CAMERAS]; // protected by mServiceLock - int mNumberOfCameras; - - // atomics to record whether the hardware is allocated to some client. - volatile int32_t mBusy[MAX_CAMERAS]; - void setCameraBusy(int cameraId); - void setCameraFree(int cameraId); - - // sounds - Mutex mSoundLock; - sp<MediaPlayer> mSoundPlayer[NUM_SOUNDS]; - int mSoundRef; // reference count (release all MediaPlayer when 0) - - class Client : public BnCamera - { - public: - // ICamera interface (see ICamera for details) - virtual void disconnect(); - virtual status_t connect(const sp<ICameraClient>& client); - virtual status_t lock(); - virtual status_t unlock(); - virtual status_t setPreviewDisplay(const sp<ISurface>& surface); - virtual void setPreviewCallbackFlag(int flag); - virtual status_t startPreview(); - virtual void stopPreview(); - virtual bool previewEnabled(); - virtual status_t startRecording(); - virtual void stopRecording(); - virtual bool recordingEnabled(); - virtual void releaseRecordingFrame(const sp<IMemory>& mem); - virtual status_t autoFocus(); - virtual status_t cancelAutoFocus(); - virtual status_t takePicture(); - virtual status_t setParameters(const String8& params); - virtual String8 getParameters() const; - virtual status_t sendCommand(int32_t cmd, int32_t arg1, int32_t arg2); - private: - friend class CameraService; - Client(const sp<CameraService>& cameraService, - const sp<ICameraClient>& cameraClient, - int cameraId, - int clientPid); - ~Client(); - - // return our camera client - const sp<ICameraClient>& getCameraClient() { return mCameraClient; } - - // check whether the calling process matches mClientPid. - status_t checkPid() const; - status_t checkPidAndHardware() const; // also check mHardware != 0 - - // these are internal functions used to set up preview buffers - status_t registerPreviewBuffers(); - status_t setOverlay(); - - // camera operation mode - enum camera_mode { - CAMERA_PREVIEW_MODE = 0, // frame automatically released - CAMERA_RECORDING_MODE = 1, // frame has to be explicitly released by releaseRecordingFrame() - }; - // these are internal functions used for preview/recording - status_t startCameraMode(camera_mode mode); - status_t startPreviewMode(); - status_t startRecordingMode(); - - // these are static callback functions - static void notifyCallback(int32_t msgType, int32_t ext1, int32_t ext2, void* user); - static void dataCallback(int32_t msgType, const sp<IMemory>& dataPtr, void* user); - static void dataCallbackTimestamp(nsecs_t timestamp, int32_t msgType, const sp<IMemory>& dataPtr, void* user); - // convert client from cookie - static sp<Client> getClientFromCookie(void* user); - // handlers for messages - void handleShutter(image_rect_type *size); - void handlePreviewData(const sp<IMemory>& mem); - void handlePostview(const sp<IMemory>& mem); - void handleRawPicture(const sp<IMemory>& mem); - void handleCompressedPicture(const sp<IMemory>& mem); - void handleGenericNotify(int32_t msgType, int32_t ext1, int32_t ext2); - void handleGenericData(int32_t msgType, const sp<IMemory>& dataPtr); - void handleGenericDataTimestamp(nsecs_t timestamp, int32_t msgType, const sp<IMemory>& dataPtr); - - void copyFrameAndPostCopiedFrame( - const sp<ICameraClient>& client, - const sp<IMemoryHeap>& heap, - size_t offset, size_t size); - - // these are initialized in the constructor. - sp<CameraService> mCameraService; // immutable after constructor - sp<ICameraClient> mCameraClient; - int mCameraId; // immutable after constructor - pid_t mClientPid; - sp<CameraHardwareInterface> mHardware; // cleared after disconnect() - bool mUseOverlay; // immutable after constructor - sp<OverlayRef> mOverlayRef; - int mOverlayW; - int mOverlayH; - int mPreviewCallbackFlag; - int mOrientation; - - // Ensures atomicity among the public methods - mutable Mutex mLock; - sp<ISurface> mSurface; - - // If the user want us to return a copy of the preview frame (instead - // of the original one), we allocate mPreviewBuffer and reuse it if possible. - sp<MemoryHeapBase> mPreviewBuffer; - - // We need to avoid the deadlock when the incoming command thread and - // the CameraHardwareInterface callback thread both want to grab mLock. - // An extra flag is used to tell the callback thread that it should stop - // trying to deliver the callback messages if the client is not - // interested in it anymore. For example, if the client is calling - // stopPreview(), the preview frame messages do not need to be delivered - // anymore. - - // This function takes the same parameter as the enableMsgType() and - // disableMsgType() functions in CameraHardwareInterface. - void enableMsgType(int32_t msgType); - void disableMsgType(int32_t msgType); - volatile int32_t mMsgEnabled; - - // This function keeps trying to grab mLock, or give up if the message - // is found to be disabled. It returns true if mLock is grabbed. - bool lockIfMessageWanted(int32_t msgType); - }; -}; - -} // namespace android - -#endif diff --git a/camera/libcameraservice/CannedJpeg.h b/camera/libcameraservice/CannedJpeg.h deleted file mode 100644 index b6266fb..0000000 --- a/camera/libcameraservice/CannedJpeg.h +++ /dev/null @@ -1,734 +0,0 @@ -const int kCannedJpegWidth = 320; -const int kCannedJpegHeight = 240; -const int kCannedJpegSize = 8733; - -const char kCannedJpeg[] = { - 0xff, 0xd8, 0xff, 0xe0, 0x00, 0x10, 0x4a, 0x46, 0x49, 0x46, 0x00, 0x01, - 0x01, 0x01, 0x00, 0x60, 0x00, 0x60, 0x00, 0x00, 0xff, 0xe1, 0x00, 0x66, - 0x45, 0x78, 0x69, 0x66, 0x00, 0x00, 0x49, 0x49, 0x2a, 0x00, 0x08, 0x00, - 0x00, 0x00, 0x04, 0x00, 0x1a, 0x01, 0x05, 0x00, 0x01, 0x00, 0x00, 0x00, - 0x3e, 0x00, 0x00, 0x00, 0x1b, 0x01, 0x05, 0x00, 0x01, 0x00, 0x00, 0x00, - 0x46, 0x00, 0x00, 0x00, 0x28, 0x01, 0x03, 0x00, 0x01, 0x00, 0x00, 0x00, - 0x02, 0x00, 0x00, 0x00, 0x31, 0x01, 0x02, 0x00, 0x10, 0x00, 0x00, 0x00, - 0x4e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x60, 0x00, 0x00, 0x00, - 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0x62, 0x82, 0x70, 0x33, 0x40, 0x11, 0xca, 0xdd, 0xaa, 0x3a, 0x52, 0x72, - 0x73, 0x42, 0x8d, 0xc4, 0x0a, 0x00, 0x92, 0x25, 0xc0, 0xcd, 0x3e, 0x8e, - 0x94, 0x8c, 0x76, 0x82, 0x68, 0x02, 0x39, 0x5b, 0x27, 0x1e, 0x94, 0xca, - 0x3a, 0xd3, 0x91, 0x77, 0x35, 0x00, 0x49, 0x1a, 0xe1, 0x7d, 0xcd, 0x3a, - 0x8a, 0x47, 0x6d, 0xab, 0x40, 0x11, 0xc8, 0xd9, 0x6f, 0x61, 0x4c, 0xa2, - 0x9d, 0x1a, 0xee, 0x6f, 0x61, 0x40, 0x1f, 0xff, 0xd9 -}; diff --git a/camera/libcameraservice/FakeCamera.cpp b/camera/libcameraservice/FakeCamera.cpp deleted file mode 100644 index f3a6a67..0000000 --- a/camera/libcameraservice/FakeCamera.cpp +++ /dev/null @@ -1,433 +0,0 @@ -/* -** -** Copyright 2008, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#define LOG_TAG "FakeCamera" -#include <utils/Log.h> - -#include <string.h> -#include <stdlib.h> -#include <utils/String8.h> - -#include "FakeCamera.h" - - -namespace android { - -// TODO: All this rgb to yuv should probably be in a util class. - -// TODO: I think something is wrong in this class because the shadow is kBlue -// and the square color should alternate between kRed and kGreen. However on the -// emulator screen these are all shades of gray. Y seems ok but the U and V are -// probably not. - -static int tables_initialized = 0; -uint8_t *gYTable, *gCbTable, *gCrTable; - -static int -clamp(int x) -{ - if (x > 255) return 255; - if (x < 0) return 0; - return x; -} - -/* the equation used by the video code to translate YUV to RGB looks like this - * - * Y = (Y0 - 16)*k0 - * Cb = Cb0 - 128 - * Cr = Cr0 - 128 - * - * G = ( Y - k1*Cr - k2*Cb ) - * R = ( Y + k3*Cr ) - * B = ( Y + k4*Cb ) - * - */ - -static const double k0 = 1.164; -static const double k1 = 0.813; -static const double k2 = 0.391; -static const double k3 = 1.596; -static const double k4 = 2.018; - -/* let's try to extract the value of Y - * - * G + k1/k3*R + k2/k4*B = Y*( 1 + k1/k3 + k2/k4 ) - * - * Y = ( G + k1/k3*R + k2/k4*B ) / (1 + k1/k3 + k2/k4) - * Y0 = ( G0 + k1/k3*R0 + k2/k4*B0 ) / ((1 + k1/k3 + k2/k4)*k0) + 16 - * - * let define: - * kYr = k1/k3 - * kYb = k2/k4 - * kYy = k0 * ( 1 + kYr + kYb ) - * - * we have: - * Y = ( G + kYr*R + kYb*B ) - * Y0 = clamp[ Y/kYy + 16 ] - */ - -static const double kYr = k1/k3; -static const double kYb = k2/k4; -static const double kYy = k0*( 1. + kYr + kYb ); - -static void -initYtab( void ) -{ - const int imax = (int)( (kYr + kYb)*(31 << 2) + (61 << 3) + 0.1 ); - int i; - - gYTable = (uint8_t *)malloc(imax); - - for(i=0; i<imax; i++) { - int x = (int)(i/kYy + 16.5); - if (x < 16) x = 16; - else if (x > 235) x = 235; - gYTable[i] = (uint8_t) x; - } -} - -/* - * the source is RGB565, so adjust for 8-bit range of input values: - * - * G = (pixels >> 3) & 0xFC; - * R = (pixels >> 8) & 0xF8; - * B = (pixels & 0x1f) << 3; - * - * R2 = (pixels >> 11) R = R2*8 - * B2 = (pixels & 0x1f) B = B2*8 - * - * kYr*R = kYr2*R2 => kYr2 = kYr*8 - * kYb*B = kYb2*B2 => kYb2 = kYb*8 - * - * we want to use integer multiplications: - * - * SHIFT1 = 9 - * - * (ALPHA*R2) >> SHIFT1 == R*kYr => ALPHA = kYr*8*(1 << SHIFT1) - * - * ALPHA = kYr*(1 << (SHIFT1+3)) - * BETA = kYb*(1 << (SHIFT1+3)) - */ - -static const int SHIFT1 = 9; -static const int ALPHA = (int)( kYr*(1 << (SHIFT1+3)) + 0.5 ); -static const int BETA = (int)( kYb*(1 << (SHIFT1+3)) + 0.5 ); - -/* - * now let's try to get the values of Cb and Cr - * - * R-B = (k3*Cr - k4*Cb) - * - * k3*Cr = k4*Cb + (R-B) - * k4*Cb = k3*Cr - (R-B) - * - * R-G = (k1+k3)*Cr + k2*Cb - * = (k1+k3)*Cr + k2/k4*(k3*Cr - (R-B)/k0) - * = (k1 + k3 + k2*k3/k4)*Cr - k2/k4*(R-B) - * - * kRr*Cr = (R-G) + kYb*(R-B) - * - * Cr = ((R-G) + kYb*(R-B))/kRr - * Cr0 = clamp(Cr + 128) - */ - -static const double kRr = (k1 + k3 + k2*k3/k4); - -static void -initCrtab( void ) -{ - uint8_t *pTable; - int i; - - gCrTable = (uint8_t *)malloc(768*2); - - pTable = gCrTable + 384; - for(i=-384; i<384; i++) - pTable[i] = (uint8_t) clamp( i/kRr + 128.5 ); -} - -/* - * B-G = (k2 + k4)*Cb + k1*Cr - * = (k2 + k4)*Cb + k1/k3*(k4*Cb + (R-B)) - * = (k2 + k4 + k1*k4/k3)*Cb + k1/k3*(R-B) - * - * kBb*Cb = (B-G) - kYr*(R-B) - * - * Cb = ((B-G) - kYr*(R-B))/kBb - * Cb0 = clamp(Cb + 128) - * - */ - -static const double kBb = (k2 + k4 + k1*k4/k3); - -static void -initCbtab( void ) -{ - uint8_t *pTable; - int i; - - gCbTable = (uint8_t *)malloc(768*2); - - pTable = gCbTable + 384; - for(i=-384; i<384; i++) - pTable[i] = (uint8_t) clamp( i/kBb + 128.5 ); -} - -/* - * SHIFT2 = 16 - * - * DELTA = kYb*(1 << SHIFT2) - * GAMMA = kYr*(1 << SHIFT2) - */ - -static const int SHIFT2 = 16; -static const int DELTA = kYb*(1 << SHIFT2); -static const int GAMMA = kYr*(1 << SHIFT2); - -int32_t ccrgb16toyuv_wo_colorkey(uint8_t *rgb16, uint8_t *yuv420, - uint32_t *param, uint8_t *table[]) -{ - uint16_t *inputRGB = (uint16_t*)rgb16; - uint8_t *outYUV = yuv420; - int32_t width_dst = param[0]; - int32_t height_dst = param[1]; - int32_t pitch_dst = param[2]; - int32_t mheight_dst = param[3]; - int32_t pitch_src = param[4]; - uint8_t *y_tab = table[0]; - uint8_t *cb_tab = table[1]; - uint8_t *cr_tab = table[2]; - - int32_t size16 = pitch_dst*mheight_dst; - int32_t i,j,count; - int32_t ilimit,jlimit; - uint8_t *tempY,*tempU,*tempV; - uint16_t pixels; - int tmp; -uint32_t temp; - - tempY = outYUV; - tempU = outYUV + (height_dst * pitch_dst); - tempV = tempU + 1; - - jlimit = height_dst; - ilimit = width_dst; - - for(j=0; j<jlimit; j+=1) - { - for (i=0; i<ilimit; i+=2) - { - int32_t G_ds = 0, B_ds = 0, R_ds = 0; - uint8_t y0, y1, u, v; - - pixels = inputRGB[i]; - temp = (BETA*(pixels & 0x001F) + ALPHA*(pixels>>11) ); - y0 = y_tab[(temp>>SHIFT1) + ((pixels>>3) & 0x00FC)]; - - G_ds += (pixels>>1) & 0x03E0; - B_ds += (pixels<<5) & 0x03E0; - R_ds += (pixels>>6) & 0x03E0; - - pixels = inputRGB[i+1]; - temp = (BETA*(pixels & 0x001F) + ALPHA*(pixels>>11) ); - y1 = y_tab[(temp>>SHIFT1) + ((pixels>>3) & 0x00FC)]; - - G_ds += (pixels>>1) & 0x03E0; - B_ds += (pixels<<5) & 0x03E0; - R_ds += (pixels>>6) & 0x03E0; - - R_ds >>= 1; - B_ds >>= 1; - G_ds >>= 1; - - tmp = R_ds - B_ds; - - u = cb_tab[(((B_ds-G_ds)<<SHIFT2) - GAMMA*tmp)>>(SHIFT2+2)]; - v = cr_tab[(((R_ds-G_ds)<<SHIFT2) + DELTA*tmp)>>(SHIFT2+2)]; - - tempY[0] = y0; - tempY[1] = y1; - tempY += 2; - - if ((j&1) == 0) { - tempU[0] = u; - tempV[0] = v; - tempU += 2; - tempV += 2; - } - } - - inputRGB += pitch_src; - } - - return 1; -} - -#define min(a,b) ((a)<(b)?(a):(b)) -#define max(a,b) ((a)>(b)?(a):(b)) - -static void convert_rgb16_to_yuv420(uint8_t *rgb, uint8_t *yuv, int width, int height) -{ - if (!tables_initialized) { - initYtab(); - initCrtab(); - initCbtab(); - tables_initialized = 1; - } - - uint32_t param[6]; - param[0] = (uint32_t) width; - param[1] = (uint32_t) height; - param[2] = (uint32_t) width; - param[3] = (uint32_t) height; - param[4] = (uint32_t) width; - param[5] = (uint32_t) 0; - - uint8_t *table[3]; - table[0] = gYTable; - table[1] = gCbTable + 384; - table[2] = gCrTable + 384; - - ccrgb16toyuv_wo_colorkey(rgb, yuv, param, table); -} - -const int FakeCamera::kRed; -const int FakeCamera::kGreen; -const int FakeCamera::kBlue; - -FakeCamera::FakeCamera(int width, int height) - : mTmpRgb16Buffer(0) -{ - setSize(width, height); -} - -FakeCamera::~FakeCamera() -{ - delete[] mTmpRgb16Buffer; -} - -void FakeCamera::setSize(int width, int height) -{ - mWidth = width; - mHeight = height; - mCounter = 0; - mCheckX = 0; - mCheckY = 0; - - // This will cause it to be reallocated on the next call - // to getNextFrameAsYuv420(). - delete[] mTmpRgb16Buffer; - mTmpRgb16Buffer = 0; -} - -void FakeCamera::getNextFrameAsRgb565(uint16_t *buffer) -{ - int size = mWidth / 10; - - drawCheckerboard(buffer, size); - - int x = ((mCounter*3)&255); - if(x>128) x = 255 - x; - int y = ((mCounter*5)&255); - if(y>128) y = 255 - y; - - drawSquare(buffer, x*size/32, y*size/32, (size*5)>>1, (mCounter&0x100)?kRed:kGreen, kBlue); - - mCounter++; -} - -void FakeCamera::getNextFrameAsYuv420(uint8_t *buffer) -{ - if (mTmpRgb16Buffer == 0) - mTmpRgb16Buffer = new uint16_t[mWidth * mHeight]; - - getNextFrameAsRgb565(mTmpRgb16Buffer); - convert_rgb16_to_yuv420((uint8_t*)mTmpRgb16Buffer, buffer, mWidth, mHeight); -} - -void FakeCamera::drawSquare(uint16_t *dst, int x, int y, int size, int color, int shadow) -{ - int square_xstop, square_ystop, shadow_xstop, shadow_ystop; - - square_xstop = min(mWidth, x+size); - square_ystop = min(mHeight, y+size); - shadow_xstop = min(mWidth, x+size+(size/4)); - shadow_ystop = min(mHeight, y+size+(size/4)); - - // Do the shadow. - uint16_t *sh = &dst[(y+(size/4))*mWidth]; - for (int j = y + (size/4); j < shadow_ystop; j++) { - for (int i = x + (size/4); i < shadow_xstop; i++) { - sh[i] &= shadow; - } - sh += mWidth; - } - - // Draw the square. - uint16_t *sq = &dst[y*mWidth]; - for (int j = y; j < square_ystop; j++) { - for (int i = x; i < square_xstop; i++) { - sq[i] = color; - } - sq += mWidth; - } -} - -void FakeCamera::drawCheckerboard(uint16_t *dst, int size) -{ - bool black = true; - - if((mCheckX/size)&1) - black = false; - if((mCheckY/size)&1) - black = !black; - - int county = mCheckY%size; - int checkxremainder = mCheckX%size; - - for(int y=0;y<mHeight;y++) { - int countx = checkxremainder; - bool current = black; - for(int x=0;x<mWidth;x++) { - dst[y*mWidth+x] = current?0:0xffff; - if(countx++ >= size) { - countx=0; - current = !current; - } - } - if(county++ >= size) { - county=0; - black = !black; - } - } - mCheckX += 3; - mCheckY++; -} - - -void FakeCamera::dump(int fd) const -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - snprintf(buffer, 255, " width x height (%d x %d), counter (%d), check x-y coordinate(%d, %d)\n", mWidth, mHeight, mCounter, mCheckX, mCheckY); - result.append(buffer); - ::write(fd, result.string(), result.size()); -} - - -}; // namespace android diff --git a/camera/libcameraservice/FakeCamera.h b/camera/libcameraservice/FakeCamera.h deleted file mode 100644 index 724de20..0000000 --- a/camera/libcameraservice/FakeCamera.h +++ /dev/null @@ -1,67 +0,0 @@ -/* -** -** Copyright 2008, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#ifndef ANDROID_HARDWARE_FAKECAMERA_H -#define ANDROID_HARDWARE_FAKECAMERA_H - -#include <sys/types.h> -#include <stdint.h> - -namespace android { - -/* - * FakeCamera is used in the CameraHardwareStub to provide a fake video feed - * when the system does not have a camera in hardware. - * The fake video is a moving black and white checkerboard background with a - * bouncing gray square in the foreground. - * This class is not thread-safe. - * - * TODO: Since the major methods provides a raw/uncompressed video feed, rename - * this class to RawVideoSource. - */ - -class FakeCamera { -public: - FakeCamera(int width, int height); - ~FakeCamera(); - - void setSize(int width, int height); - void getNextFrameAsYuv420(uint8_t *buffer); - // Write to the fd a string representing the current state. - void dump(int fd) const; - -private: - // TODO: remove the uint16_t buffer param everywhere since it is a field of - // this class. - void getNextFrameAsRgb565(uint16_t *buffer); - - void drawSquare(uint16_t *buffer, int x, int y, int size, int color, int shadow); - void drawCheckerboard(uint16_t *buffer, int size); - - static const int kRed = 0xf800; - static const int kGreen = 0x07c0; - static const int kBlue = 0x003e; - - int mWidth, mHeight; - int mCounter; - int mCheckX, mCheckY; - uint16_t *mTmpRgb16Buffer; -}; - -}; // namespace android - -#endif // ANDROID_HARDWARE_FAKECAMERA_H diff --git a/camera/tests/CameraServiceTest/Android.mk b/camera/tests/CameraServiceTest/Android.mk deleted file mode 100644 index cf4e42f..0000000 --- a/camera/tests/CameraServiceTest/Android.mk +++ /dev/null @@ -1,26 +0,0 @@ -LOCAL_PATH:= $(call my-dir) - -include $(CLEAR_VARS) - -LOCAL_SRC_FILES:= CameraServiceTest.cpp - -LOCAL_MODULE:= CameraServiceTest - -LOCAL_MODULE_TAGS := tests - -LOCAL_C_INCLUDES += \ - frameworks/base/libs - -LOCAL_CFLAGS := - -LOCAL_SHARED_LIBRARIES += \ - libbinder \ - libcutils \ - libutils \ - libui \ - libcamera_client \ - libsurfaceflinger_client - -# Disable it because the ISurface interface may change, and before we have a -# chance to fix this test, we don't want to break normal builds. -#include $(BUILD_EXECUTABLE) diff --git a/camera/tests/CameraServiceTest/CameraServiceTest.cpp b/camera/tests/CameraServiceTest/CameraServiceTest.cpp deleted file mode 100644 index 3c8d553..0000000 --- a/camera/tests/CameraServiceTest/CameraServiceTest.cpp +++ /dev/null @@ -1,919 +0,0 @@ -#define LOG_TAG "CameraServiceTest" - -#include <stdio.h> -#include <stdlib.h> -#include <string.h> -#include <sys/types.h> -#include <sys/wait.h> -#include <unistd.h> -#include <surfaceflinger/ISurface.h> -#include <camera/Camera.h> -#include <camera/CameraParameters.h> -#include <ui/GraphicBuffer.h> -#include <camera/ICamera.h> -#include <camera/ICameraClient.h> -#include <camera/ICameraService.h> -#include <ui/Overlay.h> -#include <binder/IPCThreadState.h> -#include <binder/IServiceManager.h> -#include <binder/ProcessState.h> -#include <utils/KeyedVector.h> -#include <utils/Log.h> -#include <utils/Vector.h> -#include <utils/threads.h> - -using namespace android; - -// -// Assertion and Logging utilities -// -#define INFO(...) \ - do { \ - printf(__VA_ARGS__); \ - printf("\n"); \ - LOGD(__VA_ARGS__); \ - } while(0) - -void assert_fail(const char *file, int line, const char *func, const char *expr) { - INFO("assertion failed at file %s, line %d, function %s:", - file, line, func); - INFO("%s", expr); - abort(); -} - -void assert_eq_fail(const char *file, int line, const char *func, - const char *expr, int actual) { - INFO("assertion failed at file %s, line %d, function %s:", - file, line, func); - INFO("(expected) %s != (actual) %d", expr, actual); - abort(); -} - -#define ASSERT(e) \ - do { \ - if (!(e)) \ - assert_fail(__FILE__, __LINE__, __func__, #e); \ - } while(0) - -#define ASSERT_EQ(expected, actual) \ - do { \ - int _x = (actual); \ - if (_x != (expected)) \ - assert_eq_fail(__FILE__, __LINE__, __func__, #expected, _x); \ - } while(0) - -// -// Holder service for pass objects between processes. -// -class IHolder : public IInterface { -protected: - enum { - HOLDER_PUT = IBinder::FIRST_CALL_TRANSACTION, - HOLDER_GET, - HOLDER_CLEAR - }; -public: - DECLARE_META_INTERFACE(Holder); - - virtual void put(sp<IBinder> obj) = 0; - virtual sp<IBinder> get() = 0; - virtual void clear() = 0; -}; - -class BnHolder : public BnInterface<IHolder> { - virtual status_t onTransact(uint32_t code, - const Parcel& data, - Parcel* reply, - uint32_t flags = 0); -}; - -class BpHolder : public BpInterface<IHolder> { -public: - BpHolder(const sp<IBinder>& impl) - : BpInterface<IHolder>(impl) { - } - - virtual void put(sp<IBinder> obj) { - Parcel data, reply; - data.writeStrongBinder(obj); - remote()->transact(HOLDER_PUT, data, &reply, IBinder::FLAG_ONEWAY); - } - - virtual sp<IBinder> get() { - Parcel data, reply; - remote()->transact(HOLDER_GET, data, &reply); - return reply.readStrongBinder(); - } - - virtual void clear() { - Parcel data, reply; - remote()->transact(HOLDER_CLEAR, data, &reply); - } -}; - -IMPLEMENT_META_INTERFACE(Holder, "CameraServiceTest.Holder"); - -status_t BnHolder::onTransact( - uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { - switch(code) { - case HOLDER_PUT: { - put(data.readStrongBinder()); - return NO_ERROR; - } break; - case HOLDER_GET: { - reply->writeStrongBinder(get()); - return NO_ERROR; - } break; - case HOLDER_CLEAR: { - clear(); - return NO_ERROR; - } break; - default: - return BBinder::onTransact(code, data, reply, flags); - } -} - -class HolderService : public BnHolder { - virtual void put(sp<IBinder> obj) { - mObj = obj; - } - virtual sp<IBinder> get() { - return mObj; - } - virtual void clear() { - mObj.clear(); - } -private: - sp<IBinder> mObj; -}; - -// -// A mock CameraClient -// -class MCameraClient : public BnCameraClient { -public: - virtual void notifyCallback(int32_t msgType, int32_t ext1, int32_t ext2); - virtual void dataCallback(int32_t msgType, const sp<IMemory>& data); - virtual void dataCallbackTimestamp(nsecs_t timestamp, - int32_t msgType, const sp<IMemory>& data); - - // new functions - void clearStat(); - enum OP { EQ, GE, LE, GT, LT }; - void assertNotify(int32_t msgType, OP op, int count); - void assertData(int32_t msgType, OP op, int count); - void waitNotify(int32_t msgType, OP op, int count); - void waitData(int32_t msgType, OP op, int count); - void assertDataSize(int32_t msgType, OP op, int dataSize); - - void setReleaser(ICamera *releaser) { - mReleaser = releaser; - } -private: - Mutex mLock; - Condition mCond; - DefaultKeyedVector<int32_t, int> mNotifyCount; - DefaultKeyedVector<int32_t, int> mDataCount; - DefaultKeyedVector<int32_t, int> mDataSize; - bool test(OP op, int v1, int v2); - void assertTest(OP op, int v1, int v2); - - ICamera *mReleaser; -}; - -void MCameraClient::clearStat() { - Mutex::Autolock _l(mLock); - mNotifyCount.clear(); - mDataCount.clear(); - mDataSize.clear(); -} - -bool MCameraClient::test(OP op, int v1, int v2) { - switch (op) { - case EQ: return v1 == v2; - case GT: return v1 > v2; - case LT: return v1 < v2; - case GE: return v1 >= v2; - case LE: return v1 <= v2; - default: ASSERT(0); break; - } - return false; -} - -void MCameraClient::assertTest(OP op, int v1, int v2) { - if (!test(op, v1, v2)) { - LOGE("assertTest failed: op=%d, v1=%d, v2=%d", op, v1, v2); - ASSERT(0); - } -} - -void MCameraClient::assertNotify(int32_t msgType, OP op, int count) { - Mutex::Autolock _l(mLock); - int v = mNotifyCount.valueFor(msgType); - assertTest(op, v, count); -} - -void MCameraClient::assertData(int32_t msgType, OP op, int count) { - Mutex::Autolock _l(mLock); - int v = mDataCount.valueFor(msgType); - assertTest(op, v, count); -} - -void MCameraClient::assertDataSize(int32_t msgType, OP op, int dataSize) { - Mutex::Autolock _l(mLock); - int v = mDataSize.valueFor(msgType); - assertTest(op, v, dataSize); -} - -void MCameraClient::notifyCallback(int32_t msgType, int32_t ext1, int32_t ext2) { - INFO("%s", __func__); - Mutex::Autolock _l(mLock); - ssize_t i = mNotifyCount.indexOfKey(msgType); - if (i < 0) { - mNotifyCount.add(msgType, 1); - } else { - ++mNotifyCount.editValueAt(i); - } - mCond.signal(); -} - -void MCameraClient::dataCallback(int32_t msgType, const sp<IMemory>& data) { - INFO("%s", __func__); - int dataSize = data->size(); - INFO("data type = %d, size = %d", msgType, dataSize); - Mutex::Autolock _l(mLock); - ssize_t i = mDataCount.indexOfKey(msgType); - if (i < 0) { - mDataCount.add(msgType, 1); - mDataSize.add(msgType, dataSize); - } else { - ++mDataCount.editValueAt(i); - mDataSize.editValueAt(i) = dataSize; - } - mCond.signal(); - - if (msgType == CAMERA_MSG_VIDEO_FRAME) { - ASSERT(mReleaser != NULL); - mReleaser->releaseRecordingFrame(data); - } -} - -void MCameraClient::dataCallbackTimestamp(nsecs_t timestamp, int32_t msgType, - const sp<IMemory>& data) { - dataCallback(msgType, data); -} - -void MCameraClient::waitNotify(int32_t msgType, OP op, int count) { - INFO("waitNotify: %d, %d, %d", msgType, op, count); - Mutex::Autolock _l(mLock); - while (true) { - int v = mNotifyCount.valueFor(msgType); - if (test(op, v, count)) { - break; - } - mCond.wait(mLock); - } -} - -void MCameraClient::waitData(int32_t msgType, OP op, int count) { - INFO("waitData: %d, %d, %d", msgType, op, count); - Mutex::Autolock _l(mLock); - while (true) { - int v = mDataCount.valueFor(msgType); - if (test(op, v, count)) { - break; - } - mCond.wait(mLock); - } -} - -// -// A mock Surface -// -class MSurface : public BnSurface { -public: - virtual status_t registerBuffers(const BufferHeap& buffers); - virtual void postBuffer(ssize_t offset); - virtual void unregisterBuffers(); - virtual sp<OverlayRef> createOverlay( - uint32_t w, uint32_t h, int32_t format, int32_t orientation); - virtual sp<GraphicBuffer> requestBuffer(int bufferIdx, int usage); - virtual status_t setBufferCount(int bufferCount); - - // new functions - void clearStat(); - void waitUntil(int c0, int c1, int c2); - -private: - // check callback count - Condition mCond; - Mutex mLock; - int registerBuffersCount; - int postBufferCount; - int unregisterBuffersCount; -}; - -status_t MSurface::registerBuffers(const BufferHeap& buffers) { - INFO("%s", __func__); - Mutex::Autolock _l(mLock); - ++registerBuffersCount; - mCond.signal(); - return NO_ERROR; -} - -void MSurface::postBuffer(ssize_t offset) { - // INFO("%s", __func__); - Mutex::Autolock _l(mLock); - ++postBufferCount; - mCond.signal(); -} - -void MSurface::unregisterBuffers() { - INFO("%s", __func__); - Mutex::Autolock _l(mLock); - ++unregisterBuffersCount; - mCond.signal(); -} - -sp<GraphicBuffer> MSurface::requestBuffer(int bufferIdx, int usage) { - INFO("%s", __func__); - return NULL; -} - -status_t MSurface::setBufferCount(int bufferCount) { - INFO("%s", __func__); - return NULL; -} - -void MSurface::clearStat() { - Mutex::Autolock _l(mLock); - registerBuffersCount = 0; - postBufferCount = 0; - unregisterBuffersCount = 0; -} - -void MSurface::waitUntil(int c0, int c1, int c2) { - INFO("waitUntil: %d %d %d", c0, c1, c2); - Mutex::Autolock _l(mLock); - while (true) { - if (registerBuffersCount >= c0 && - postBufferCount >= c1 && - unregisterBuffersCount >= c2) { - break; - } - mCond.wait(mLock); - } -} - -sp<OverlayRef> MSurface::createOverlay(uint32_t w, uint32_t h, int32_t format, - int32_t orientation) { - // Not implemented. - ASSERT(0); - return NULL; -} - -// -// Utilities to use the Holder service -// -sp<IHolder> getHolder() { - sp<IServiceManager> sm = defaultServiceManager(); - ASSERT(sm != 0); - sp<IBinder> binder = sm->getService(String16("CameraServiceTest.Holder")); - ASSERT(binder != 0); - sp<IHolder> holder = interface_cast<IHolder>(binder); - ASSERT(holder != 0); - return holder; -} - -void putTempObject(sp<IBinder> obj) { - INFO("%s", __func__); - getHolder()->put(obj); -} - -sp<IBinder> getTempObject() { - INFO("%s", __func__); - return getHolder()->get(); -} - -void clearTempObject() { - INFO("%s", __func__); - getHolder()->clear(); -} - -// -// Get a Camera Service -// -sp<ICameraService> getCameraService() { - sp<IServiceManager> sm = defaultServiceManager(); - ASSERT(sm != 0); - sp<IBinder> binder = sm->getService(String16("media.camera")); - ASSERT(binder != 0); - sp<ICameraService> cs = interface_cast<ICameraService>(binder); - ASSERT(cs != 0); - return cs; -} - -int getNumberOfCameras() { - sp<ICameraService> cs = getCameraService(); - return cs->getNumberOfCameras(); -} - -// -// Various Connect Tests -// -void testConnect(int cameraId) { - INFO("%s", __func__); - sp<ICameraService> cs = getCameraService(); - sp<MCameraClient> cc = new MCameraClient(); - sp<ICamera> c = cs->connect(cc, cameraId); - ASSERT(c != 0); - c->disconnect(); -} - -void testAllowConnectOnceOnly(int cameraId) { - INFO("%s", __func__); - sp<ICameraService> cs = getCameraService(); - // Connect the first client. - sp<MCameraClient> cc = new MCameraClient(); - sp<ICamera> c = cs->connect(cc, cameraId); - ASSERT(c != 0); - // Same client -- ok. - ASSERT(cs->connect(cc, cameraId) != 0); - // Different client -- not ok. - sp<MCameraClient> cc2 = new MCameraClient(); - ASSERT(cs->connect(cc2, cameraId) == 0); - c->disconnect(); -} - -void testReconnectFailed() { - INFO("%s", __func__); - sp<ICamera> c = interface_cast<ICamera>(getTempObject()); - sp<MCameraClient> cc = new MCameraClient(); - ASSERT(c->connect(cc) != NO_ERROR); -} - -void testReconnectSuccess() { - INFO("%s", __func__); - sp<ICamera> c = interface_cast<ICamera>(getTempObject()); - sp<MCameraClient> cc = new MCameraClient(); - ASSERT(c->connect(cc) == NO_ERROR); - c->disconnect(); -} - -void testLockFailed() { - INFO("%s", __func__); - sp<ICamera> c = interface_cast<ICamera>(getTempObject()); - ASSERT(c->lock() != NO_ERROR); -} - -void testLockUnlockSuccess() { - INFO("%s", __func__); - sp<ICamera> c = interface_cast<ICamera>(getTempObject()); - ASSERT(c->lock() == NO_ERROR); - ASSERT(c->unlock() == NO_ERROR); -} - -void testLockSuccess() { - INFO("%s", __func__); - sp<ICamera> c = interface_cast<ICamera>(getTempObject()); - ASSERT(c->lock() == NO_ERROR); - c->disconnect(); -} - -// -// Run the connect tests in another process. -// -const char *gExecutable; - -struct FunctionTableEntry { - const char *name; - void (*func)(); -}; - -FunctionTableEntry function_table[] = { -#define ENTRY(x) {#x, &x} - ENTRY(testReconnectFailed), - ENTRY(testReconnectSuccess), - ENTRY(testLockUnlockSuccess), - ENTRY(testLockFailed), - ENTRY(testLockSuccess), -#undef ENTRY -}; - -void runFunction(const char *tag) { - INFO("runFunction: %s", tag); - int entries = sizeof(function_table) / sizeof(function_table[0]); - for (int i = 0; i < entries; i++) { - if (strcmp(function_table[i].name, tag) == 0) { - (*function_table[i].func)(); - return; - } - } - ASSERT(0); -} - -void runInAnotherProcess(const char *tag) { - pid_t pid = fork(); - if (pid == 0) { - execlp(gExecutable, gExecutable, tag, NULL); - ASSERT(0); - } else { - int status; - ASSERT_EQ(pid, wait(&status)); - ASSERT_EQ(0, status); - } -} - -void testReconnect(int cameraId) { - INFO("%s", __func__); - sp<ICameraService> cs = getCameraService(); - sp<MCameraClient> cc = new MCameraClient(); - sp<ICamera> c = cs->connect(cc, cameraId); - ASSERT(c != 0); - // Reconnect to the same client -- ok. - ASSERT(c->connect(cc) == NO_ERROR); - // Reconnect to a different client (but the same pid) -- ok. - sp<MCameraClient> cc2 = new MCameraClient(); - ASSERT(c->connect(cc2) == NO_ERROR); - c->disconnect(); - cc->assertNotify(CAMERA_MSG_ERROR, MCameraClient::EQ, 0); -} - -void testLockUnlock(int cameraId) { - sp<ICameraService> cs = getCameraService(); - sp<MCameraClient> cc = new MCameraClient(); - sp<ICamera> c = cs->connect(cc, cameraId); - ASSERT(c != 0); - // We can lock as many times as we want. - ASSERT(c->lock() == NO_ERROR); - ASSERT(c->lock() == NO_ERROR); - // Lock from a different process -- not ok. - putTempObject(c->asBinder()); - runInAnotherProcess("testLockFailed"); - // Unlock then lock from a different process -- ok. - ASSERT(c->unlock() == NO_ERROR); - runInAnotherProcess("testLockUnlockSuccess"); - // Unlock then lock from a different process -- ok. - runInAnotherProcess("testLockSuccess"); - clearTempObject(); -} - -void testReconnectFromAnotherProcess(int cameraId) { - INFO("%s", __func__); - - sp<ICameraService> cs = getCameraService(); - sp<MCameraClient> cc = new MCameraClient(); - sp<ICamera> c = cs->connect(cc, cameraId); - ASSERT(c != 0); - // Reconnect from a different process -- not ok. - putTempObject(c->asBinder()); - runInAnotherProcess("testReconnectFailed"); - // Unlock then reconnect from a different process -- ok. - ASSERT(c->unlock() == NO_ERROR); - runInAnotherProcess("testReconnectSuccess"); - clearTempObject(); -} - -// We need to flush the command buffer after the reference -// to ICamera is gone. The sleep is for the server to run -// the destructor for it. -static void flushCommands() { - IPCThreadState::self()->flushCommands(); - usleep(200000); // 200ms -} - -// Run a test case -#define RUN(class_name, cameraId) do { \ - { \ - INFO(#class_name); \ - class_name instance; \ - instance.init(cameraId); \ - instance.run(); \ - } \ - flushCommands(); \ -} while(0) - -// Base test case after the the camera is connected. -class AfterConnect { -public: - void init(int cameraId) { - cs = getCameraService(); - cc = new MCameraClient(); - c = cs->connect(cc, cameraId); - ASSERT(c != 0); - } - -protected: - sp<ICameraService> cs; - sp<MCameraClient> cc; - sp<ICamera> c; - - ~AfterConnect() { - c->disconnect(); - c.clear(); - cc.clear(); - cs.clear(); - } -}; - -class TestSetPreviewDisplay : public AfterConnect { -public: - void run() { - sp<MSurface> surface = new MSurface(); - ASSERT(c->setPreviewDisplay(surface) == NO_ERROR); - c->disconnect(); - cc->assertNotify(CAMERA_MSG_ERROR, MCameraClient::EQ, 0); - } -}; - -class TestStartPreview : public AfterConnect { -public: - void run() { - sp<MSurface> surface = new MSurface(); - ASSERT(c->setPreviewDisplay(surface) == NO_ERROR); - - ASSERT(c->startPreview() == NO_ERROR); - ASSERT(c->previewEnabled() == true); - - surface->waitUntil(1, 10, 0); // needs 1 registerBuffers and 10 postBuffer - surface->clearStat(); - - sp<MSurface> another_surface = new MSurface(); - c->setPreviewDisplay(another_surface); // just to make sure unregisterBuffers - // is called. - surface->waitUntil(0, 0, 1); // needs unregisterBuffers - - cc->assertNotify(CAMERA_MSG_ERROR, MCameraClient::EQ, 0); - } -}; - -class TestStartPreviewWithoutDisplay : public AfterConnect { -public: - void run() { - ASSERT(c->startPreview() == NO_ERROR); - ASSERT(c->previewEnabled() == true); - c->disconnect(); - cc->assertNotify(CAMERA_MSG_ERROR, MCameraClient::EQ, 0); - } -}; - -// Base test case after the the camera is connected and the preview is started. -class AfterStartPreview : public AfterConnect { -public: - void init(int cameraId) { - AfterConnect::init(cameraId); - surface = new MSurface(); - ASSERT(c->setPreviewDisplay(surface) == NO_ERROR); - ASSERT(c->startPreview() == NO_ERROR); - } - -protected: - sp<MSurface> surface; - - ~AfterStartPreview() { - surface.clear(); - } -}; - -class TestAutoFocus : public AfterStartPreview { -public: - void run() { - cc->assertNotify(CAMERA_MSG_FOCUS, MCameraClient::EQ, 0); - c->autoFocus(); - cc->waitNotify(CAMERA_MSG_FOCUS, MCameraClient::EQ, 1); - c->disconnect(); - cc->assertNotify(CAMERA_MSG_ERROR, MCameraClient::EQ, 0); - } -}; - -class TestStopPreview : public AfterStartPreview { -public: - void run() { - ASSERT(c->previewEnabled() == true); - c->stopPreview(); - ASSERT(c->previewEnabled() == false); - c->disconnect(); - cc->assertNotify(CAMERA_MSG_ERROR, MCameraClient::EQ, 0); - } -}; - -class TestTakePicture: public AfterStartPreview { -public: - void run() { - ASSERT(c->takePicture() == NO_ERROR); - cc->waitNotify(CAMERA_MSG_SHUTTER, MCameraClient::EQ, 1); - cc->waitData(CAMERA_MSG_RAW_IMAGE, MCameraClient::EQ, 1); - cc->waitData(CAMERA_MSG_COMPRESSED_IMAGE, MCameraClient::EQ, 1); - c->stopPreview(); - c->disconnect(); - cc->assertNotify(CAMERA_MSG_ERROR, MCameraClient::EQ, 0); - } -}; - -class TestTakeMultiplePictures: public AfterStartPreview { -public: - void run() { - for (int i = 0; i < 10; i++) { - cc->clearStat(); - ASSERT(c->takePicture() == NO_ERROR); - cc->waitNotify(CAMERA_MSG_SHUTTER, MCameraClient::EQ, 1); - cc->waitData(CAMERA_MSG_RAW_IMAGE, MCameraClient::EQ, 1); - cc->waitData(CAMERA_MSG_COMPRESSED_IMAGE, MCameraClient::EQ, 1); - } - c->disconnect(); - cc->assertNotify(CAMERA_MSG_ERROR, MCameraClient::EQ, 0); - } -}; - -class TestGetParameters: public AfterStartPreview { -public: - void run() { - String8 param_str = c->getParameters(); - INFO("%s", static_cast<const char*>(param_str)); - } -}; - -static bool getNextSize(const char **ptrS, int *w, int *h) { - const char *s = *ptrS; - - // skip over ',' - if (*s == ',') s++; - - // remember start position in p - const char *p = s; - while (*s != '\0' && *s != 'x') { - s++; - } - if (*s == '\0') return false; - - // get the width - *w = atoi(p); - - // skip over 'x' - ASSERT(*s == 'x'); - p = s + 1; - while (*s != '\0' && *s != ',') { - s++; - } - - // get the height - *h = atoi(p); - *ptrS = s; - return true; -} - -class TestPictureSize : public AfterStartPreview { -public: - void checkOnePicture(int w, int h) { - const float rate = 0.9; // byte per pixel limit - int pixels = w * h; - - CameraParameters param(c->getParameters()); - param.setPictureSize(w, h); - // disable thumbnail to get more accurate size. - param.set(CameraParameters::KEY_JPEG_THUMBNAIL_WIDTH, 0); - param.set(CameraParameters::KEY_JPEG_THUMBNAIL_HEIGHT, 0); - c->setParameters(param.flatten()); - - cc->clearStat(); - ASSERT(c->takePicture() == NO_ERROR); - cc->waitData(CAMERA_MSG_RAW_IMAGE, MCameraClient::EQ, 1); - //cc->assertDataSize(CAMERA_MSG_RAW_IMAGE, MCameraClient::EQ, pixels*3/2); - cc->waitData(CAMERA_MSG_COMPRESSED_IMAGE, MCameraClient::EQ, 1); - cc->assertDataSize(CAMERA_MSG_COMPRESSED_IMAGE, MCameraClient::LT, - int(pixels * rate)); - cc->assertDataSize(CAMERA_MSG_COMPRESSED_IMAGE, MCameraClient::GT, 0); - cc->assertNotify(CAMERA_MSG_ERROR, MCameraClient::EQ, 0); - } - - void run() { - CameraParameters param(c->getParameters()); - int w, h; - const char *s = param.get(CameraParameters::KEY_SUPPORTED_PICTURE_SIZES); - while (getNextSize(&s, &w, &h)) { - LOGD("checking picture size %dx%d", w, h); - checkOnePicture(w, h); - } - } -}; - -class TestPreviewCallbackFlag : public AfterConnect { -public: - void run() { - sp<MSurface> surface = new MSurface(); - ASSERT(c->setPreviewDisplay(surface) == NO_ERROR); - - // Try all flag combinations. - for (int v = 0; v < 8; v++) { - LOGD("TestPreviewCallbackFlag: flag=%d", v); - usleep(100000); // sleep a while to clear the in-flight callbacks. - cc->clearStat(); - c->setPreviewCallbackFlag(v); - ASSERT(c->previewEnabled() == false); - ASSERT(c->startPreview() == NO_ERROR); - ASSERT(c->previewEnabled() == true); - sleep(2); - c->stopPreview(); - if ((v & FRAME_CALLBACK_FLAG_ENABLE_MASK) == 0) { - cc->assertData(CAMERA_MSG_PREVIEW_FRAME, MCameraClient::EQ, 0); - } else { - if ((v & FRAME_CALLBACK_FLAG_ONE_SHOT_MASK) == 0) { - cc->assertData(CAMERA_MSG_PREVIEW_FRAME, MCameraClient::GE, 10); - } else { - cc->assertData(CAMERA_MSG_PREVIEW_FRAME, MCameraClient::EQ, 1); - } - } - } - } -}; - -class TestRecording : public AfterConnect { -public: - void run() { - ASSERT(c->recordingEnabled() == false); - sp<MSurface> surface = new MSurface(); - ASSERT(c->setPreviewDisplay(surface) == NO_ERROR); - c->setPreviewCallbackFlag(FRAME_CALLBACK_FLAG_ENABLE_MASK); - cc->setReleaser(c.get()); - c->startRecording(); - ASSERT(c->recordingEnabled() == true); - sleep(2); - c->stopRecording(); - usleep(100000); // sleep a while to clear the in-flight callbacks. - cc->setReleaser(NULL); - cc->assertData(CAMERA_MSG_VIDEO_FRAME, MCameraClient::GE, 10); - } -}; - -class TestPreviewSize : public AfterStartPreview { -public: - void checkOnePicture(int w, int h) { - int size = w*h*3/2; // should read from parameters - - c->stopPreview(); - - CameraParameters param(c->getParameters()); - param.setPreviewSize(w, h); - c->setPreviewCallbackFlag(FRAME_CALLBACK_FLAG_ENABLE_MASK); - c->setParameters(param.flatten()); - - c->startPreview(); - - cc->clearStat(); - cc->waitData(CAMERA_MSG_PREVIEW_FRAME, MCameraClient::GE, 1); - cc->assertDataSize(CAMERA_MSG_PREVIEW_FRAME, MCameraClient::EQ, size); - } - - void run() { - CameraParameters param(c->getParameters()); - int w, h; - const char *s = param.get(CameraParameters::KEY_SUPPORTED_PREVIEW_SIZES); - while (getNextSize(&s, &w, &h)) { - LOGD("checking preview size %dx%d", w, h); - checkOnePicture(w, h); - } - } -}; - -void runHolderService() { - defaultServiceManager()->addService( - String16("CameraServiceTest.Holder"), new HolderService()); - ProcessState::self()->startThreadPool(); -} - -int main(int argc, char **argv) -{ - if (argc != 1) { - runFunction(argv[1]); - return 0; - } - INFO("CameraServiceTest start"); - gExecutable = argv[0]; - runHolderService(); - int n = getNumberOfCameras(); - INFO("%d Cameras available", n); - - for (int id = 0; id < n; id++) { - INFO("Testing camera %d", id); - testConnect(id); flushCommands(); - testAllowConnectOnceOnly(id); flushCommands(); - testReconnect(id); flushCommands(); - testLockUnlock(id); flushCommands(); - testReconnectFromAnotherProcess(id); flushCommands(); - - RUN(TestSetPreviewDisplay, id); - RUN(TestStartPreview, id); - RUN(TestStartPreviewWithoutDisplay, id); - RUN(TestAutoFocus, id); - RUN(TestStopPreview, id); - RUN(TestTakePicture, id); - RUN(TestTakeMultiplePictures, id); - RUN(TestGetParameters, id); - RUN(TestPictureSize, id); - RUN(TestPreviewCallbackFlag, id); - RUN(TestRecording, id); - RUN(TestPreviewSize, id); - } - - INFO("CameraServiceTest finished"); -} diff --git a/cmds/surfaceflinger/Android.mk b/cmds/surfaceflinger/Android.mk index bfa58a1..1df32bb 100644 --- a/cmds/surfaceflinger/Android.mk +++ b/cmds/surfaceflinger/Android.mk @@ -10,7 +10,7 @@ LOCAL_SHARED_LIBRARIES := \ libutils LOCAL_C_INCLUDES := \ - $(LOCAL_PATH)/../../libs/surfaceflinger + $(LOCAL_PATH)/../../services/surfaceflinger LOCAL_MODULE:= surfaceflinger diff --git a/libs/audioflinger/A2dpAudioInterface.cpp b/libs/audioflinger/A2dpAudioInterface.cpp deleted file mode 100644 index 995e31c..0000000 --- a/libs/audioflinger/A2dpAudioInterface.cpp +++ /dev/null @@ -1,466 +0,0 @@ -/* - * Copyright (C) 2008 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#include <math.h> - -//#define LOG_NDEBUG 0 -#define LOG_TAG "A2dpAudioInterface" -#include <utils/Log.h> -#include <utils/String8.h> - -#include "A2dpAudioInterface.h" -#include "audio/liba2dp.h" - - -namespace android { - -// ---------------------------------------------------------------------------- - -//AudioHardwareInterface* A2dpAudioInterface::createA2dpInterface() -//{ -// AudioHardwareInterface* hw = 0; -// -// hw = AudioHardwareInterface::create(); -// LOGD("new A2dpAudioInterface(hw: %p)", hw); -// hw = new A2dpAudioInterface(hw); -// return hw; -//} - -A2dpAudioInterface::A2dpAudioInterface(AudioHardwareInterface* hw) : - mOutput(0), mHardwareInterface(hw), mBluetoothEnabled(true), mSuspended(false) -{ -} - -A2dpAudioInterface::~A2dpAudioInterface() -{ - closeOutputStream((AudioStreamOut *)mOutput); - delete mHardwareInterface; -} - -status_t A2dpAudioInterface::initCheck() -{ - if (mHardwareInterface == 0) return NO_INIT; - return mHardwareInterface->initCheck(); -} - -AudioStreamOut* A2dpAudioInterface::openOutputStream( - uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) -{ - if (!AudioSystem::isA2dpDevice((AudioSystem::audio_devices)devices)) { - LOGV("A2dpAudioInterface::openOutputStream() open HW device: %x", devices); - return mHardwareInterface->openOutputStream(devices, format, channels, sampleRate, status); - } - - status_t err = 0; - - // only one output stream allowed - if (mOutput) { - if (status) - *status = -1; - return NULL; - } - - // create new output stream - A2dpAudioStreamOut* out = new A2dpAudioStreamOut(); - if ((err = out->set(devices, format, channels, sampleRate)) == NO_ERROR) { - mOutput = out; - mOutput->setBluetoothEnabled(mBluetoothEnabled); - mOutput->setSuspended(mSuspended); - } else { - delete out; - } - - if (status) - *status = err; - return mOutput; -} - -void A2dpAudioInterface::closeOutputStream(AudioStreamOut* out) { - if (mOutput == 0 || mOutput != out) { - mHardwareInterface->closeOutputStream(out); - } - else { - delete mOutput; - mOutput = 0; - } -} - - -AudioStreamIn* A2dpAudioInterface::openInputStream( - uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status, - AudioSystem::audio_in_acoustics acoustics) -{ - return mHardwareInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics); -} - -void A2dpAudioInterface::closeInputStream(AudioStreamIn* in) -{ - return mHardwareInterface->closeInputStream(in); -} - -status_t A2dpAudioInterface::setMode(int mode) -{ - return mHardwareInterface->setMode(mode); -} - -status_t A2dpAudioInterface::setMicMute(bool state) -{ - return mHardwareInterface->setMicMute(state); -} - -status_t A2dpAudioInterface::getMicMute(bool* state) -{ - return mHardwareInterface->getMicMute(state); -} - -status_t A2dpAudioInterface::setParameters(const String8& keyValuePairs) -{ - AudioParameter param = AudioParameter(keyValuePairs); - String8 value; - String8 key; - status_t status = NO_ERROR; - - LOGV("setParameters() %s", keyValuePairs.string()); - - key = "bluetooth_enabled"; - if (param.get(key, value) == NO_ERROR) { - mBluetoothEnabled = (value == "true"); - if (mOutput) { - mOutput->setBluetoothEnabled(mBluetoothEnabled); - } - param.remove(key); - } - key = String8("A2dpSuspended"); - if (param.get(key, value) == NO_ERROR) { - mSuspended = (value == "true"); - if (mOutput) { - mOutput->setSuspended(mSuspended); - } - param.remove(key); - } - - if (param.size()) { - status_t hwStatus = mHardwareInterface->setParameters(param.toString()); - if (status == NO_ERROR) { - status = hwStatus; - } - } - - return status; -} - -String8 A2dpAudioInterface::getParameters(const String8& keys) -{ - AudioParameter param = AudioParameter(keys); - AudioParameter a2dpParam = AudioParameter(); - String8 value; - String8 key; - - key = "bluetooth_enabled"; - if (param.get(key, value) == NO_ERROR) { - value = mBluetoothEnabled ? "true" : "false"; - a2dpParam.add(key, value); - param.remove(key); - } - key = "A2dpSuspended"; - if (param.get(key, value) == NO_ERROR) { - value = mSuspended ? "true" : "false"; - a2dpParam.add(key, value); - param.remove(key); - } - - String8 keyValuePairs = a2dpParam.toString(); - - if (param.size()) { - if (keyValuePairs != "") { - keyValuePairs += ";"; - } - keyValuePairs += mHardwareInterface->getParameters(param.toString()); - } - - LOGV("getParameters() %s", keyValuePairs.string()); - return keyValuePairs; -} - -size_t A2dpAudioInterface::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) -{ - return mHardwareInterface->getInputBufferSize(sampleRate, format, channelCount); -} - -status_t A2dpAudioInterface::setVoiceVolume(float v) -{ - return mHardwareInterface->setVoiceVolume(v); -} - -status_t A2dpAudioInterface::setMasterVolume(float v) -{ - return mHardwareInterface->setMasterVolume(v); -} - -status_t A2dpAudioInterface::dump(int fd, const Vector<String16>& args) -{ - return mHardwareInterface->dumpState(fd, args); -} - -// ---------------------------------------------------------------------------- - -A2dpAudioInterface::A2dpAudioStreamOut::A2dpAudioStreamOut() : - mFd(-1), mStandby(true), mStartCount(0), mRetryCount(0), mData(NULL), - // assume BT enabled to start, this is safe because its only the - // enabled->disabled transition we are worried about - mBluetoothEnabled(true), mDevice(0), mClosing(false), mSuspended(false) -{ - // use any address by default - strcpy(mA2dpAddress, "00:00:00:00:00:00"); - init(); -} - -status_t A2dpAudioInterface::A2dpAudioStreamOut::set( - uint32_t device, int *pFormat, uint32_t *pChannels, uint32_t *pRate) -{ - int lFormat = pFormat ? *pFormat : 0; - uint32_t lChannels = pChannels ? *pChannels : 0; - uint32_t lRate = pRate ? *pRate : 0; - - LOGD("A2dpAudioStreamOut::set %x, %d, %d, %d\n", device, lFormat, lChannels, lRate); - - // fix up defaults - if (lFormat == 0) lFormat = format(); - if (lChannels == 0) lChannels = channels(); - if (lRate == 0) lRate = sampleRate(); - - // check values - if ((lFormat != format()) || - (lChannels != channels()) || - (lRate != sampleRate())){ - if (pFormat) *pFormat = format(); - if (pChannels) *pChannels = channels(); - if (pRate) *pRate = sampleRate(); - return BAD_VALUE; - } - - if (pFormat) *pFormat = lFormat; - if (pChannels) *pChannels = lChannels; - if (pRate) *pRate = lRate; - - mDevice = device; - return NO_ERROR; -} - -A2dpAudioInterface::A2dpAudioStreamOut::~A2dpAudioStreamOut() -{ - LOGV("A2dpAudioStreamOut destructor"); - standby(); - close(); - LOGV("A2dpAudioStreamOut destructor returning from close()"); -} - -ssize_t A2dpAudioInterface::A2dpAudioStreamOut::write(const void* buffer, size_t bytes) -{ - Mutex::Autolock lock(mLock); - - size_t remaining = bytes; - status_t status = -1; - - if (!mBluetoothEnabled || mClosing || mSuspended) { - LOGV("A2dpAudioStreamOut::write(), but bluetooth disabled \ - mBluetoothEnabled %d, mClosing %d, mSuspended %d", - mBluetoothEnabled, mClosing, mSuspended); - goto Error; - } - - status = init(); - if (status < 0) - goto Error; - - while (remaining > 0) { - status = a2dp_write(mData, buffer, remaining); - if (status <= 0) { - LOGE("a2dp_write failed err: %d\n", status); - goto Error; - } - remaining -= status; - buffer = ((char *)buffer) + status; - } - - mStandby = false; - - return bytes; - -Error: - // Simulate audio output timing in case of error - usleep(((bytes * 1000 )/ frameSize() / sampleRate()) * 1000); - - return status; -} - -status_t A2dpAudioInterface::A2dpAudioStreamOut::init() -{ - if (!mData) { - status_t status = a2dp_init(44100, 2, &mData); - if (status < 0) { - LOGE("a2dp_init failed err: %d\n", status); - mData = NULL; - return status; - } - a2dp_set_sink(mData, mA2dpAddress); - } - - return 0; -} - -status_t A2dpAudioInterface::A2dpAudioStreamOut::standby() -{ - int result = 0; - - if (mClosing) { - LOGV("Ignore standby, closing"); - return result; - } - - Mutex::Autolock lock(mLock); - - if (!mStandby) { - result = a2dp_stop(mData); - if (result == 0) - mStandby = true; - } - - return result; -} - -status_t A2dpAudioInterface::A2dpAudioStreamOut::setParameters(const String8& keyValuePairs) -{ - AudioParameter param = AudioParameter(keyValuePairs); - String8 value; - String8 key = String8("a2dp_sink_address"); - status_t status = NO_ERROR; - int device; - LOGV("A2dpAudioStreamOut::setParameters() %s", keyValuePairs.string()); - - if (param.get(key, value) == NO_ERROR) { - if (value.length() != strlen("00:00:00:00:00:00")) { - status = BAD_VALUE; - } else { - setAddress(value.string()); - } - param.remove(key); - } - key = String8("closing"); - if (param.get(key, value) == NO_ERROR) { - mClosing = (value == "true"); - param.remove(key); - } - key = AudioParameter::keyRouting; - if (param.getInt(key, device) == NO_ERROR) { - if (AudioSystem::isA2dpDevice((AudioSystem::audio_devices)device)) { - mDevice = device; - status = NO_ERROR; - } else { - status = BAD_VALUE; - } - param.remove(key); - } - - if (param.size()) { - status = BAD_VALUE; - } - return status; -} - -String8 A2dpAudioInterface::A2dpAudioStreamOut::getParameters(const String8& keys) -{ - AudioParameter param = AudioParameter(keys); - String8 value; - String8 key = String8("a2dp_sink_address"); - - if (param.get(key, value) == NO_ERROR) { - value = mA2dpAddress; - param.add(key, value); - } - key = AudioParameter::keyRouting; - if (param.get(key, value) == NO_ERROR) { - param.addInt(key, (int)mDevice); - } - - LOGV("A2dpAudioStreamOut::getParameters() %s", param.toString().string()); - return param.toString(); -} - -status_t A2dpAudioInterface::A2dpAudioStreamOut::setAddress(const char* address) -{ - Mutex::Autolock lock(mLock); - - if (strlen(address) != strlen("00:00:00:00:00:00")) - return -EINVAL; - - strcpy(mA2dpAddress, address); - if (mData) - a2dp_set_sink(mData, mA2dpAddress); - - return NO_ERROR; -} - -status_t A2dpAudioInterface::A2dpAudioStreamOut::setBluetoothEnabled(bool enabled) -{ - LOGD("setBluetoothEnabled %d", enabled); - - Mutex::Autolock lock(mLock); - - mBluetoothEnabled = enabled; - if (!enabled) { - return close_l(); - } - return NO_ERROR; -} - -status_t A2dpAudioInterface::A2dpAudioStreamOut::setSuspended(bool onOff) -{ - LOGV("setSuspended %d", onOff); - mSuspended = onOff; - standby(); - return NO_ERROR; -} - -status_t A2dpAudioInterface::A2dpAudioStreamOut::close() -{ - Mutex::Autolock lock(mLock); - LOGV("A2dpAudioStreamOut::close() calling close_l()"); - return close_l(); -} - -status_t A2dpAudioInterface::A2dpAudioStreamOut::close_l() -{ - if (mData) { - LOGV("A2dpAudioStreamOut::close_l() calling a2dp_cleanup(mData)"); - a2dp_cleanup(mData); - mData = NULL; - } - return NO_ERROR; -} - -status_t A2dpAudioInterface::A2dpAudioStreamOut::dump(int fd, const Vector<String16>& args) -{ - return NO_ERROR; -} - -status_t A2dpAudioInterface::A2dpAudioStreamOut::getRenderPosition(uint32_t *driverFrames) -{ - //TODO: enable when supported by driver - return INVALID_OPERATION; -} - -}; // namespace android diff --git a/libs/audioflinger/A2dpAudioInterface.h b/libs/audioflinger/A2dpAudioInterface.h deleted file mode 100644 index 48154f9..0000000 --- a/libs/audioflinger/A2dpAudioInterface.h +++ /dev/null @@ -1,135 +0,0 @@ -/* - * Copyright (C) 2008 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#ifndef A2DP_AUDIO_HARDWARE_H -#define A2DP_AUDIO_HARDWARE_H - -#include <stdint.h> -#include <sys/types.h> - -#include <utils/threads.h> - -#include <hardware_legacy/AudioHardwareBase.h> - - -namespace android { - -class A2dpAudioInterface : public AudioHardwareBase -{ - class A2dpAudioStreamOut; - -public: - A2dpAudioInterface(AudioHardwareInterface* hw); - virtual ~A2dpAudioInterface(); - virtual status_t initCheck(); - - virtual status_t setVoiceVolume(float volume); - virtual status_t setMasterVolume(float volume); - - virtual status_t setMode(int mode); - - // mic mute - virtual status_t setMicMute(bool state); - virtual status_t getMicMute(bool* state); - - virtual status_t setParameters(const String8& keyValuePairs); - virtual String8 getParameters(const String8& keys); - - virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount); - - // create I/O streams - virtual AudioStreamOut* openOutputStream( - uint32_t devices, - int *format=0, - uint32_t *channels=0, - uint32_t *sampleRate=0, - status_t *status=0); - virtual void closeOutputStream(AudioStreamOut* out); - - virtual AudioStreamIn* openInputStream( - uint32_t devices, - int *format, - uint32_t *channels, - uint32_t *sampleRate, - status_t *status, - AudioSystem::audio_in_acoustics acoustics); - virtual void closeInputStream(AudioStreamIn* in); -// static AudioHardwareInterface* createA2dpInterface(); - -protected: - virtual status_t dump(int fd, const Vector<String16>& args); - -private: - class A2dpAudioStreamOut : public AudioStreamOut { - public: - A2dpAudioStreamOut(); - virtual ~A2dpAudioStreamOut(); - status_t set(uint32_t device, - int *pFormat, - uint32_t *pChannels, - uint32_t *pRate); - virtual uint32_t sampleRate() const { return 44100; } - // SBC codec wants a multiple of 512 - virtual size_t bufferSize() const { return 512 * 20; } - virtual uint32_t channels() const { return AudioSystem::CHANNEL_OUT_STEREO; } - virtual int format() const { return AudioSystem::PCM_16_BIT; } - virtual uint32_t latency() const { return ((1000*bufferSize())/frameSize())/sampleRate() + 200; } - virtual status_t setVolume(float left, float right) { return INVALID_OPERATION; } - virtual ssize_t write(const void* buffer, size_t bytes); - status_t standby(); - virtual status_t dump(int fd, const Vector<String16>& args); - virtual status_t setParameters(const String8& keyValuePairs); - virtual String8 getParameters(const String8& keys); - virtual status_t getRenderPosition(uint32_t *dspFrames); - - private: - friend class A2dpAudioInterface; - status_t init(); - status_t close(); - status_t close_l(); - status_t setAddress(const char* address); - status_t setBluetoothEnabled(bool enabled); - status_t setSuspended(bool onOff); - - private: - int mFd; - bool mStandby; - int mStartCount; - int mRetryCount; - char mA2dpAddress[20]; - void* mData; - Mutex mLock; - bool mBluetoothEnabled; - uint32_t mDevice; - bool mClosing; - bool mSuspended; - }; - - friend class A2dpAudioStreamOut; - - A2dpAudioStreamOut* mOutput; - AudioHardwareInterface *mHardwareInterface; - char mA2dpAddress[20]; - bool mBluetoothEnabled; - bool mSuspended; -}; - - -// ---------------------------------------------------------------------------- - -}; // namespace android - -#endif // A2DP_AUDIO_HARDWARE_H diff --git a/libs/audioflinger/Android.mk b/libs/audioflinger/Android.mk deleted file mode 100644 index 22ecc54..0000000 --- a/libs/audioflinger/Android.mk +++ /dev/null @@ -1,131 +0,0 @@ -LOCAL_PATH:= $(call my-dir) - -#AUDIO_POLICY_TEST := true -#ENABLE_AUDIO_DUMP := true - -include $(CLEAR_VARS) - - -ifeq ($(AUDIO_POLICY_TEST),true) - ENABLE_AUDIO_DUMP := true -endif - - -LOCAL_SRC_FILES:= \ - AudioHardwareGeneric.cpp \ - AudioHardwareStub.cpp \ - AudioHardwareInterface.cpp - -ifeq ($(ENABLE_AUDIO_DUMP),true) - LOCAL_SRC_FILES += AudioDumpInterface.cpp - LOCAL_CFLAGS += -DENABLE_AUDIO_DUMP -endif - -LOCAL_SHARED_LIBRARIES := \ - libcutils \ - libutils \ - libbinder \ - libmedia \ - libhardware_legacy - -ifeq ($(strip $(BOARD_USES_GENERIC_AUDIO)),true) - LOCAL_CFLAGS += -DGENERIC_AUDIO -endif - -LOCAL_MODULE:= libaudiointerface - -ifeq ($(BOARD_HAVE_BLUETOOTH),true) - LOCAL_SRC_FILES += A2dpAudioInterface.cpp - LOCAL_SHARED_LIBRARIES += liba2dp - LOCAL_CFLAGS += -DWITH_BLUETOOTH -DWITH_A2DP - LOCAL_C_INCLUDES += $(call include-path-for, bluez) -endif - -include $(BUILD_STATIC_LIBRARY) - - -include $(CLEAR_VARS) - -LOCAL_SRC_FILES:= \ - AudioPolicyManagerBase.cpp - -LOCAL_SHARED_LIBRARIES := \ - libcutils \ - libutils \ - libmedia - -ifeq ($(TARGET_SIMULATOR),true) - LOCAL_LDLIBS += -ldl -else - LOCAL_SHARED_LIBRARIES += libdl -endif - -LOCAL_MODULE:= libaudiopolicybase - -ifeq ($(BOARD_HAVE_BLUETOOTH),true) - LOCAL_CFLAGS += -DWITH_A2DP -endif - -ifeq ($(AUDIO_POLICY_TEST),true) - LOCAL_CFLAGS += -DAUDIO_POLICY_TEST -endif - -include $(BUILD_STATIC_LIBRARY) - -include $(CLEAR_VARS) - -LOCAL_SRC_FILES:= \ - AudioFlinger.cpp \ - AudioMixer.cpp.arm \ - AudioResampler.cpp.arm \ - AudioResamplerSinc.cpp.arm \ - AudioResamplerCubic.cpp.arm \ - AudioPolicyService.cpp - -LOCAL_SHARED_LIBRARIES := \ - libcutils \ - libutils \ - libbinder \ - libmedia \ - libhardware_legacy \ - libeffects - -ifeq ($(strip $(BOARD_USES_GENERIC_AUDIO)),true) - LOCAL_STATIC_LIBRARIES += libaudiointerface libaudiopolicybase - LOCAL_CFLAGS += -DGENERIC_AUDIO -else - LOCAL_SHARED_LIBRARIES += libaudio libaudiopolicy -endif - -ifeq ($(TARGET_SIMULATOR),true) - LOCAL_LDLIBS += -ldl -else - LOCAL_SHARED_LIBRARIES += libdl -endif - -LOCAL_MODULE:= libaudioflinger - -ifeq ($(BOARD_HAVE_BLUETOOTH),true) - LOCAL_CFLAGS += -DWITH_BLUETOOTH -DWITH_A2DP - LOCAL_SHARED_LIBRARIES += liba2dp -endif - -ifeq ($(AUDIO_POLICY_TEST),true) - LOCAL_CFLAGS += -DAUDIO_POLICY_TEST -endif - -ifeq ($(TARGET_SIMULATOR),true) - ifeq ($(HOST_OS),linux) - LOCAL_LDLIBS += -lrt -lpthread - endif -endif - -ifeq ($(BOARD_USE_LVMX),true) - LOCAL_CFLAGS += -DLVMX - LOCAL_C_INCLUDES += vendor/nxp - LOCAL_STATIC_LIBRARIES += liblifevibes - LOCAL_SHARED_LIBRARIES += liblvmxservice -# LOCAL_SHARED_LIBRARIES += liblvmxipc -endif - -include $(BUILD_SHARED_LIBRARY) diff --git a/libs/audioflinger/AudioBufferProvider.h b/libs/audioflinger/AudioBufferProvider.h deleted file mode 100644 index 81c5c39..0000000 --- a/libs/audioflinger/AudioBufferProvider.h +++ /dev/null @@ -1,49 +0,0 @@ -/* - * Copyright (C) 2007 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#ifndef ANDROID_AUDIO_BUFFER_PROVIDER_H -#define ANDROID_AUDIO_BUFFER_PROVIDER_H - -#include <stdint.h> -#include <sys/types.h> -#include <utils/Errors.h> - -namespace android { -// ---------------------------------------------------------------------------- - -class AudioBufferProvider -{ -public: - - struct Buffer { - union { - void* raw; - short* i16; - int8_t* i8; - }; - size_t frameCount; - }; - - virtual ~AudioBufferProvider() {} - - virtual status_t getNextBuffer(Buffer* buffer) = 0; - virtual void releaseBuffer(Buffer* buffer) = 0; -}; - -// ---------------------------------------------------------------------------- -}; // namespace android - -#endif // ANDROID_AUDIO_BUFFER_PROVIDER_H diff --git a/libs/audioflinger/AudioDumpInterface.cpp b/libs/audioflinger/AudioDumpInterface.cpp deleted file mode 100644 index 6c11114..0000000 --- a/libs/audioflinger/AudioDumpInterface.cpp +++ /dev/null @@ -1,573 +0,0 @@ -/* //device/servers/AudioFlinger/AudioDumpInterface.cpp -** -** Copyright 2008, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#define LOG_TAG "AudioFlingerDump" -//#define LOG_NDEBUG 0 - -#include <stdint.h> -#include <sys/types.h> -#include <utils/Log.h> - -#include <stdlib.h> -#include <unistd.h> - -#include "AudioDumpInterface.h" - -namespace android { - -// ---------------------------------------------------------------------------- - -AudioDumpInterface::AudioDumpInterface(AudioHardwareInterface* hw) - : mPolicyCommands(String8("")), mFileName(String8("")) -{ - if(hw == 0) { - LOGE("Dump construct hw = 0"); - } - mFinalInterface = hw; - LOGV("Constructor %p, mFinalInterface %p", this, mFinalInterface); -} - - -AudioDumpInterface::~AudioDumpInterface() -{ - for (size_t i = 0; i < mOutputs.size(); i++) { - closeOutputStream((AudioStreamOut *)mOutputs[i]); - } - - for (size_t i = 0; i < mInputs.size(); i++) { - closeInputStream((AudioStreamIn *)mInputs[i]); - } - - if(mFinalInterface) delete mFinalInterface; -} - - -AudioStreamOut* AudioDumpInterface::openOutputStream( - uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) -{ - AudioStreamOut* outFinal = NULL; - int lFormat = AudioSystem::PCM_16_BIT; - uint32_t lChannels = AudioSystem::CHANNEL_OUT_STEREO; - uint32_t lRate = 44100; - - - outFinal = mFinalInterface->openOutputStream(devices, format, channels, sampleRate, status); - if (outFinal != 0) { - lFormat = outFinal->format(); - lChannels = outFinal->channels(); - lRate = outFinal->sampleRate(); - } else { - if (format != 0) { - if (*format != 0) { - lFormat = *format; - } else { - *format = lFormat; - } - } - if (channels != 0) { - if (*channels != 0) { - lChannels = *channels; - } else { - *channels = lChannels; - } - } - if (sampleRate != 0) { - if (*sampleRate != 0) { - lRate = *sampleRate; - } else { - *sampleRate = lRate; - } - } - if (status) *status = NO_ERROR; - } - LOGV("openOutputStream(), outFinal %p", outFinal); - - AudioStreamOutDump *dumOutput = new AudioStreamOutDump(this, mOutputs.size(), outFinal, - devices, lFormat, lChannels, lRate); - mOutputs.add(dumOutput); - - return dumOutput; -} - -void AudioDumpInterface::closeOutputStream(AudioStreamOut* out) -{ - AudioStreamOutDump *dumpOut = (AudioStreamOutDump *)out; - - if (mOutputs.indexOf(dumpOut) < 0) { - LOGW("Attempt to close invalid output stream"); - return; - } - - LOGV("closeOutputStream() output %p", out); - - dumpOut->standby(); - if (dumpOut->finalStream() != NULL) { - mFinalInterface->closeOutputStream(dumpOut->finalStream()); - } - - mOutputs.remove(dumpOut); - delete dumpOut; -} - -AudioStreamIn* AudioDumpInterface::openInputStream(uint32_t devices, int *format, uint32_t *channels, - uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) -{ - AudioStreamIn* inFinal = NULL; - int lFormat = AudioSystem::PCM_16_BIT; - uint32_t lChannels = AudioSystem::CHANNEL_IN_MONO; - uint32_t lRate = 8000; - - inFinal = mFinalInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics); - if (inFinal != 0) { - lFormat = inFinal->format(); - lChannels = inFinal->channels(); - lRate = inFinal->sampleRate(); - } else { - if (format != 0) { - if (*format != 0) { - lFormat = *format; - } else { - *format = lFormat; - } - } - if (channels != 0) { - if (*channels != 0) { - lChannels = *channels; - } else { - *channels = lChannels; - } - } - if (sampleRate != 0) { - if (*sampleRate != 0) { - lRate = *sampleRate; - } else { - *sampleRate = lRate; - } - } - if (status) *status = NO_ERROR; - } - LOGV("openInputStream(), inFinal %p", inFinal); - - AudioStreamInDump *dumInput = new AudioStreamInDump(this, mInputs.size(), inFinal, - devices, lFormat, lChannels, lRate); - mInputs.add(dumInput); - - return dumInput; -} -void AudioDumpInterface::closeInputStream(AudioStreamIn* in) -{ - AudioStreamInDump *dumpIn = (AudioStreamInDump *)in; - - if (mInputs.indexOf(dumpIn) < 0) { - LOGW("Attempt to close invalid input stream"); - return; - } - dumpIn->standby(); - if (dumpIn->finalStream() != NULL) { - mFinalInterface->closeInputStream(dumpIn->finalStream()); - } - - mInputs.remove(dumpIn); - delete dumpIn; -} - - -status_t AudioDumpInterface::setParameters(const String8& keyValuePairs) -{ - AudioParameter param = AudioParameter(keyValuePairs); - String8 value; - int valueInt; - LOGV("setParameters %s", keyValuePairs.string()); - - if (param.get(String8("test_cmd_file_name"), value) == NO_ERROR) { - mFileName = value; - param.remove(String8("test_cmd_file_name")); - } - if (param.get(String8("test_cmd_policy"), value) == NO_ERROR) { - Mutex::Autolock _l(mLock); - param.remove(String8("test_cmd_policy")); - mPolicyCommands = param.toString(); - LOGV("test_cmd_policy command %s written", mPolicyCommands.string()); - return NO_ERROR; - } - - if (mFinalInterface != 0 ) return mFinalInterface->setParameters(keyValuePairs); - return NO_ERROR; -} - -String8 AudioDumpInterface::getParameters(const String8& keys) -{ - AudioParameter param = AudioParameter(keys); - AudioParameter response; - String8 value; - -// LOGV("getParameters %s", keys.string()); - if (param.get(String8("test_cmd_policy"), value) == NO_ERROR) { - Mutex::Autolock _l(mLock); - if (mPolicyCommands.length() != 0) { - response = AudioParameter(mPolicyCommands); - response.addInt(String8("test_cmd_policy"), 1); - } else { - response.addInt(String8("test_cmd_policy"), 0); - } - param.remove(String8("test_cmd_policy")); -// LOGV("test_cmd_policy command %s read", mPolicyCommands.string()); - } - - if (param.get(String8("test_cmd_file_name"), value) == NO_ERROR) { - response.add(String8("test_cmd_file_name"), mFileName); - param.remove(String8("test_cmd_file_name")); - } - - String8 keyValuePairs = response.toString(); - - if (param.size() && mFinalInterface != 0 ) { - keyValuePairs += ";"; - keyValuePairs += mFinalInterface->getParameters(param.toString()); - } - - return keyValuePairs; -} - -status_t AudioDumpInterface::setMode(int mode) -{ - return mFinalInterface->setMode(mode); -} - -size_t AudioDumpInterface::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) -{ - return mFinalInterface->getInputBufferSize(sampleRate, format, channelCount); -} - -// ---------------------------------------------------------------------------- - -AudioStreamOutDump::AudioStreamOutDump(AudioDumpInterface *interface, - int id, - AudioStreamOut* finalStream, - uint32_t devices, - int format, - uint32_t channels, - uint32_t sampleRate) - : mInterface(interface), mId(id), - mSampleRate(sampleRate), mFormat(format), mChannels(channels), mLatency(0), mDevice(devices), - mBufferSize(1024), mFinalStream(finalStream), mFile(0), mFileCount(0) -{ - LOGV("AudioStreamOutDump Constructor %p, mInterface %p, mFinalStream %p", this, mInterface, mFinalStream); -} - - -AudioStreamOutDump::~AudioStreamOutDump() -{ - LOGV("AudioStreamOutDump destructor"); - Close(); -} - -ssize_t AudioStreamOutDump::write(const void* buffer, size_t bytes) -{ - ssize_t ret; - - if (mFinalStream) { - ret = mFinalStream->write(buffer, bytes); - } else { - usleep((((bytes * 1000) / frameSize()) / sampleRate()) * 1000); - ret = bytes; - } - if(!mFile) { - if (mInterface->fileName() != "") { - char name[255]; - sprintf(name, "%s_out_%d_%d.pcm", mInterface->fileName().string(), mId, ++mFileCount); - mFile = fopen(name, "wb"); - LOGV("Opening dump file %s, fh %p", name, mFile); - } - } - if (mFile) { - fwrite(buffer, bytes, 1, mFile); - } - return ret; -} - -status_t AudioStreamOutDump::standby() -{ - LOGV("AudioStreamOutDump standby(), mFile %p, mFinalStream %p", mFile, mFinalStream); - - Close(); - if (mFinalStream != 0 ) return mFinalStream->standby(); - return NO_ERROR; -} - -uint32_t AudioStreamOutDump::sampleRate() const -{ - if (mFinalStream != 0 ) return mFinalStream->sampleRate(); - return mSampleRate; -} - -size_t AudioStreamOutDump::bufferSize() const -{ - if (mFinalStream != 0 ) return mFinalStream->bufferSize(); - return mBufferSize; -} - -uint32_t AudioStreamOutDump::channels() const -{ - if (mFinalStream != 0 ) return mFinalStream->channels(); - return mChannels; -} -int AudioStreamOutDump::format() const -{ - if (mFinalStream != 0 ) return mFinalStream->format(); - return mFormat; -} -uint32_t AudioStreamOutDump::latency() const -{ - if (mFinalStream != 0 ) return mFinalStream->latency(); - return 0; -} -status_t AudioStreamOutDump::setVolume(float left, float right) -{ - if (mFinalStream != 0 ) return mFinalStream->setVolume(left, right); - return NO_ERROR; -} -status_t AudioStreamOutDump::setParameters(const String8& keyValuePairs) -{ - LOGV("AudioStreamOutDump::setParameters %s", keyValuePairs.string()); - - if (mFinalStream != 0 ) { - return mFinalStream->setParameters(keyValuePairs); - } - - AudioParameter param = AudioParameter(keyValuePairs); - String8 value; - int valueInt; - status_t status = NO_ERROR; - - if (param.getInt(String8("set_id"), valueInt) == NO_ERROR) { - mId = valueInt; - } - - if (param.getInt(String8("format"), valueInt) == NO_ERROR) { - if (mFile == 0) { - mFormat = valueInt; - } else { - status = INVALID_OPERATION; - } - } - if (param.getInt(String8("channels"), valueInt) == NO_ERROR) { - if (valueInt == AudioSystem::CHANNEL_OUT_STEREO || valueInt == AudioSystem::CHANNEL_OUT_MONO) { - mChannels = valueInt; - } else { - status = BAD_VALUE; - } - } - if (param.getInt(String8("sampling_rate"), valueInt) == NO_ERROR) { - if (valueInt > 0 && valueInt <= 48000) { - if (mFile == 0) { - mSampleRate = valueInt; - } else { - status = INVALID_OPERATION; - } - } else { - status = BAD_VALUE; - } - } - return status; -} - -String8 AudioStreamOutDump::getParameters(const String8& keys) -{ - if (mFinalStream != 0 ) return mFinalStream->getParameters(keys); - - AudioParameter param = AudioParameter(keys); - return param.toString(); -} - -status_t AudioStreamOutDump::dump(int fd, const Vector<String16>& args) -{ - if (mFinalStream != 0 ) return mFinalStream->dump(fd, args); - return NO_ERROR; -} - -void AudioStreamOutDump::Close() -{ - if(mFile) { - fclose(mFile); - mFile = 0; - } -} - -status_t AudioStreamOutDump::getRenderPosition(uint32_t *dspFrames) -{ - if (mFinalStream != 0 ) return mFinalStream->getRenderPosition(dspFrames); - return INVALID_OPERATION; -} - -// ---------------------------------------------------------------------------- - -AudioStreamInDump::AudioStreamInDump(AudioDumpInterface *interface, - int id, - AudioStreamIn* finalStream, - uint32_t devices, - int format, - uint32_t channels, - uint32_t sampleRate) - : mInterface(interface), mId(id), - mSampleRate(sampleRate), mFormat(format), mChannels(channels), mDevice(devices), - mBufferSize(1024), mFinalStream(finalStream), mFile(0), mFileCount(0) -{ - LOGV("AudioStreamInDump Constructor %p, mInterface %p, mFinalStream %p", this, mInterface, mFinalStream); -} - - -AudioStreamInDump::~AudioStreamInDump() -{ - Close(); -} - -ssize_t AudioStreamInDump::read(void* buffer, ssize_t bytes) -{ - ssize_t ret; - - if (mFinalStream) { - ret = mFinalStream->read(buffer, bytes); - if(!mFile) { - if (mInterface->fileName() != "") { - char name[255]; - sprintf(name, "%s_in_%d_%d.pcm", mInterface->fileName().string(), mId, ++mFileCount); - mFile = fopen(name, "wb"); - LOGV("Opening input dump file %s, fh %p", name, mFile); - } - } - if (mFile) { - fwrite(buffer, bytes, 1, mFile); - } - } else { - usleep((((bytes * 1000) / frameSize()) / sampleRate()) * 1000); - ret = bytes; - if(!mFile) { - char name[255]; - strcpy(name, "/sdcard/music/sine440"); - if (channels() == AudioSystem::CHANNEL_IN_MONO) { - strcat(name, "_mo"); - } else { - strcat(name, "_st"); - } - if (format() == AudioSystem::PCM_16_BIT) { - strcat(name, "_16b"); - } else { - strcat(name, "_8b"); - } - if (sampleRate() < 16000) { - strcat(name, "_8k"); - } else if (sampleRate() < 32000) { - strcat(name, "_22k"); - } else if (sampleRate() < 48000) { - strcat(name, "_44k"); - } else { - strcat(name, "_48k"); - } - strcat(name, ".wav"); - mFile = fopen(name, "rb"); - LOGV("Opening input read file %s, fh %p", name, mFile); - if (mFile) { - fseek(mFile, AUDIO_DUMP_WAVE_HDR_SIZE, SEEK_SET); - } - } - if (mFile) { - ssize_t bytesRead = fread(buffer, bytes, 1, mFile); - if (bytesRead >=0 && bytesRead < bytes) { - fseek(mFile, AUDIO_DUMP_WAVE_HDR_SIZE, SEEK_SET); - fread((uint8_t *)buffer+bytesRead, bytes-bytesRead, 1, mFile); - } - } - } - - return ret; -} - -status_t AudioStreamInDump::standby() -{ - LOGV("AudioStreamInDump standby(), mFile %p, mFinalStream %p", mFile, mFinalStream); - - Close(); - if (mFinalStream != 0 ) return mFinalStream->standby(); - return NO_ERROR; -} - -status_t AudioStreamInDump::setGain(float gain) -{ - if (mFinalStream != 0 ) return mFinalStream->setGain(gain); - return NO_ERROR; -} - -uint32_t AudioStreamInDump::sampleRate() const -{ - if (mFinalStream != 0 ) return mFinalStream->sampleRate(); - return mSampleRate; -} - -size_t AudioStreamInDump::bufferSize() const -{ - if (mFinalStream != 0 ) return mFinalStream->bufferSize(); - return mBufferSize; -} - -uint32_t AudioStreamInDump::channels() const -{ - if (mFinalStream != 0 ) return mFinalStream->channels(); - return mChannels; -} - -int AudioStreamInDump::format() const -{ - if (mFinalStream != 0 ) return mFinalStream->format(); - return mFormat; -} - -status_t AudioStreamInDump::setParameters(const String8& keyValuePairs) -{ - LOGV("AudioStreamInDump::setParameters()"); - if (mFinalStream != 0 ) return mFinalStream->setParameters(keyValuePairs); - return NO_ERROR; -} - -String8 AudioStreamInDump::getParameters(const String8& keys) -{ - if (mFinalStream != 0 ) return mFinalStream->getParameters(keys); - - AudioParameter param = AudioParameter(keys); - return param.toString(); -} - -unsigned int AudioStreamInDump::getInputFramesLost() const -{ - if (mFinalStream != 0 ) return mFinalStream->getInputFramesLost(); - return 0; -} - -status_t AudioStreamInDump::dump(int fd, const Vector<String16>& args) -{ - if (mFinalStream != 0 ) return mFinalStream->dump(fd, args); - return NO_ERROR; -} - -void AudioStreamInDump::Close() -{ - if(mFile) { - fclose(mFile); - mFile = 0; - } -} -}; // namespace android diff --git a/libs/audioflinger/AudioDumpInterface.h b/libs/audioflinger/AudioDumpInterface.h deleted file mode 100644 index 814ce5f..0000000 --- a/libs/audioflinger/AudioDumpInterface.h +++ /dev/null @@ -1,170 +0,0 @@ -/* //device/servers/AudioFlinger/AudioDumpInterface.h -** -** Copyright 2008, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#ifndef ANDROID_AUDIO_DUMP_INTERFACE_H -#define ANDROID_AUDIO_DUMP_INTERFACE_H - -#include <stdint.h> -#include <sys/types.h> -#include <utils/String8.h> -#include <utils/SortedVector.h> - -#include <hardware_legacy/AudioHardwareBase.h> - -namespace android { - -#define AUDIO_DUMP_WAVE_HDR_SIZE 44 - -class AudioDumpInterface; - -class AudioStreamOutDump : public AudioStreamOut { -public: - AudioStreamOutDump(AudioDumpInterface *interface, - int id, - AudioStreamOut* finalStream, - uint32_t devices, - int format, - uint32_t channels, - uint32_t sampleRate); - ~AudioStreamOutDump(); - - virtual ssize_t write(const void* buffer, size_t bytes); - virtual uint32_t sampleRate() const; - virtual size_t bufferSize() const; - virtual uint32_t channels() const; - virtual int format() const; - virtual uint32_t latency() const; - virtual status_t setVolume(float left, float right); - virtual status_t standby(); - virtual status_t setParameters(const String8& keyValuePairs); - virtual String8 getParameters(const String8& keys); - virtual status_t dump(int fd, const Vector<String16>& args); - void Close(void); - AudioStreamOut* finalStream() { return mFinalStream; } - uint32_t device() { return mDevice; } - int getId() { return mId; } - virtual status_t getRenderPosition(uint32_t *dspFrames); - -private: - AudioDumpInterface *mInterface; - int mId; - uint32_t mSampleRate; // - uint32_t mFormat; // - uint32_t mChannels; // output configuration - uint32_t mLatency; // - uint32_t mDevice; // current device this output is routed to - size_t mBufferSize; - AudioStreamOut *mFinalStream; - FILE *mFile; // output file - int mFileCount; -}; - -class AudioStreamInDump : public AudioStreamIn { -public: - AudioStreamInDump(AudioDumpInterface *interface, - int id, - AudioStreamIn* finalStream, - uint32_t devices, - int format, - uint32_t channels, - uint32_t sampleRate); - ~AudioStreamInDump(); - - virtual uint32_t sampleRate() const; - virtual size_t bufferSize() const; - virtual uint32_t channels() const; - virtual int format() const; - - virtual status_t setGain(float gain); - virtual ssize_t read(void* buffer, ssize_t bytes); - virtual status_t standby(); - virtual status_t setParameters(const String8& keyValuePairs); - virtual String8 getParameters(const String8& keys); - virtual unsigned int getInputFramesLost() const; - virtual status_t dump(int fd, const Vector<String16>& args); - void Close(void); - AudioStreamIn* finalStream() { return mFinalStream; } - uint32_t device() { return mDevice; } - -private: - AudioDumpInterface *mInterface; - int mId; - uint32_t mSampleRate; // - uint32_t mFormat; // - uint32_t mChannels; // output configuration - uint32_t mDevice; // current device this output is routed to - size_t mBufferSize; - AudioStreamIn *mFinalStream; - FILE *mFile; // output file - int mFileCount; -}; - -class AudioDumpInterface : public AudioHardwareBase -{ - -public: - AudioDumpInterface(AudioHardwareInterface* hw); - virtual AudioStreamOut* openOutputStream( - uint32_t devices, - int *format=0, - uint32_t *channels=0, - uint32_t *sampleRate=0, - status_t *status=0); - virtual void closeOutputStream(AudioStreamOut* out); - - virtual ~AudioDumpInterface(); - - virtual status_t initCheck() - {return mFinalInterface->initCheck();} - virtual status_t setVoiceVolume(float volume) - {return mFinalInterface->setVoiceVolume(volume);} - virtual status_t setMasterVolume(float volume) - {return mFinalInterface->setMasterVolume(volume);} - - virtual status_t setMode(int mode); - - // mic mute - virtual status_t setMicMute(bool state) - {return mFinalInterface->setMicMute(state);} - virtual status_t getMicMute(bool* state) - {return mFinalInterface->getMicMute(state);} - - virtual status_t setParameters(const String8& keyValuePairs); - virtual String8 getParameters(const String8& keys); - - virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount); - - virtual AudioStreamIn* openInputStream(uint32_t devices, int *format, uint32_t *channels, - uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics); - virtual void closeInputStream(AudioStreamIn* in); - - virtual status_t dump(int fd, const Vector<String16>& args) { return mFinalInterface->dumpState(fd, args); } - - String8 fileName() const { return mFileName; } -protected: - - AudioHardwareInterface *mFinalInterface; - SortedVector<AudioStreamOutDump *> mOutputs; - SortedVector<AudioStreamInDump *> mInputs; - Mutex mLock; - String8 mPolicyCommands; - String8 mFileName; -}; - -}; // namespace android - -#endif // ANDROID_AUDIO_DUMP_INTERFACE_H diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp deleted file mode 100644 index 97eb6c0..0000000 --- a/libs/audioflinger/AudioFlinger.cpp +++ /dev/null @@ -1,6078 +0,0 @@ -/* //device/include/server/AudioFlinger/AudioFlinger.cpp -** -** Copyright 2007, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - - -#define LOG_TAG "AudioFlinger" -//#define LOG_NDEBUG 0 - -#include <math.h> -#include <signal.h> -#include <sys/time.h> -#include <sys/resource.h> - -#include <binder/IServiceManager.h> -#include <utils/Log.h> -#include <binder/Parcel.h> -#include <binder/IPCThreadState.h> -#include <utils/String16.h> -#include <utils/threads.h> - -#include <cutils/properties.h> - -#include <media/AudioTrack.h> -#include <media/AudioRecord.h> - -#include <private/media/AudioTrackShared.h> -#include <private/media/AudioEffectShared.h> -#include <hardware_legacy/AudioHardwareInterface.h> - -#include "AudioMixer.h" -#include "AudioFlinger.h" - -#ifdef WITH_A2DP -#include "A2dpAudioInterface.h" -#endif - -#ifdef LVMX -#include "lifevibes.h" -#endif - -#include <media/EffectsFactoryApi.h> -#include <media/EffectVisualizerApi.h> - -// ---------------------------------------------------------------------------- -// the sim build doesn't have gettid - -#ifndef HAVE_GETTID -# define gettid getpid -#endif - -// ---------------------------------------------------------------------------- - -namespace android { - -static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; -static const char* kHardwareLockedString = "Hardware lock is taken\n"; - -//static const nsecs_t kStandbyTimeInNsecs = seconds(3); -static const float MAX_GAIN = 4096.0f; -static const float MAX_GAIN_INT = 0x1000; - -// retry counts for buffer fill timeout -// 50 * ~20msecs = 1 second -static const int8_t kMaxTrackRetries = 50; -static const int8_t kMaxTrackStartupRetries = 50; -// allow less retry attempts on direct output thread. -// direct outputs can be a scarce resource in audio hardware and should -// be released as quickly as possible. -static const int8_t kMaxTrackRetriesDirect = 2; - -static const int kDumpLockRetries = 50; -static const int kDumpLockSleep = 20000; - -static const nsecs_t kWarningThrottle = seconds(5); - - -#define AUDIOFLINGER_SECURITY_ENABLED 1 - -// ---------------------------------------------------------------------------- - -static bool recordingAllowed() { -#ifndef HAVE_ANDROID_OS - return true; -#endif -#if AUDIOFLINGER_SECURITY_ENABLED - if (getpid() == IPCThreadState::self()->getCallingPid()) return true; - bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); - if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); - return ok; -#else - if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO"))) - LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest"); - return true; -#endif -} - -static bool settingsAllowed() { -#ifndef HAVE_ANDROID_OS - return true; -#endif -#if AUDIOFLINGER_SECURITY_ENABLED - if (getpid() == IPCThreadState::self()->getCallingPid()) return true; - bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); - if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); - return ok; -#else - if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"))) - LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest"); - return true; -#endif -} - -// ---------------------------------------------------------------------------- - -AudioFlinger::AudioFlinger() - : BnAudioFlinger(), - mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), - mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0) -{ - mHardwareStatus = AUDIO_HW_IDLE; - - mAudioHardware = AudioHardwareInterface::create(); - - mHardwareStatus = AUDIO_HW_INIT; - if (mAudioHardware->initCheck() == NO_ERROR) { - // open 16-bit output stream for s/w mixer - mMode = AudioSystem::MODE_NORMAL; - setMode(mMode); - - setMasterVolume(1.0f); - setMasterMute(false); - } else { - LOGE("Couldn't even initialize the stubbed audio hardware!"); - } -#ifdef LVMX - LifeVibes::init(); - mLifeVibesClientPid = -1; -#endif -} - -AudioFlinger::~AudioFlinger() -{ - while (!mRecordThreads.isEmpty()) { - // closeInput() will remove first entry from mRecordThreads - closeInput(mRecordThreads.keyAt(0)); - } - while (!mPlaybackThreads.isEmpty()) { - // closeOutput() will remove first entry from mPlaybackThreads - closeOutput(mPlaybackThreads.keyAt(0)); - } - if (mAudioHardware) { - delete mAudioHardware; - } -} - - - -status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - result.append("Clients:\n"); - for (size_t i = 0; i < mClients.size(); ++i) { - wp<Client> wClient = mClients.valueAt(i); - if (wClient != 0) { - sp<Client> client = wClient.promote(); - if (client != 0) { - snprintf(buffer, SIZE, " pid: %d\n", client->pid()); - result.append(buffer); - } - } - } - write(fd, result.string(), result.size()); - return NO_ERROR; -} - - -status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - int hardwareStatus = mHardwareStatus; - - snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); - result.append(buffer); - write(fd, result.string(), result.size()); - return NO_ERROR; -} - -status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - snprintf(buffer, SIZE, "Permission Denial: " - "can't dump AudioFlinger from pid=%d, uid=%d\n", - IPCThreadState::self()->getCallingPid(), - IPCThreadState::self()->getCallingUid()); - result.append(buffer); - write(fd, result.string(), result.size()); - return NO_ERROR; -} - -static bool tryLock(Mutex& mutex) -{ - bool locked = false; - for (int i = 0; i < kDumpLockRetries; ++i) { - if (mutex.tryLock() == NO_ERROR) { - locked = true; - break; - } - usleep(kDumpLockSleep); - } - return locked; -} - -status_t AudioFlinger::dump(int fd, const Vector<String16>& args) -{ - if (checkCallingPermission(String16("android.permission.DUMP")) == false) { - dumpPermissionDenial(fd, args); - } else { - // get state of hardware lock - bool hardwareLocked = tryLock(mHardwareLock); - if (!hardwareLocked) { - String8 result(kHardwareLockedString); - write(fd, result.string(), result.size()); - } else { - mHardwareLock.unlock(); - } - - bool locked = tryLock(mLock); - - // failed to lock - AudioFlinger is probably deadlocked - if (!locked) { - String8 result(kDeadlockedString); - write(fd, result.string(), result.size()); - } - - dumpClients(fd, args); - dumpInternals(fd, args); - - // dump playback threads - for (size_t i = 0; i < mPlaybackThreads.size(); i++) { - mPlaybackThreads.valueAt(i)->dump(fd, args); - } - - // dump record threads - for (size_t i = 0; i < mRecordThreads.size(); i++) { - mRecordThreads.valueAt(i)->dump(fd, args); - } - - if (mAudioHardware) { - mAudioHardware->dumpState(fd, args); - } - if (locked) mLock.unlock(); - } - return NO_ERROR; -} - - -// IAudioFlinger interface - - -sp<IAudioTrack> AudioFlinger::createTrack( - pid_t pid, - int streamType, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount, - uint32_t flags, - const sp<IMemory>& sharedBuffer, - int output, - int *sessionId, - status_t *status) -{ - sp<PlaybackThread::Track> track; - sp<TrackHandle> trackHandle; - sp<Client> client; - wp<Client> wclient; - status_t lStatus; - int lSessionId; - - if (streamType >= AudioSystem::NUM_STREAM_TYPES) { - LOGE("invalid stream type"); - lStatus = BAD_VALUE; - goto Exit; - } - - { - Mutex::Autolock _l(mLock); - PlaybackThread *thread = checkPlaybackThread_l(output); - if (thread == NULL) { - LOGE("unknown output thread"); - lStatus = BAD_VALUE; - goto Exit; - } - - wclient = mClients.valueFor(pid); - - if (wclient != NULL) { - client = wclient.promote(); - } else { - client = new Client(this, pid); - mClients.add(pid, client); - } - - // If no audio session id is provided, create one here - // TODO: enforce same stream type for all tracks in same audio session? - // TODO: prevent same audio session on different output threads - LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); - if (sessionId != NULL && *sessionId != 0) { - lSessionId = *sessionId; - } else { - lSessionId = nextUniqueId(); - if (sessionId != NULL) { - *sessionId = lSessionId; - } - } - LOGV("createTrack() lSessionId: %d", lSessionId); - - track = thread->createTrack_l(client, streamType, sampleRate, format, - channelCount, frameCount, sharedBuffer, lSessionId, &lStatus); - } - if (lStatus == NO_ERROR) { - trackHandle = new TrackHandle(track); - } else { - // remove local strong reference to Client before deleting the Track so that the Client - // destructor is called by the TrackBase destructor with mLock held - client.clear(); - track.clear(); - } - -Exit: - if(status) { - *status = lStatus; - } - return trackHandle; -} - -uint32_t AudioFlinger::sampleRate(int output) const -{ - Mutex::Autolock _l(mLock); - PlaybackThread *thread = checkPlaybackThread_l(output); - if (thread == NULL) { - LOGW("sampleRate() unknown thread %d", output); - return 0; - } - return thread->sampleRate(); -} - -int AudioFlinger::channelCount(int output) const -{ - Mutex::Autolock _l(mLock); - PlaybackThread *thread = checkPlaybackThread_l(output); - if (thread == NULL) { - LOGW("channelCount() unknown thread %d", output); - return 0; - } - return thread->channelCount(); -} - -int AudioFlinger::format(int output) const -{ - Mutex::Autolock _l(mLock); - PlaybackThread *thread = checkPlaybackThread_l(output); - if (thread == NULL) { - LOGW("format() unknown thread %d", output); - return 0; - } - return thread->format(); -} - -size_t AudioFlinger::frameCount(int output) const -{ - Mutex::Autolock _l(mLock); - PlaybackThread *thread = checkPlaybackThread_l(output); - if (thread == NULL) { - LOGW("frameCount() unknown thread %d", output); - return 0; - } - return thread->frameCount(); -} - -uint32_t AudioFlinger::latency(int output) const -{ - Mutex::Autolock _l(mLock); - PlaybackThread *thread = checkPlaybackThread_l(output); - if (thread == NULL) { - LOGW("latency() unknown thread %d", output); - return 0; - } - return thread->latency(); -} - -status_t AudioFlinger::setMasterVolume(float value) -{ - // check calling permissions - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - - // when hw supports master volume, don't scale in sw mixer - AutoMutex lock(mHardwareLock); - mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; - if (mAudioHardware->setMasterVolume(value) == NO_ERROR) { - value = 1.0f; - } - mHardwareStatus = AUDIO_HW_IDLE; - - mMasterVolume = value; - for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) - mPlaybackThreads.valueAt(i)->setMasterVolume(value); - - return NO_ERROR; -} - -status_t AudioFlinger::setMode(int mode) -{ - status_t ret; - - // check calling permissions - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) { - LOGW("Illegal value: setMode(%d)", mode); - return BAD_VALUE; - } - - { // scope for the lock - AutoMutex lock(mHardwareLock); - mHardwareStatus = AUDIO_HW_SET_MODE; - ret = mAudioHardware->setMode(mode); - mHardwareStatus = AUDIO_HW_IDLE; - } - - if (NO_ERROR == ret) { - Mutex::Autolock _l(mLock); - mMode = mode; - for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) - mPlaybackThreads.valueAt(i)->setMode(mode); -#ifdef LVMX - LifeVibes::setMode(mode); -#endif - } - - return ret; -} - -status_t AudioFlinger::setMicMute(bool state) -{ - // check calling permissions - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - - AutoMutex lock(mHardwareLock); - mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; - status_t ret = mAudioHardware->setMicMute(state); - mHardwareStatus = AUDIO_HW_IDLE; - return ret; -} - -bool AudioFlinger::getMicMute() const -{ - bool state = AudioSystem::MODE_INVALID; - mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; - mAudioHardware->getMicMute(&state); - mHardwareStatus = AUDIO_HW_IDLE; - return state; -} - -status_t AudioFlinger::setMasterMute(bool muted) -{ - // check calling permissions - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - - mMasterMute = muted; - for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) - mPlaybackThreads.valueAt(i)->setMasterMute(muted); - - return NO_ERROR; -} - -float AudioFlinger::masterVolume() const -{ - return mMasterVolume; -} - -bool AudioFlinger::masterMute() const -{ - return mMasterMute; -} - -status_t AudioFlinger::setStreamVolume(int stream, float value, int output) -{ - // check calling permissions - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - - if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { - return BAD_VALUE; - } - - AutoMutex lock(mLock); - PlaybackThread *thread = NULL; - if (output) { - thread = checkPlaybackThread_l(output); - if (thread == NULL) { - return BAD_VALUE; - } - } - - mStreamTypes[stream].volume = value; - - if (thread == NULL) { - for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { - mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); - } - } else { - thread->setStreamVolume(stream, value); - } - - return NO_ERROR; -} - -status_t AudioFlinger::setStreamMute(int stream, bool muted) -{ - // check calling permissions - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - - if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES || - uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) { - return BAD_VALUE; - } - - mStreamTypes[stream].mute = muted; - for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) - mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); - - return NO_ERROR; -} - -float AudioFlinger::streamVolume(int stream, int output) const -{ - if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { - return 0.0f; - } - - AutoMutex lock(mLock); - float volume; - if (output) { - PlaybackThread *thread = checkPlaybackThread_l(output); - if (thread == NULL) { - return 0.0f; - } - volume = thread->streamVolume(stream); - } else { - volume = mStreamTypes[stream].volume; - } - - return volume; -} - -bool AudioFlinger::streamMute(int stream) const -{ - if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) { - return true; - } - - return mStreamTypes[stream].mute; -} - -bool AudioFlinger::isStreamActive(int stream) const -{ - Mutex::Autolock _l(mLock); - for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { - if (mPlaybackThreads.valueAt(i)->isStreamActive(stream)) { - return true; - } - } - return false; -} - -status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) -{ - status_t result; - - LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", - ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); - // check calling permissions - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - -#ifdef LVMX - AudioParameter param = AudioParameter(keyValuePairs); - LifeVibes::setParameters(ioHandle,keyValuePairs); - String8 key = String8(AudioParameter::keyRouting); - int device; - if (NO_ERROR != param.getInt(key, device)) { - device = -1; - } - - key = String8(LifevibesTag); - String8 value; - int musicEnabled = -1; - if (NO_ERROR == param.get(key, value)) { - if (value == LifevibesEnable) { - mLifeVibesClientPid = IPCThreadState::self()->getCallingPid(); - musicEnabled = 1; - } else if (value == LifevibesDisable) { - mLifeVibesClientPid = -1; - musicEnabled = 0; - } - } -#endif - - // ioHandle == 0 means the parameters are global to the audio hardware interface - if (ioHandle == 0) { - AutoMutex lock(mHardwareLock); - mHardwareStatus = AUDIO_SET_PARAMETER; - result = mAudioHardware->setParameters(keyValuePairs); -#ifdef LVMX - if (musicEnabled != -1) { - LifeVibes::enableMusic((bool) musicEnabled); - } -#endif - mHardwareStatus = AUDIO_HW_IDLE; - return result; - } - - // hold a strong ref on thread in case closeOutput() or closeInput() is called - // and the thread is exited once the lock is released - sp<ThreadBase> thread; - { - Mutex::Autolock _l(mLock); - thread = checkPlaybackThread_l(ioHandle); - if (thread == NULL) { - thread = checkRecordThread_l(ioHandle); - } - } - if (thread != NULL) { - result = thread->setParameters(keyValuePairs); -#ifdef LVMX - if ((NO_ERROR == result) && (device != -1)) { - LifeVibes::setDevice(LifeVibes::threadIdToAudioOutputType(thread->id()), device); - } -#endif - return result; - } - return BAD_VALUE; -} - -String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) -{ -// LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", -// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); - - if (ioHandle == 0) { - return mAudioHardware->getParameters(keys); - } - - Mutex::Autolock _l(mLock); - - PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); - if (playbackThread != NULL) { - return playbackThread->getParameters(keys); - } - RecordThread *recordThread = checkRecordThread_l(ioHandle); - if (recordThread != NULL) { - return recordThread->getParameters(keys); - } - return String8(""); -} - -size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) -{ - return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount); -} - -unsigned int AudioFlinger::getInputFramesLost(int ioHandle) -{ - if (ioHandle == 0) { - return 0; - } - - Mutex::Autolock _l(mLock); - - RecordThread *recordThread = checkRecordThread_l(ioHandle); - if (recordThread != NULL) { - return recordThread->getInputFramesLost(); - } - return 0; -} - -status_t AudioFlinger::setVoiceVolume(float value) -{ - // check calling permissions - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - - AutoMutex lock(mHardwareLock); - mHardwareStatus = AUDIO_SET_VOICE_VOLUME; - status_t ret = mAudioHardware->setVoiceVolume(value); - mHardwareStatus = AUDIO_HW_IDLE; - - return ret; -} - -status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) -{ - status_t status; - - Mutex::Autolock _l(mLock); - - PlaybackThread *playbackThread = checkPlaybackThread_l(output); - if (playbackThread != NULL) { - return playbackThread->getRenderPosition(halFrames, dspFrames); - } - - return BAD_VALUE; -} - -void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) -{ - - Mutex::Autolock _l(mLock); - - int pid = IPCThreadState::self()->getCallingPid(); - if (mNotificationClients.indexOfKey(pid) < 0) { - sp<NotificationClient> notificationClient = new NotificationClient(this, - client, - pid); - LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); - - mNotificationClients.add(pid, notificationClient); - - sp<IBinder> binder = client->asBinder(); - binder->linkToDeath(notificationClient); - - // the config change is always sent from playback or record threads to avoid deadlock - // with AudioSystem::gLock - for (size_t i = 0; i < mPlaybackThreads.size(); i++) { - mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); - } - - for (size_t i = 0; i < mRecordThreads.size(); i++) { - mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); - } - } -} - -void AudioFlinger::removeNotificationClient(pid_t pid) -{ - Mutex::Autolock _l(mLock); - - int index = mNotificationClients.indexOfKey(pid); - if (index >= 0) { - sp <NotificationClient> client = mNotificationClients.valueFor(pid); - LOGV("removeNotificationClient() %p, pid %d", client.get(), pid); -#ifdef LVMX - if (pid == mLifeVibesClientPid) { - LOGV("Disabling lifevibes"); - LifeVibes::enableMusic(false); - mLifeVibesClientPid = -1; - } -#endif - mNotificationClients.removeItem(pid); - } -} - -// audioConfigChanged_l() must be called with AudioFlinger::mLock held -void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) -{ - size_t size = mNotificationClients.size(); - for (size_t i = 0; i < size; i++) { - mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); - } -} - -// removeClient_l() must be called with AudioFlinger::mLock held -void AudioFlinger::removeClient_l(pid_t pid) -{ - LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); - mClients.removeItem(pid); -} - - -// ---------------------------------------------------------------------------- - -AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id) - : Thread(false), - mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), - mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false) -{ -} - -AudioFlinger::ThreadBase::~ThreadBase() -{ - mParamCond.broadcast(); - mNewParameters.clear(); -} - -void AudioFlinger::ThreadBase::exit() -{ - // keep a strong ref on ourself so that we wont get - // destroyed in the middle of requestExitAndWait() - sp <ThreadBase> strongMe = this; - - LOGV("ThreadBase::exit"); - { - AutoMutex lock(&mLock); - mExiting = true; - requestExit(); - mWaitWorkCV.signal(); - } - requestExitAndWait(); -} - -uint32_t AudioFlinger::ThreadBase::sampleRate() const -{ - return mSampleRate; -} - -int AudioFlinger::ThreadBase::channelCount() const -{ - return (int)mChannelCount; -} - -int AudioFlinger::ThreadBase::format() const -{ - return mFormat; -} - -size_t AudioFlinger::ThreadBase::frameCount() const -{ - return mFrameCount; -} - -status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) -{ - status_t status; - - LOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); - Mutex::Autolock _l(mLock); - - mNewParameters.add(keyValuePairs); - mWaitWorkCV.signal(); - // wait condition with timeout in case the thread loop has exited - // before the request could be processed - if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) { - status = mParamStatus; - mWaitWorkCV.signal(); - } else { - status = TIMED_OUT; - } - return status; -} - -void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) -{ - Mutex::Autolock _l(mLock); - sendConfigEvent_l(event, param); -} - -// sendConfigEvent_l() must be called with ThreadBase::mLock held -void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) -{ - ConfigEvent *configEvent = new ConfigEvent(); - configEvent->mEvent = event; - configEvent->mParam = param; - mConfigEvents.add(configEvent); - LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); - mWaitWorkCV.signal(); -} - -void AudioFlinger::ThreadBase::processConfigEvents() -{ - mLock.lock(); - while(!mConfigEvents.isEmpty()) { - LOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); - ConfigEvent *configEvent = mConfigEvents[0]; - mConfigEvents.removeAt(0); - // release mLock before locking AudioFlinger mLock: lock order is always - // AudioFlinger then ThreadBase to avoid cross deadlock - mLock.unlock(); - mAudioFlinger->mLock.lock(); - audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); - mAudioFlinger->mLock.unlock(); - delete configEvent; - mLock.lock(); - } - mLock.unlock(); -} - -status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - bool locked = tryLock(mLock); - if (!locked) { - snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); - write(fd, buffer, strlen(buffer)); - } - - snprintf(buffer, SIZE, "standby: %d\n", mStandby); - result.append(buffer); - snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); - result.append(buffer); - snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); - result.append(buffer); - snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); - result.append(buffer); - snprintf(buffer, SIZE, "Format: %d\n", mFormat); - result.append(buffer); - snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); - result.append(buffer); - - snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); - result.append(buffer); - result.append(" Index Command"); - for (size_t i = 0; i < mNewParameters.size(); ++i) { - snprintf(buffer, SIZE, "\n %02d ", i); - result.append(buffer); - result.append(mNewParameters[i]); - } - - snprintf(buffer, SIZE, "\n\nPending config events: \n"); - result.append(buffer); - snprintf(buffer, SIZE, " Index event param\n"); - result.append(buffer); - for (size_t i = 0; i < mConfigEvents.size(); i++) { - snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); - result.append(buffer); - } - result.append("\n"); - - write(fd, result.string(), result.size()); - - if (locked) { - mLock.unlock(); - } - return NO_ERROR; -} - - -// ---------------------------------------------------------------------------- - -AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) - : ThreadBase(audioFlinger, id), - mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), - mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), - mDevice(device) -{ - readOutputParameters(); - - mMasterVolume = mAudioFlinger->masterVolume(); - mMasterMute = mAudioFlinger->masterMute(); - - for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { - mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); - mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); - } -} - -AudioFlinger::PlaybackThread::~PlaybackThread() -{ - delete [] mMixBuffer; -} - -status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) -{ - dumpInternals(fd, args); - dumpTracks(fd, args); - dumpEffectChains(fd, args); - return NO_ERROR; -} - -status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "Output thread %p tracks\n", this); - result.append(buffer); - result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); - for (size_t i = 0; i < mTracks.size(); ++i) { - sp<Track> track = mTracks[i]; - if (track != 0) { - track->dump(buffer, SIZE); - result.append(buffer); - } - } - - snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); - result.append(buffer); - result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); - for (size_t i = 0; i < mActiveTracks.size(); ++i) { - wp<Track> wTrack = mActiveTracks[i]; - if (wTrack != 0) { - sp<Track> track = wTrack.promote(); - if (track != 0) { - track->dump(buffer, SIZE); - result.append(buffer); - } - } - } - write(fd, result.string(), result.size()); - return NO_ERROR; -} - -status_t AudioFlinger::PlaybackThread::dumpEffectChains(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); - write(fd, buffer, strlen(buffer)); - - for (size_t i = 0; i < mEffectChains.size(); ++i) { - sp<EffectChain> chain = mEffectChains[i]; - if (chain != 0) { - chain->dump(fd, args); - } - } - return NO_ERROR; -} - -status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); - result.append(buffer); - snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); - result.append(buffer); - snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); - result.append(buffer); - snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); - result.append(buffer); - snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); - result.append(buffer); - snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); - result.append(buffer); - snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); - result.append(buffer); - write(fd, result.string(), result.size()); - - dumpBase(fd, args); - - return NO_ERROR; -} - -// Thread virtuals -status_t AudioFlinger::PlaybackThread::readyToRun() -{ - if (mSampleRate == 0) { - LOGE("No working audio driver found."); - return NO_INIT; - } - LOGI("AudioFlinger's thread %p ready to run", this); - return NO_ERROR; -} - -void AudioFlinger::PlaybackThread::onFirstRef() -{ - const size_t SIZE = 256; - char buffer[SIZE]; - - snprintf(buffer, SIZE, "Playback Thread %p", this); - - run(buffer, ANDROID_PRIORITY_URGENT_AUDIO); -} - -// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held -sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( - const sp<AudioFlinger::Client>& client, - int streamType, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId, - status_t *status) -{ - sp<Track> track; - status_t lStatus; - - if (mType == DIRECT) { - if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) { - LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p", - sampleRate, format, channelCount, mOutput); - lStatus = BAD_VALUE; - goto Exit; - } - } else { - // Resampler implementation limits input sampling rate to 2 x output sampling rate. - if (sampleRate > mSampleRate*2) { - LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); - lStatus = BAD_VALUE; - goto Exit; - } - } - - if (mOutput == 0) { - LOGE("Audio driver not initialized."); - lStatus = NO_INIT; - goto Exit; - } - - { // scope for mLock - Mutex::Autolock _l(mLock); - track = new Track(this, client, streamType, sampleRate, format, - channelCount, frameCount, sharedBuffer, sessionId); - if (track->getCblk() == NULL || track->name() < 0) { - lStatus = NO_MEMORY; - goto Exit; - } - mTracks.add(track); - - sp<EffectChain> chain = getEffectChain_l(sessionId); - if (chain != 0) { - LOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); - track->setMainBuffer(chain->inBuffer()); - } - } - lStatus = NO_ERROR; - -Exit: - if(status) { - *status = lStatus; - } - return track; -} - -uint32_t AudioFlinger::PlaybackThread::latency() const -{ - if (mOutput) { - return mOutput->latency(); - } - else { - return 0; - } -} - -status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) -{ -#ifdef LVMX - int audioOutputType = LifeVibes::getMixerType(mId, mType); - if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { - LifeVibes::setMasterVolume(audioOutputType, value); - } -#endif - mMasterVolume = value; - return NO_ERROR; -} - -status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) -{ -#ifdef LVMX - int audioOutputType = LifeVibes::getMixerType(mId, mType); - if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { - LifeVibes::setMasterMute(audioOutputType, muted); - } -#endif - mMasterMute = muted; - return NO_ERROR; -} - -float AudioFlinger::PlaybackThread::masterVolume() const -{ - return mMasterVolume; -} - -bool AudioFlinger::PlaybackThread::masterMute() const -{ - return mMasterMute; -} - -status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) -{ -#ifdef LVMX - int audioOutputType = LifeVibes::getMixerType(mId, mType); - if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { - LifeVibes::setStreamVolume(audioOutputType, stream, value); - } -#endif - mStreamTypes[stream].volume = value; - return NO_ERROR; -} - -status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) -{ -#ifdef LVMX - int audioOutputType = LifeVibes::getMixerType(mId, mType); - if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { - LifeVibes::setStreamMute(audioOutputType, stream, muted); - } -#endif - mStreamTypes[stream].mute = muted; - return NO_ERROR; -} - -float AudioFlinger::PlaybackThread::streamVolume(int stream) const -{ - return mStreamTypes[stream].volume; -} - -bool AudioFlinger::PlaybackThread::streamMute(int stream) const -{ - return mStreamTypes[stream].mute; -} - -bool AudioFlinger::PlaybackThread::isStreamActive(int stream) const -{ - Mutex::Autolock _l(mLock); - size_t count = mActiveTracks.size(); - for (size_t i = 0 ; i < count ; ++i) { - sp<Track> t = mActiveTracks[i].promote(); - if (t == 0) continue; - Track* const track = t.get(); - if (t->type() == stream) - return true; - } - return false; -} - -// addTrack_l() must be called with ThreadBase::mLock held -status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) -{ - status_t status = ALREADY_EXISTS; - - // set retry count for buffer fill - track->mRetryCount = kMaxTrackStartupRetries; - if (mActiveTracks.indexOf(track) < 0) { - // the track is newly added, make sure it fills up all its - // buffers before playing. This is to ensure the client will - // effectively get the latency it requested. - track->mFillingUpStatus = Track::FS_FILLING; - track->mResetDone = false; - mActiveTracks.add(track); - if (track->mainBuffer() != mMixBuffer) { - sp<EffectChain> chain = getEffectChain_l(track->sessionId()); - if (chain != 0) { - LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); - chain->startTrack(); - } - } - - status = NO_ERROR; - } - - LOGV("mWaitWorkCV.broadcast"); - mWaitWorkCV.broadcast(); - - return status; -} - -// destroyTrack_l() must be called with ThreadBase::mLock held -void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) -{ - track->mState = TrackBase::TERMINATED; - if (mActiveTracks.indexOf(track) < 0) { - mTracks.remove(track); - deleteTrackName_l(track->name()); - } -} - -String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) -{ - return mOutput->getParameters(keys); -} - -// destroyTrack_l() must be called with AudioFlinger::mLock held -void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { - AudioSystem::OutputDescriptor desc; - void *param2 = 0; - - LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); - - switch (event) { - case AudioSystem::OUTPUT_OPENED: - case AudioSystem::OUTPUT_CONFIG_CHANGED: - desc.channels = mChannels; - desc.samplingRate = mSampleRate; - desc.format = mFormat; - desc.frameCount = mFrameCount; - desc.latency = latency(); - param2 = &desc; - break; - - case AudioSystem::STREAM_CONFIG_CHANGED: - param2 = ¶m; - case AudioSystem::OUTPUT_CLOSED: - default: - break; - } - mAudioFlinger->audioConfigChanged_l(event, mId, param2); -} - -void AudioFlinger::PlaybackThread::readOutputParameters() -{ - mSampleRate = mOutput->sampleRate(); - mChannels = mOutput->channels(); - mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); - mFormat = mOutput->format(); - mFrameSize = (uint16_t)mOutput->frameSize(); - mFrameCount = mOutput->bufferSize() / mFrameSize; - - // FIXME - Current mixer implementation only supports stereo output: Always - // Allocate a stereo buffer even if HW output is mono. - if (mMixBuffer != NULL) delete[] mMixBuffer; - mMixBuffer = new int16_t[mFrameCount * 2]; - memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); - - //TODO handle effects reconfig -} - -status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) -{ - if (halFrames == 0 || dspFrames == 0) { - return BAD_VALUE; - } - if (mOutput == 0) { - return INVALID_OPERATION; - } - *halFrames = mBytesWritten/mOutput->frameSize(); - - return mOutput->getRenderPosition(dspFrames); -} - -bool AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) -{ - Mutex::Autolock _l(mLock); - if (getEffectChain_l(sessionId) != 0) { - return true; - } - - for (size_t i = 0; i < mTracks.size(); ++i) { - sp<Track> track = mTracks[i]; - if (sessionId == track->sessionId()) { - return true; - } - } - - return false; -} - -sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain(int sessionId) -{ - Mutex::Autolock _l(mLock); - return getEffectChain_l(sessionId); -} - -sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain_l(int sessionId) -{ - sp<EffectChain> chain; - - size_t size = mEffectChains.size(); - for (size_t i = 0; i < size; i++) { - if (mEffectChains[i]->sessionId() == sessionId) { - chain = mEffectChains[i]; - break; - } - } - return chain; -} - -void AudioFlinger::PlaybackThread::setMode(uint32_t mode) -{ - Mutex::Autolock _l(mLock); - size_t size = mEffectChains.size(); - for (size_t i = 0; i < size; i++) { - mEffectChains[i]->setMode(mode); - } -} - -// ---------------------------------------------------------------------------- - -AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) - : PlaybackThread(audioFlinger, output, id, device), - mAudioMixer(0) -{ - mType = PlaybackThread::MIXER; - mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); - - // FIXME - Current mixer implementation only supports stereo output - if (mChannelCount == 1) { - LOGE("Invalid audio hardware channel count"); - } -} - -AudioFlinger::MixerThread::~MixerThread() -{ - delete mAudioMixer; -} - -bool AudioFlinger::MixerThread::threadLoop() -{ - Vector< sp<Track> > tracksToRemove; - uint32_t mixerStatus = MIXER_IDLE; - nsecs_t standbyTime = systemTime(); - size_t mixBufferSize = mFrameCount * mFrameSize; - // FIXME: Relaxed timing because of a certain device that can't meet latency - // Should be reduced to 2x after the vendor fixes the driver issue - nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3; - nsecs_t lastWarning = 0; - bool longStandbyExit = false; - uint32_t activeSleepTime = activeSleepTimeUs(); - uint32_t idleSleepTime = idleSleepTimeUs(); - uint32_t sleepTime = idleSleepTime; - Vector< sp<EffectChain> > effectChains; - - while (!exitPending()) - { - processConfigEvents(); - - mixerStatus = MIXER_IDLE; - { // scope for mLock - - Mutex::Autolock _l(mLock); - - if (checkForNewParameters_l()) { - mixBufferSize = mFrameCount * mFrameSize; - // FIXME: Relaxed timing because of a certain device that can't meet latency - // Should be reduced to 2x after the vendor fixes the driver issue - maxPeriod = seconds(mFrameCount) / mSampleRate * 3; - activeSleepTime = activeSleepTimeUs(); - idleSleepTime = idleSleepTimeUs(); - } - - const SortedVector< wp<Track> >& activeTracks = mActiveTracks; - - // put audio hardware into standby after short delay - if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || - mSuspended) { - if (!mStandby) { - LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); - mOutput->standby(); - mStandby = true; - mBytesWritten = 0; - } - - if (!activeTracks.size() && mConfigEvents.isEmpty()) { - // we're about to wait, flush the binder command buffer - IPCThreadState::self()->flushCommands(); - - if (exitPending()) break; - - // wait until we have something to do... - LOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); - mWaitWorkCV.wait(mLock); - LOGV("MixerThread %p TID %d waking up\n", this, gettid()); - - if (mMasterMute == false) { - char value[PROPERTY_VALUE_MAX]; - property_get("ro.audio.silent", value, "0"); - if (atoi(value)) { - LOGD("Silence is golden"); - setMasterMute(true); - } - } - - standbyTime = systemTime() + kStandbyTimeInNsecs; - sleepTime = idleSleepTime; - continue; - } - } - - mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); - - // prevent any changes in effect chain list and in each effect chain - // during mixing and effect process as the audio buffers could be deleted - // or modified if an effect is created or deleted - effectChains = mEffectChains; - lockEffectChains_l(); - } - - if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { - // mix buffers... - mAudioMixer->process(); - sleepTime = 0; - standbyTime = systemTime() + kStandbyTimeInNsecs; - //TODO: delay standby when effects have a tail - } else { - // If no tracks are ready, sleep once for the duration of an output - // buffer size, then write 0s to the output - if (sleepTime == 0) { - if (mixerStatus == MIXER_TRACKS_ENABLED) { - sleepTime = activeSleepTime; - } else { - sleepTime = idleSleepTime; - } - } else if (mBytesWritten != 0 || - (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { - memset (mMixBuffer, 0, mixBufferSize); - sleepTime = 0; - LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); - } - // TODO add standby time extension fct of effect tail - } - - if (mSuspended) { - sleepTime = idleSleepTime; - } - // sleepTime == 0 means we must write to audio hardware - if (sleepTime == 0) { - for (size_t i = 0; i < effectChains.size(); i ++) { - effectChains[i]->process_l(); - } - // enable changes in effect chain - unlockEffectChains(); -#ifdef LVMX - int audioOutputType = LifeVibes::getMixerType(mId, mType); - if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { - LifeVibes::process(audioOutputType, mMixBuffer, mixBufferSize); - } -#endif - mLastWriteTime = systemTime(); - mInWrite = true; - mBytesWritten += mixBufferSize; - - int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); - if (bytesWritten < 0) mBytesWritten -= mixBufferSize; - mNumWrites++; - mInWrite = false; - nsecs_t now = systemTime(); - nsecs_t delta = now - mLastWriteTime; - if (delta > maxPeriod) { - mNumDelayedWrites++; - if ((now - lastWarning) > kWarningThrottle) { - LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", - ns2ms(delta), mNumDelayedWrites, this); - lastWarning = now; - } - if (mStandby) { - longStandbyExit = true; - } - } - mStandby = false; - } else { - // enable changes in effect chain - unlockEffectChains(); - usleep(sleepTime); - } - - // finally let go of all our tracks, without the lock held - // since we can't guarantee the destructors won't acquire that - // same lock. - tracksToRemove.clear(); - - // Effect chains will be actually deleted here if they were removed from - // mEffectChains list during mixing or effects processing - effectChains.clear(); - } - - if (!mStandby) { - mOutput->standby(); - } - - LOGV("MixerThread %p exiting", this); - return false; -} - -// prepareTracks_l() must be called with ThreadBase::mLock held -uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) -{ - - uint32_t mixerStatus = MIXER_IDLE; - // find out which tracks need to be processed - size_t count = activeTracks.size(); - size_t mixedTracks = 0; - size_t tracksWithEffect = 0; - - float masterVolume = mMasterVolume; - bool masterMute = mMasterMute; - -#ifdef LVMX - bool tracksConnectedChanged = false; - bool stateChanged = false; - - int audioOutputType = LifeVibes::getMixerType(mId, mType); - if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) - { - int activeTypes = 0; - for (size_t i=0 ; i<count ; i++) { - sp<Track> t = activeTracks[i].promote(); - if (t == 0) continue; - Track* const track = t.get(); - int iTracktype=track->type(); - activeTypes |= 1<<track->type(); - } - LifeVibes::computeVolumes(audioOutputType, activeTypes, tracksConnectedChanged, stateChanged, masterVolume, masterMute); - } -#endif - // Delegate master volume control to effect in output mix effect chain if needed - sp<EffectChain> chain = getEffectChain_l(0); - if (chain != 0) { - uint32_t v = (uint32_t)(masterVolume * (1 << 24)); - chain->setVolume(&v, &v); - masterVolume = (float)((v + (1 << 23)) >> 24); - chain.clear(); - } - - for (size_t i=0 ; i<count ; i++) { - sp<Track> t = activeTracks[i].promote(); - if (t == 0) continue; - - Track* const track = t.get(); - audio_track_cblk_t* cblk = track->cblk(); - - // The first time a track is added we wait - // for all its buffers to be filled before processing it - mAudioMixer->setActiveTrack(track->name()); - if (cblk->framesReady() && (track->isReady() || track->isStopped()) && - !track->isPaused() && !track->isTerminated()) - { - //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); - - mixedTracks++; - - // track->mainBuffer() != mMixBuffer means there is an effect chain - // connected to the track - chain.clear(); - if (track->mainBuffer() != mMixBuffer) { - chain = getEffectChain_l(track->sessionId()); - // Delegate volume control to effect in track effect chain if needed - if (chain != 0) { - tracksWithEffect++; - } else { - LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", - track->name(), track->sessionId()); - } - } - - - int param = AudioMixer::VOLUME; - if (track->mFillingUpStatus == Track::FS_FILLED) { - // no ramp for the first volume setting - track->mFillingUpStatus = Track::FS_ACTIVE; - if (track->mState == TrackBase::RESUMING) { - track->mState = TrackBase::ACTIVE; - param = AudioMixer::RAMP_VOLUME; - } - } else if (cblk->server != 0) { - // If the track is stopped before the first frame was mixed, - // do not apply ramp - param = AudioMixer::RAMP_VOLUME; - } - - // compute volume for this track - int16_t left, right, aux; - if (track->isMuted() || masterMute || track->isPausing() || - mStreamTypes[track->type()].mute) { - left = right = aux = 0; - if (track->isPausing()) { - track->setPaused(); - } - } else { - // read original volumes with volume control - float typeVolume = mStreamTypes[track->type()].volume; -#ifdef LVMX - bool streamMute=false; - // read the volume from the LivesVibes audio engine. - if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) - { - LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute); - if (streamMute) { - typeVolume = 0; - } - } -#endif - float v = masterVolume * typeVolume; - uint32_t vl = (uint32_t)(v * cblk->volume[0]) << 12; - uint32_t vr = (uint32_t)(v * cblk->volume[1]) << 12; - - // Delegate volume control to effect in track effect chain if needed - if (chain != 0 && chain->setVolume(&vl, &vr)) { - // Do not ramp volume is volume is controlled by effect - param = AudioMixer::VOLUME; - } - - // Convert volumes from 8.24 to 4.12 format - uint32_t v_clamped = (vl + (1 << 11)) >> 12; - if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; - left = int16_t(v_clamped); - v_clamped = (vr + (1 << 11)) >> 12; - if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; - right = int16_t(v_clamped); - - v_clamped = (uint32_t)(v * cblk->sendLevel); - if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; - aux = int16_t(v_clamped); - } - -#ifdef LVMX - if ( tracksConnectedChanged || stateChanged ) - { - // only do the ramp when the volume is changed by the user / application - param = AudioMixer::VOLUME; - } -#endif - - // XXX: these things DON'T need to be done each time - mAudioMixer->setBufferProvider(track); - mAudioMixer->enable(AudioMixer::MIXING); - - mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); - mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); - mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); - mAudioMixer->setParameter( - AudioMixer::TRACK, - AudioMixer::FORMAT, (void *)track->format()); - mAudioMixer->setParameter( - AudioMixer::TRACK, - AudioMixer::CHANNEL_COUNT, (void *)track->channelCount()); - mAudioMixer->setParameter( - AudioMixer::RESAMPLE, - AudioMixer::SAMPLE_RATE, - (void *)(cblk->sampleRate)); - mAudioMixer->setParameter( - AudioMixer::TRACK, - AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); - mAudioMixer->setParameter( - AudioMixer::TRACK, - AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); - - // reset retry count - track->mRetryCount = kMaxTrackRetries; - mixerStatus = MIXER_TRACKS_READY; - } else { - //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); - if (track->isStopped()) { - track->reset(); - } - if (track->isTerminated() || track->isStopped() || track->isPaused()) { - // We have consumed all the buffers of this track. - // Remove it from the list of active tracks. - tracksToRemove->add(track); - } else { - // No buffers for this track. Give it a few chances to - // fill a buffer, then remove it from active list. - if (--(track->mRetryCount) <= 0) { - LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); - tracksToRemove->add(track); - } else if (mixerStatus != MIXER_TRACKS_READY) { - mixerStatus = MIXER_TRACKS_ENABLED; - } - } - mAudioMixer->disable(AudioMixer::MIXING); - } - } - - // remove all the tracks that need to be... - count = tracksToRemove->size(); - if (UNLIKELY(count)) { - for (size_t i=0 ; i<count ; i++) { - const sp<Track>& track = tracksToRemove->itemAt(i); - mActiveTracks.remove(track); - if (track->mainBuffer() != mMixBuffer) { - chain = getEffectChain_l(track->sessionId()); - if (chain != 0) { - LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); - chain->stopTrack(); - } - } - if (track->isTerminated()) { - mTracks.remove(track); - deleteTrackName_l(track->mName); - } - } - } - - // mix buffer must be cleared if all tracks are connected to an - // effect chain as in this case the mixer will not write to - // mix buffer and track effects will accumulate into it - if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { - memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); - } - - return mixerStatus; -} - -void AudioFlinger::MixerThread::invalidateTracks(int streamType) -{ - LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", this, streamType, mTracks.size()); - Mutex::Autolock _l(mLock); - size_t size = mTracks.size(); - for (size_t i = 0; i < size; i++) { - sp<Track> t = mTracks[i]; - if (t->type() == streamType) { - t->mCblk->lock.lock(); - t->mCblk->flags |= CBLK_INVALID_ON; - t->mCblk->cv.signal(); - t->mCblk->lock.unlock(); - } - } -} - - -// getTrackName_l() must be called with ThreadBase::mLock held -int AudioFlinger::MixerThread::getTrackName_l() -{ - return mAudioMixer->getTrackName(); -} - -// deleteTrackName_l() must be called with ThreadBase::mLock held -void AudioFlinger::MixerThread::deleteTrackName_l(int name) -{ - LOGV("remove track (%d) and delete from mixer", name); - mAudioMixer->deleteTrackName(name); -} - -// checkForNewParameters_l() must be called with ThreadBase::mLock held -bool AudioFlinger::MixerThread::checkForNewParameters_l() -{ - bool reconfig = false; - - while (!mNewParameters.isEmpty()) { - status_t status = NO_ERROR; - String8 keyValuePair = mNewParameters[0]; - AudioParameter param = AudioParameter(keyValuePair); - int value; - - if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { - reconfig = true; - } - if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { - if (value != AudioSystem::PCM_16_BIT) { - status = BAD_VALUE; - } else { - reconfig = true; - } - } - if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { - if (value != AudioSystem::CHANNEL_OUT_STEREO) { - status = BAD_VALUE; - } else { - reconfig = true; - } - } - if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { - // do not accept frame count changes if tracks are open as the track buffer - // size depends on frame count and correct behavior would not be garantied - // if frame count is changed after track creation - if (!mTracks.isEmpty()) { - status = INVALID_OPERATION; - } else { - reconfig = true; - } - } - if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { - // forward device change to effects that have requested to be - // aware of attached audio device. - mDevice = (uint32_t)value; - for (size_t i = 0; i < mEffectChains.size(); i++) { - mEffectChains[i]->setDevice(mDevice); - } - } - - if (status == NO_ERROR) { - status = mOutput->setParameters(keyValuePair); - if (!mStandby && status == INVALID_OPERATION) { - mOutput->standby(); - mStandby = true; - mBytesWritten = 0; - status = mOutput->setParameters(keyValuePair); - } - if (status == NO_ERROR && reconfig) { - delete mAudioMixer; - readOutputParameters(); - mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); - for (size_t i = 0; i < mTracks.size() ; i++) { - int name = getTrackName_l(); - if (name < 0) break; - mTracks[i]->mName = name; - // limit track sample rate to 2 x new output sample rate - if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { - mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); - } - } - sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); - } - } - - mNewParameters.removeAt(0); - - mParamStatus = status; - mParamCond.signal(); - mWaitWorkCV.wait(mLock); - } - return reconfig; -} - -status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - PlaybackThread::dumpInternals(fd, args); - - snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); - result.append(buffer); - write(fd, result.string(), result.size()); - return NO_ERROR; -} - -uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() -{ - return (uint32_t)(mOutput->latency() * 1000) / 2; -} - -uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() -{ - return (uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000; -} - -// ---------------------------------------------------------------------------- -AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) - : PlaybackThread(audioFlinger, output, id, device) -{ - mType = PlaybackThread::DIRECT; -} - -AudioFlinger::DirectOutputThread::~DirectOutputThread() -{ -} - - -static inline int16_t clamp16(int32_t sample) -{ - if ((sample>>15) ^ (sample>>31)) - sample = 0x7FFF ^ (sample>>31); - return sample; -} - -static inline -int32_t mul(int16_t in, int16_t v) -{ -#if defined(__arm__) && !defined(__thumb__) - int32_t out; - asm( "smulbb %[out], %[in], %[v] \n" - : [out]"=r"(out) - : [in]"%r"(in), [v]"r"(v) - : ); - return out; -#else - return in * int32_t(v); -#endif -} - -void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) -{ - // Do not apply volume on compressed audio - if (!AudioSystem::isLinearPCM(mFormat)) { - return; - } - - // convert to signed 16 bit before volume calculation - if (mFormat == AudioSystem::PCM_8_BIT) { - size_t count = mFrameCount * mChannelCount; - uint8_t *src = (uint8_t *)mMixBuffer + count-1; - int16_t *dst = mMixBuffer + count-1; - while(count--) { - *dst-- = (int16_t)(*src--^0x80) << 8; - } - } - - size_t frameCount = mFrameCount; - int16_t *out = mMixBuffer; - if (ramp) { - if (mChannelCount == 1) { - int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; - int32_t vlInc = d / (int32_t)frameCount; - int32_t vl = ((int32_t)mLeftVolShort << 16); - do { - out[0] = clamp16(mul(out[0], vl >> 16) >> 12); - out++; - vl += vlInc; - } while (--frameCount); - - } else { - int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; - int32_t vlInc = d / (int32_t)frameCount; - d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; - int32_t vrInc = d / (int32_t)frameCount; - int32_t vl = ((int32_t)mLeftVolShort << 16); - int32_t vr = ((int32_t)mRightVolShort << 16); - do { - out[0] = clamp16(mul(out[0], vl >> 16) >> 12); - out[1] = clamp16(mul(out[1], vr >> 16) >> 12); - out += 2; - vl += vlInc; - vr += vrInc; - } while (--frameCount); - } - } else { - if (mChannelCount == 1) { - do { - out[0] = clamp16(mul(out[0], leftVol) >> 12); - out++; - } while (--frameCount); - } else { - do { - out[0] = clamp16(mul(out[0], leftVol) >> 12); - out[1] = clamp16(mul(out[1], rightVol) >> 12); - out += 2; - } while (--frameCount); - } - } - - // convert back to unsigned 8 bit after volume calculation - if (mFormat == AudioSystem::PCM_8_BIT) { - size_t count = mFrameCount * mChannelCount; - int16_t *src = mMixBuffer; - uint8_t *dst = (uint8_t *)mMixBuffer; - while(count--) { - *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; - } - } - - mLeftVolShort = leftVol; - mRightVolShort = rightVol; -} - -bool AudioFlinger::DirectOutputThread::threadLoop() -{ - uint32_t mixerStatus = MIXER_IDLE; - sp<Track> trackToRemove; - sp<Track> activeTrack; - nsecs_t standbyTime = systemTime(); - int8_t *curBuf; - size_t mixBufferSize = mFrameCount*mFrameSize; - uint32_t activeSleepTime = activeSleepTimeUs(); - uint32_t idleSleepTime = idleSleepTimeUs(); - uint32_t sleepTime = idleSleepTime; - // use shorter standby delay as on normal output to release - // hardware resources as soon as possible - nsecs_t standbyDelay = microseconds(activeSleepTime*2); - - - while (!exitPending()) - { - bool rampVolume; - uint16_t leftVol; - uint16_t rightVol; - Vector< sp<EffectChain> > effectChains; - - processConfigEvents(); - - mixerStatus = MIXER_IDLE; - - { // scope for the mLock - - Mutex::Autolock _l(mLock); - - if (checkForNewParameters_l()) { - mixBufferSize = mFrameCount*mFrameSize; - activeSleepTime = activeSleepTimeUs(); - idleSleepTime = idleSleepTimeUs(); - standbyDelay = microseconds(activeSleepTime*2); - } - - // put audio hardware into standby after short delay - if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || - mSuspended) { - // wait until we have something to do... - if (!mStandby) { - LOGV("Audio hardware entering standby, mixer %p\n", this); - mOutput->standby(); - mStandby = true; - mBytesWritten = 0; - } - - if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { - // we're about to wait, flush the binder command buffer - IPCThreadState::self()->flushCommands(); - - if (exitPending()) break; - - LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); - mWaitWorkCV.wait(mLock); - LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); - - if (mMasterMute == false) { - char value[PROPERTY_VALUE_MAX]; - property_get("ro.audio.silent", value, "0"); - if (atoi(value)) { - LOGD("Silence is golden"); - setMasterMute(true); - } - } - - standbyTime = systemTime() + standbyDelay; - sleepTime = idleSleepTime; - continue; - } - } - - effectChains = mEffectChains; - - // find out which tracks need to be processed - if (mActiveTracks.size() != 0) { - sp<Track> t = mActiveTracks[0].promote(); - if (t == 0) continue; - - Track* const track = t.get(); - audio_track_cblk_t* cblk = track->cblk(); - - // The first time a track is added we wait - // for all its buffers to be filled before processing it - if (cblk->framesReady() && (track->isReady() || track->isStopped()) && - !track->isPaused() && !track->isTerminated()) - { - //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); - - if (track->mFillingUpStatus == Track::FS_FILLED) { - track->mFillingUpStatus = Track::FS_ACTIVE; - mLeftVolFloat = mRightVolFloat = 0; - mLeftVolShort = mRightVolShort = 0; - if (track->mState == TrackBase::RESUMING) { - track->mState = TrackBase::ACTIVE; - rampVolume = true; - } - } else if (cblk->server != 0) { - // If the track is stopped before the first frame was mixed, - // do not apply ramp - rampVolume = true; - } - // compute volume for this track - float left, right; - if (track->isMuted() || mMasterMute || track->isPausing() || - mStreamTypes[track->type()].mute) { - left = right = 0; - if (track->isPausing()) { - track->setPaused(); - } - } else { - float typeVolume = mStreamTypes[track->type()].volume; - float v = mMasterVolume * typeVolume; - float v_clamped = v * cblk->volume[0]; - if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; - left = v_clamped/MAX_GAIN; - v_clamped = v * cblk->volume[1]; - if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; - right = v_clamped/MAX_GAIN; - } - - if (left != mLeftVolFloat || right != mRightVolFloat) { - mLeftVolFloat = left; - mRightVolFloat = right; - - // If audio HAL implements volume control, - // force software volume to nominal value - if (mOutput->setVolume(left, right) == NO_ERROR) { - left = 1.0f; - right = 1.0f; - } - - // Convert volumes from float to 8.24 - uint32_t vl = (uint32_t)(left * (1 << 24)); - uint32_t vr = (uint32_t)(right * (1 << 24)); - - // Delegate volume control to effect in track effect chain if needed - // only one effect chain can be present on DirectOutputThread, so if - // there is one, the track is connected to it - if (!effectChains.isEmpty()) { - // Do not ramp volume is volume is controlled by effect - if(effectChains[0]->setVolume(&vl, &vr)) { - rampVolume = false; - } - } - - // Convert volumes from 8.24 to 4.12 format - uint32_t v_clamped = (vl + (1 << 11)) >> 12; - if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; - leftVol = (uint16_t)v_clamped; - v_clamped = (vr + (1 << 11)) >> 12; - if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; - rightVol = (uint16_t)v_clamped; - } else { - leftVol = mLeftVolShort; - rightVol = mRightVolShort; - rampVolume = false; - } - - // reset retry count - track->mRetryCount = kMaxTrackRetriesDirect; - activeTrack = t; - mixerStatus = MIXER_TRACKS_READY; - } else { - //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); - if (track->isStopped()) { - track->reset(); - } - if (track->isTerminated() || track->isStopped() || track->isPaused()) { - // We have consumed all the buffers of this track. - // Remove it from the list of active tracks. - trackToRemove = track; - } else { - // No buffers for this track. Give it a few chances to - // fill a buffer, then remove it from active list. - if (--(track->mRetryCount) <= 0) { - LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); - trackToRemove = track; - } else { - mixerStatus = MIXER_TRACKS_ENABLED; - } - } - } - } - - // remove all the tracks that need to be... - if (UNLIKELY(trackToRemove != 0)) { - mActiveTracks.remove(trackToRemove); - if (!effectChains.isEmpty()) { - LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), trackToRemove->sessionId()); - effectChains[0]->stopTrack(); - } - if (trackToRemove->isTerminated()) { - mTracks.remove(trackToRemove); - deleteTrackName_l(trackToRemove->mName); - } - } - - lockEffectChains_l(); - } - - if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { - AudioBufferProvider::Buffer buffer; - size_t frameCount = mFrameCount; - curBuf = (int8_t *)mMixBuffer; - // output audio to hardware - while (frameCount) { - buffer.frameCount = frameCount; - activeTrack->getNextBuffer(&buffer); - if (UNLIKELY(buffer.raw == 0)) { - memset(curBuf, 0, frameCount * mFrameSize); - break; - } - memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); - frameCount -= buffer.frameCount; - curBuf += buffer.frameCount * mFrameSize; - activeTrack->releaseBuffer(&buffer); - } - sleepTime = 0; - standbyTime = systemTime() + standbyDelay; - } else { - if (sleepTime == 0) { - if (mixerStatus == MIXER_TRACKS_ENABLED) { - sleepTime = activeSleepTime; - } else { - sleepTime = idleSleepTime; - } - } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) { - memset (mMixBuffer, 0, mFrameCount * mFrameSize); - sleepTime = 0; - } - } - - if (mSuspended) { - sleepTime = idleSleepTime; - } - // sleepTime == 0 means we must write to audio hardware - if (sleepTime == 0) { - if (mixerStatus == MIXER_TRACKS_READY) { - applyVolume(leftVol, rightVol, rampVolume); - } - for (size_t i = 0; i < effectChains.size(); i ++) { - effectChains[i]->process_l(); - } - unlockEffectChains(); - - mLastWriteTime = systemTime(); - mInWrite = true; - mBytesWritten += mixBufferSize; - int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); - if (bytesWritten < 0) mBytesWritten -= mixBufferSize; - mNumWrites++; - mInWrite = false; - mStandby = false; - } else { - unlockEffectChains(); - usleep(sleepTime); - } - - // finally let go of removed track, without the lock held - // since we can't guarantee the destructors won't acquire that - // same lock. - trackToRemove.clear(); - activeTrack.clear(); - - // Effect chains will be actually deleted here if they were removed from - // mEffectChains list during mixing or effects processing - effectChains.clear(); - } - - if (!mStandby) { - mOutput->standby(); - } - - LOGV("DirectOutputThread %p exiting", this); - return false; -} - -// getTrackName_l() must be called with ThreadBase::mLock held -int AudioFlinger::DirectOutputThread::getTrackName_l() -{ - return 0; -} - -// deleteTrackName_l() must be called with ThreadBase::mLock held -void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) -{ -} - -// checkForNewParameters_l() must be called with ThreadBase::mLock held -bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() -{ - bool reconfig = false; - - while (!mNewParameters.isEmpty()) { - status_t status = NO_ERROR; - String8 keyValuePair = mNewParameters[0]; - AudioParameter param = AudioParameter(keyValuePair); - int value; - - if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { - // do not accept frame count changes if tracks are open as the track buffer - // size depends on frame count and correct behavior would not be garantied - // if frame count is changed after track creation - if (!mTracks.isEmpty()) { - status = INVALID_OPERATION; - } else { - reconfig = true; - } - } - if (status == NO_ERROR) { - status = mOutput->setParameters(keyValuePair); - if (!mStandby && status == INVALID_OPERATION) { - mOutput->standby(); - mStandby = true; - mBytesWritten = 0; - status = mOutput->setParameters(keyValuePair); - } - if (status == NO_ERROR && reconfig) { - readOutputParameters(); - sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); - } - } - - mNewParameters.removeAt(0); - - mParamStatus = status; - mParamCond.signal(); - mWaitWorkCV.wait(mLock); - } - return reconfig; -} - -uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() -{ - uint32_t time; - if (AudioSystem::isLinearPCM(mFormat)) { - time = (uint32_t)(mOutput->latency() * 1000) / 2; - } else { - time = 10000; - } - return time; -} - -uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() -{ - uint32_t time; - if (AudioSystem::isLinearPCM(mFormat)) { - time = (uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000; - } else { - time = 10000; - } - return time; -} - -// ---------------------------------------------------------------------------- - -AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) - : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) -{ - mType = PlaybackThread::DUPLICATING; - addOutputTrack(mainThread); -} - -AudioFlinger::DuplicatingThread::~DuplicatingThread() -{ - for (size_t i = 0; i < mOutputTracks.size(); i++) { - mOutputTracks[i]->destroy(); - } - mOutputTracks.clear(); -} - -bool AudioFlinger::DuplicatingThread::threadLoop() -{ - Vector< sp<Track> > tracksToRemove; - uint32_t mixerStatus = MIXER_IDLE; - nsecs_t standbyTime = systemTime(); - size_t mixBufferSize = mFrameCount*mFrameSize; - SortedVector< sp<OutputTrack> > outputTracks; - uint32_t writeFrames = 0; - uint32_t activeSleepTime = activeSleepTimeUs(); - uint32_t idleSleepTime = idleSleepTimeUs(); - uint32_t sleepTime = idleSleepTime; - Vector< sp<EffectChain> > effectChains; - - while (!exitPending()) - { - processConfigEvents(); - - mixerStatus = MIXER_IDLE; - { // scope for the mLock - - Mutex::Autolock _l(mLock); - - if (checkForNewParameters_l()) { - mixBufferSize = mFrameCount*mFrameSize; - updateWaitTime(); - activeSleepTime = activeSleepTimeUs(); - idleSleepTime = idleSleepTimeUs(); - } - - const SortedVector< wp<Track> >& activeTracks = mActiveTracks; - - for (size_t i = 0; i < mOutputTracks.size(); i++) { - outputTracks.add(mOutputTracks[i]); - } - - // put audio hardware into standby after short delay - if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || - mSuspended) { - if (!mStandby) { - for (size_t i = 0; i < outputTracks.size(); i++) { - outputTracks[i]->stop(); - } - mStandby = true; - mBytesWritten = 0; - } - - if (!activeTracks.size() && mConfigEvents.isEmpty()) { - // we're about to wait, flush the binder command buffer - IPCThreadState::self()->flushCommands(); - outputTracks.clear(); - - if (exitPending()) break; - - LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); - mWaitWorkCV.wait(mLock); - LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); - if (mMasterMute == false) { - char value[PROPERTY_VALUE_MAX]; - property_get("ro.audio.silent", value, "0"); - if (atoi(value)) { - LOGD("Silence is golden"); - setMasterMute(true); - } - } - - standbyTime = systemTime() + kStandbyTimeInNsecs; - sleepTime = idleSleepTime; - continue; - } - } - - mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); - - // prevent any changes in effect chain list and in each effect chain - // during mixing and effect process as the audio buffers could be deleted - // or modified if an effect is created or deleted - effectChains = mEffectChains; - lockEffectChains_l(); - } - - if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { - // mix buffers... - if (outputsReady(outputTracks)) { - mAudioMixer->process(); - } else { - memset(mMixBuffer, 0, mixBufferSize); - } - sleepTime = 0; - writeFrames = mFrameCount; - } else { - if (sleepTime == 0) { - if (mixerStatus == MIXER_TRACKS_ENABLED) { - sleepTime = activeSleepTime; - } else { - sleepTime = idleSleepTime; - } - } else if (mBytesWritten != 0) { - // flush remaining overflow buffers in output tracks - for (size_t i = 0; i < outputTracks.size(); i++) { - if (outputTracks[i]->isActive()) { - sleepTime = 0; - writeFrames = 0; - memset(mMixBuffer, 0, mixBufferSize); - break; - } - } - } - } - - if (mSuspended) { - sleepTime = idleSleepTime; - } - // sleepTime == 0 means we must write to audio hardware - if (sleepTime == 0) { - for (size_t i = 0; i < effectChains.size(); i ++) { - effectChains[i]->process_l(); - } - // enable changes in effect chain - unlockEffectChains(); - - standbyTime = systemTime() + kStandbyTimeInNsecs; - for (size_t i = 0; i < outputTracks.size(); i++) { - outputTracks[i]->write(mMixBuffer, writeFrames); - } - mStandby = false; - mBytesWritten += mixBufferSize; - } else { - // enable changes in effect chain - unlockEffectChains(); - usleep(sleepTime); - } - - // finally let go of all our tracks, without the lock held - // since we can't guarantee the destructors won't acquire that - // same lock. - tracksToRemove.clear(); - outputTracks.clear(); - - // Effect chains will be actually deleted here if they were removed from - // mEffectChains list during mixing or effects processing - effectChains.clear(); - } - - return false; -} - -void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) -{ - int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); - OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, - this, - mSampleRate, - mFormat, - mChannelCount, - frameCount); - if (outputTrack->cblk() != NULL) { - thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f); - mOutputTracks.add(outputTrack); - LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); - updateWaitTime(); - } -} - -void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) -{ - Mutex::Autolock _l(mLock); - for (size_t i = 0; i < mOutputTracks.size(); i++) { - if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { - mOutputTracks[i]->destroy(); - mOutputTracks.removeAt(i); - updateWaitTime(); - return; - } - } - LOGV("removeOutputTrack(): unkonwn thread: %p", thread); -} - -void AudioFlinger::DuplicatingThread::updateWaitTime() -{ - mWaitTimeMs = UINT_MAX; - for (size_t i = 0; i < mOutputTracks.size(); i++) { - sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); - if (strong != NULL) { - uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); - if (waitTimeMs < mWaitTimeMs) { - mWaitTimeMs = waitTimeMs; - } - } - } -} - - -bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) -{ - for (size_t i = 0; i < outputTracks.size(); i++) { - sp <ThreadBase> thread = outputTracks[i]->thread().promote(); - if (thread == 0) { - LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); - return false; - } - PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); - if (playbackThread->standby() && !playbackThread->isSuspended()) { - LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); - return false; - } - } - return true; -} - -uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() -{ - return (mWaitTimeMs * 1000) / 2; -} - -// ---------------------------------------------------------------------------- - -// TrackBase constructor must be called with AudioFlinger::mLock held -AudioFlinger::ThreadBase::TrackBase::TrackBase( - const wp<ThreadBase>& thread, - const sp<Client>& client, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount, - uint32_t flags, - const sp<IMemory>& sharedBuffer, - int sessionId) - : RefBase(), - mThread(thread), - mClient(client), - mCblk(0), - mFrameCount(0), - mState(IDLE), - mClientTid(-1), - mFormat(format), - mFlags(flags & ~SYSTEM_FLAGS_MASK), - mSessionId(sessionId) -{ - LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); - - // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); - size_t size = sizeof(audio_track_cblk_t); - size_t bufferSize = frameCount*channelCount*sizeof(int16_t); - if (sharedBuffer == 0) { - size += bufferSize; - } - - if (client != NULL) { - mCblkMemory = client->heap()->allocate(size); - if (mCblkMemory != 0) { - mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); - if (mCblk) { // construct the shared structure in-place. - new(mCblk) audio_track_cblk_t(); - // clear all buffers - mCblk->frameCount = frameCount; - mCblk->sampleRate = sampleRate; - mCblk->channelCount = (uint8_t)channelCount; - if (sharedBuffer == 0) { - mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); - memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); - // Force underrun condition to avoid false underrun callback until first data is - // written to buffer - mCblk->flags = CBLK_UNDERRUN_ON; - } else { - mBuffer = sharedBuffer->pointer(); - } - mBufferEnd = (uint8_t *)mBuffer + bufferSize; - } - } else { - LOGE("not enough memory for AudioTrack size=%u", size); - client->heap()->dump("AudioTrack"); - return; - } - } else { - mCblk = (audio_track_cblk_t *)(new uint8_t[size]); - if (mCblk) { // construct the shared structure in-place. - new(mCblk) audio_track_cblk_t(); - // clear all buffers - mCblk->frameCount = frameCount; - mCblk->sampleRate = sampleRate; - mCblk->channelCount = (uint8_t)channelCount; - mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); - memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); - // Force underrun condition to avoid false underrun callback until first data is - // written to buffer - mCblk->flags = CBLK_UNDERRUN_ON; - mBufferEnd = (uint8_t *)mBuffer + bufferSize; - } - } -} - -AudioFlinger::ThreadBase::TrackBase::~TrackBase() -{ - if (mCblk) { - mCblk->~audio_track_cblk_t(); // destroy our shared-structure. - if (mClient == NULL) { - delete mCblk; - } - } - mCblkMemory.clear(); // and free the shared memory - if (mClient != NULL) { - Mutex::Autolock _l(mClient->audioFlinger()->mLock); - mClient.clear(); - } -} - -void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) -{ - buffer->raw = 0; - mFrameCount = buffer->frameCount; - step(); - buffer->frameCount = 0; -} - -bool AudioFlinger::ThreadBase::TrackBase::step() { - bool result; - audio_track_cblk_t* cblk = this->cblk(); - - result = cblk->stepServer(mFrameCount); - if (!result) { - LOGV("stepServer failed acquiring cblk mutex"); - mFlags |= STEPSERVER_FAILED; - } - return result; -} - -void AudioFlinger::ThreadBase::TrackBase::reset() { - audio_track_cblk_t* cblk = this->cblk(); - - cblk->user = 0; - cblk->server = 0; - cblk->userBase = 0; - cblk->serverBase = 0; - mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); - LOGV("TrackBase::reset"); -} - -sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const -{ - return mCblkMemory; -} - -int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { - return (int)mCblk->sampleRate; -} - -int AudioFlinger::ThreadBase::TrackBase::channelCount() const { - return (int)mCblk->channelCount; -} - -void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { - audio_track_cblk_t* cblk = this->cblk(); - int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; - int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; - - // Check validity of returned pointer in case the track control block would have been corrupted. - if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || - ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { - LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ - server %d, serverBase %d, user %d, userBase %d, channelCount %d", - bufferStart, bufferEnd, mBuffer, mBufferEnd, - cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount); - return 0; - } - - return bufferStart; -} - -// ---------------------------------------------------------------------------- - -// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held -AudioFlinger::PlaybackThread::Track::Track( - const wp<ThreadBase>& thread, - const sp<Client>& client, - int streamType, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId) - : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId), - mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), mAuxEffectId(0) -{ - if (mCblk != NULL) { - sp<ThreadBase> baseThread = thread.promote(); - if (baseThread != 0) { - PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); - mName = playbackThread->getTrackName_l(); - mMainBuffer = playbackThread->mixBuffer(); - } - LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); - if (mName < 0) { - LOGE("no more track names available"); - } - mVolume[0] = 1.0f; - mVolume[1] = 1.0f; - mStreamType = streamType; - // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of - // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack - mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t); - } -} - -AudioFlinger::PlaybackThread::Track::~Track() -{ - LOGV("PlaybackThread::Track destructor"); - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - Mutex::Autolock _l(thread->mLock); - mState = TERMINATED; - } -} - -void AudioFlinger::PlaybackThread::Track::destroy() -{ - // NOTE: destroyTrack_l() can remove a strong reference to this Track - // by removing it from mTracks vector, so there is a risk that this Tracks's - // desctructor is called. As the destructor needs to lock mLock, - // we must acquire a strong reference on this Track before locking mLock - // here so that the destructor is called only when exiting this function. - // On the other hand, as long as Track::destroy() is only called by - // TrackHandle destructor, the TrackHandle still holds a strong ref on - // this Track with its member mTrack. - sp<Track> keep(this); - { // scope for mLock - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - if (!isOutputTrack()) { - if (mState == ACTIVE || mState == RESUMING) { - AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType); - } - AudioSystem::releaseOutput(thread->id()); - } - Mutex::Autolock _l(thread->mLock); - PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); - playbackThread->destroyTrack_l(this); - } - } -} - -void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) -{ - snprintf(buffer, size, " %05d %05d %03u %03u %03u %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", - mName - AudioMixer::TRACK0, - (mClient == NULL) ? getpid() : mClient->pid(), - mStreamType, - mFormat, - mCblk->channelCount, - mSessionId, - mFrameCount, - mState, - mMute, - mFillingUpStatus, - mCblk->sampleRate, - mCblk->volume[0], - mCblk->volume[1], - mCblk->server, - mCblk->user, - (int)mMainBuffer, - (int)mAuxBuffer); -} - -status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) -{ - audio_track_cblk_t* cblk = this->cblk(); - uint32_t framesReady; - uint32_t framesReq = buffer->frameCount; - - // Check if last stepServer failed, try to step now - if (mFlags & TrackBase::STEPSERVER_FAILED) { - if (!step()) goto getNextBuffer_exit; - LOGV("stepServer recovered"); - mFlags &= ~TrackBase::STEPSERVER_FAILED; - } - - framesReady = cblk->framesReady(); - - if (LIKELY(framesReady)) { - uint32_t s = cblk->server; - uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; - - bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; - if (framesReq > framesReady) { - framesReq = framesReady; - } - if (s + framesReq > bufferEnd) { - framesReq = bufferEnd - s; - } - - buffer->raw = getBuffer(s, framesReq); - if (buffer->raw == 0) goto getNextBuffer_exit; - - buffer->frameCount = framesReq; - return NO_ERROR; - } - -getNextBuffer_exit: - buffer->raw = 0; - buffer->frameCount = 0; - LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); - return NOT_ENOUGH_DATA; -} - -bool AudioFlinger::PlaybackThread::Track::isReady() const { - if (mFillingUpStatus != FS_FILLING) return true; - - if (mCblk->framesReady() >= mCblk->frameCount || - (mCblk->flags & CBLK_FORCEREADY_MSK)) { - mFillingUpStatus = FS_FILLED; - mCblk->flags &= ~CBLK_FORCEREADY_MSK; - return true; - } - return false; -} - -status_t AudioFlinger::PlaybackThread::Track::start() -{ - status_t status = NO_ERROR; - LOGV("start(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - Mutex::Autolock _l(thread->mLock); - int state = mState; - // here the track could be either new, or restarted - // in both cases "unstop" the track - if (mState == PAUSED) { - mState = TrackBase::RESUMING; - LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); - } else { - mState = TrackBase::ACTIVE; - LOGV("? => ACTIVE (%d) on thread %p", mName, this); - } - - if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { - thread->mLock.unlock(); - status = AudioSystem::startOutput(thread->id(), (AudioSystem::stream_type)mStreamType); - thread->mLock.lock(); - } - if (status == NO_ERROR) { - PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); - playbackThread->addTrack_l(this); - } else { - mState = state; - } - } else { - status = BAD_VALUE; - } - return status; -} - -void AudioFlinger::PlaybackThread::Track::stop() -{ - LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - Mutex::Autolock _l(thread->mLock); - int state = mState; - if (mState > STOPPED) { - mState = STOPPED; - // If the track is not active (PAUSED and buffers full), flush buffers - PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); - if (playbackThread->mActiveTracks.indexOf(this) < 0) { - reset(); - } - LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); - } - if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { - thread->mLock.unlock(); - AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType); - thread->mLock.lock(); - } - } -} - -void AudioFlinger::PlaybackThread::Track::pause() -{ - LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - Mutex::Autolock _l(thread->mLock); - if (mState == ACTIVE || mState == RESUMING) { - mState = PAUSING; - LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); - if (!isOutputTrack()) { - thread->mLock.unlock(); - AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType); - thread->mLock.lock(); - } - } - } -} - -void AudioFlinger::PlaybackThread::Track::flush() -{ - LOGV("flush(%d)", mName); - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - Mutex::Autolock _l(thread->mLock); - if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { - return; - } - // No point remaining in PAUSED state after a flush => go to - // STOPPED state - mState = STOPPED; - - mCblk->lock.lock(); - // NOTE: reset() will reset cblk->user and cblk->server with - // the risk that at the same time, the AudioMixer is trying to read - // data. In this case, getNextBuffer() would return a NULL pointer - // as audio buffer => the AudioMixer code MUST always test that pointer - // returned by getNextBuffer() is not NULL! - reset(); - mCblk->lock.unlock(); - } -} - -void AudioFlinger::PlaybackThread::Track::reset() -{ - // Do not reset twice to avoid discarding data written just after a flush and before - // the audioflinger thread detects the track is stopped. - if (!mResetDone) { - TrackBase::reset(); - // Force underrun condition to avoid false underrun callback until first data is - // written to buffer - mCblk->flags |= CBLK_UNDERRUN_ON; - mCblk->flags &= ~CBLK_FORCEREADY_MSK; - mFillingUpStatus = FS_FILLING; - mResetDone = true; - } -} - -void AudioFlinger::PlaybackThread::Track::mute(bool muted) -{ - mMute = muted; -} - -void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) -{ - mVolume[0] = left; - mVolume[1] = right; -} - -status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) -{ - status_t status = DEAD_OBJECT; - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); - status = playbackThread->attachAuxEffect(this, EffectId); - } - return status; -} - -void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) -{ - mAuxEffectId = EffectId; - mAuxBuffer = buffer; -} - -// ---------------------------------------------------------------------------- - -// RecordTrack constructor must be called with AudioFlinger::mLock held -AudioFlinger::RecordThread::RecordTrack::RecordTrack( - const wp<ThreadBase>& thread, - const sp<Client>& client, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount, - uint32_t flags, - int sessionId) - : TrackBase(thread, client, sampleRate, format, - channelCount, frameCount, flags, 0, sessionId), - mOverflow(false) -{ - if (mCblk != NULL) { - LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); - if (format == AudioSystem::PCM_16_BIT) { - mCblk->frameSize = channelCount * sizeof(int16_t); - } else if (format == AudioSystem::PCM_8_BIT) { - mCblk->frameSize = channelCount * sizeof(int8_t); - } else { - mCblk->frameSize = sizeof(int8_t); - } - } -} - -AudioFlinger::RecordThread::RecordTrack::~RecordTrack() -{ - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - AudioSystem::releaseInput(thread->id()); - } -} - -status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) -{ - audio_track_cblk_t* cblk = this->cblk(); - uint32_t framesAvail; - uint32_t framesReq = buffer->frameCount; - - // Check if last stepServer failed, try to step now - if (mFlags & TrackBase::STEPSERVER_FAILED) { - if (!step()) goto getNextBuffer_exit; - LOGV("stepServer recovered"); - mFlags &= ~TrackBase::STEPSERVER_FAILED; - } - - framesAvail = cblk->framesAvailable_l(); - - if (LIKELY(framesAvail)) { - uint32_t s = cblk->server; - uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; - - if (framesReq > framesAvail) { - framesReq = framesAvail; - } - if (s + framesReq > bufferEnd) { - framesReq = bufferEnd - s; - } - - buffer->raw = getBuffer(s, framesReq); - if (buffer->raw == 0) goto getNextBuffer_exit; - - buffer->frameCount = framesReq; - return NO_ERROR; - } - -getNextBuffer_exit: - buffer->raw = 0; - buffer->frameCount = 0; - return NOT_ENOUGH_DATA; -} - -status_t AudioFlinger::RecordThread::RecordTrack::start() -{ - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - RecordThread *recordThread = (RecordThread *)thread.get(); - return recordThread->start(this); - } else { - return BAD_VALUE; - } -} - -void AudioFlinger::RecordThread::RecordTrack::stop() -{ - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - RecordThread *recordThread = (RecordThread *)thread.get(); - recordThread->stop(this); - TrackBase::reset(); - // Force overerrun condition to avoid false overrun callback until first data is - // read from buffer - mCblk->flags |= CBLK_UNDERRUN_ON; - } -} - -void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) -{ - snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n", - (mClient == NULL) ? getpid() : mClient->pid(), - mFormat, - mCblk->channelCount, - mSessionId, - mFrameCount, - mState, - mCblk->sampleRate, - mCblk->server, - mCblk->user); -} - - -// ---------------------------------------------------------------------------- - -AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( - const wp<ThreadBase>& thread, - DuplicatingThread *sourceThread, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount) - : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0), - mActive(false), mSourceThread(sourceThread) -{ - - PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); - if (mCblk != NULL) { - mCblk->flags |= CBLK_DIRECTION_OUT; - mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); - mCblk->volume[0] = mCblk->volume[1] = 0x1000; - mOutBuffer.frameCount = 0; - playbackThread->mTracks.add(this); - LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p", - mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd); - } else { - LOGW("Error creating output track on thread %p", playbackThread); - } -} - -AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() -{ - clearBufferQueue(); -} - -status_t AudioFlinger::PlaybackThread::OutputTrack::start() -{ - status_t status = Track::start(); - if (status != NO_ERROR) { - return status; - } - - mActive = true; - mRetryCount = 127; - return status; -} - -void AudioFlinger::PlaybackThread::OutputTrack::stop() -{ - Track::stop(); - clearBufferQueue(); - mOutBuffer.frameCount = 0; - mActive = false; -} - -bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) -{ - Buffer *pInBuffer; - Buffer inBuffer; - uint32_t channelCount = mCblk->channelCount; - bool outputBufferFull = false; - inBuffer.frameCount = frames; - inBuffer.i16 = data; - - uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); - - if (!mActive && frames != 0) { - start(); - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - MixerThread *mixerThread = (MixerThread *)thread.get(); - if (mCblk->frameCount > frames){ - if (mBufferQueue.size() < kMaxOverFlowBuffers) { - uint32_t startFrames = (mCblk->frameCount - frames); - pInBuffer = new Buffer; - pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; - pInBuffer->frameCount = startFrames; - pInBuffer->i16 = pInBuffer->mBuffer; - memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); - mBufferQueue.add(pInBuffer); - } else { - LOGW ("OutputTrack::write() %p no more buffers in queue", this); - } - } - } - } - - while (waitTimeLeftMs) { - // First write pending buffers, then new data - if (mBufferQueue.size()) { - pInBuffer = mBufferQueue.itemAt(0); - } else { - pInBuffer = &inBuffer; - } - - if (pInBuffer->frameCount == 0) { - break; - } - - if (mOutBuffer.frameCount == 0) { - mOutBuffer.frameCount = pInBuffer->frameCount; - nsecs_t startTime = systemTime(); - if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { - LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); - outputBufferFull = true; - break; - } - uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); - if (waitTimeLeftMs >= waitTimeMs) { - waitTimeLeftMs -= waitTimeMs; - } else { - waitTimeLeftMs = 0; - } - } - - uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; - memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); - mCblk->stepUser(outFrames); - pInBuffer->frameCount -= outFrames; - pInBuffer->i16 += outFrames * channelCount; - mOutBuffer.frameCount -= outFrames; - mOutBuffer.i16 += outFrames * channelCount; - - if (pInBuffer->frameCount == 0) { - if (mBufferQueue.size()) { - mBufferQueue.removeAt(0); - delete [] pInBuffer->mBuffer; - delete pInBuffer; - LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); - } else { - break; - } - } - } - - // If we could not write all frames, allocate a buffer and queue it for next time. - if (inBuffer.frameCount) { - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0 && !thread->standby()) { - if (mBufferQueue.size() < kMaxOverFlowBuffers) { - pInBuffer = new Buffer; - pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; - pInBuffer->frameCount = inBuffer.frameCount; - pInBuffer->i16 = pInBuffer->mBuffer; - memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); - mBufferQueue.add(pInBuffer); - LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); - } else { - LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); - } - } - } - - // Calling write() with a 0 length buffer, means that no more data will be written: - // If no more buffers are pending, fill output track buffer to make sure it is started - // by output mixer. - if (frames == 0 && mBufferQueue.size() == 0) { - if (mCblk->user < mCblk->frameCount) { - frames = mCblk->frameCount - mCblk->user; - pInBuffer = new Buffer; - pInBuffer->mBuffer = new int16_t[frames * channelCount]; - pInBuffer->frameCount = frames; - pInBuffer->i16 = pInBuffer->mBuffer; - memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); - mBufferQueue.add(pInBuffer); - } else if (mActive) { - stop(); - } - } - - return outputBufferFull; -} - -status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) -{ - int active; - status_t result; - audio_track_cblk_t* cblk = mCblk; - uint32_t framesReq = buffer->frameCount; - -// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); - buffer->frameCount = 0; - - uint32_t framesAvail = cblk->framesAvailable(); - - - if (framesAvail == 0) { - Mutex::Autolock _l(cblk->lock); - goto start_loop_here; - while (framesAvail == 0) { - active = mActive; - if (UNLIKELY(!active)) { - LOGV("Not active and NO_MORE_BUFFERS"); - return AudioTrack::NO_MORE_BUFFERS; - } - result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); - if (result != NO_ERROR) { - return AudioTrack::NO_MORE_BUFFERS; - } - // read the server count again - start_loop_here: - framesAvail = cblk->framesAvailable_l(); - } - } - -// if (framesAvail < framesReq) { -// return AudioTrack::NO_MORE_BUFFERS; -// } - - if (framesReq > framesAvail) { - framesReq = framesAvail; - } - - uint32_t u = cblk->user; - uint32_t bufferEnd = cblk->userBase + cblk->frameCount; - - if (u + framesReq > bufferEnd) { - framesReq = bufferEnd - u; - } - - buffer->frameCount = framesReq; - buffer->raw = (void *)cblk->buffer(u); - return NO_ERROR; -} - - -void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() -{ - size_t size = mBufferQueue.size(); - Buffer *pBuffer; - - for (size_t i = 0; i < size; i++) { - pBuffer = mBufferQueue.itemAt(i); - delete [] pBuffer->mBuffer; - delete pBuffer; - } - mBufferQueue.clear(); -} - -// ---------------------------------------------------------------------------- - -AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) - : RefBase(), - mAudioFlinger(audioFlinger), - mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), - mPid(pid) -{ - // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer -} - -// Client destructor must be called with AudioFlinger::mLock held -AudioFlinger::Client::~Client() -{ - mAudioFlinger->removeClient_l(mPid); -} - -const sp<MemoryDealer>& AudioFlinger::Client::heap() const -{ - return mMemoryDealer; -} - -// ---------------------------------------------------------------------------- - -AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, - const sp<IAudioFlingerClient>& client, - pid_t pid) - : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) -{ -} - -AudioFlinger::NotificationClient::~NotificationClient() -{ - mClient.clear(); -} - -void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) -{ - sp<NotificationClient> keep(this); - { - mAudioFlinger->removeNotificationClient(mPid); - } -} - -// ---------------------------------------------------------------------------- - -AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) - : BnAudioTrack(), - mTrack(track) -{ -} - -AudioFlinger::TrackHandle::~TrackHandle() { - // just stop the track on deletion, associated resources - // will be freed from the main thread once all pending buffers have - // been played. Unless it's not in the active track list, in which - // case we free everything now... - mTrack->destroy(); -} - -status_t AudioFlinger::TrackHandle::start() { - return mTrack->start(); -} - -void AudioFlinger::TrackHandle::stop() { - mTrack->stop(); -} - -void AudioFlinger::TrackHandle::flush() { - mTrack->flush(); -} - -void AudioFlinger::TrackHandle::mute(bool e) { - mTrack->mute(e); -} - -void AudioFlinger::TrackHandle::pause() { - mTrack->pause(); -} - -void AudioFlinger::TrackHandle::setVolume(float left, float right) { - mTrack->setVolume(left, right); -} - -sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { - return mTrack->getCblk(); -} - -status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) -{ - return mTrack->attachAuxEffect(EffectId); -} - -status_t AudioFlinger::TrackHandle::onTransact( - uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) -{ - return BnAudioTrack::onTransact(code, data, reply, flags); -} - -// ---------------------------------------------------------------------------- - -sp<IAudioRecord> AudioFlinger::openRecord( - pid_t pid, - int input, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount, - uint32_t flags, - int *sessionId, - status_t *status) -{ - sp<RecordThread::RecordTrack> recordTrack; - sp<RecordHandle> recordHandle; - sp<Client> client; - wp<Client> wclient; - status_t lStatus; - RecordThread *thread; - size_t inFrameCount; - int lSessionId; - - // check calling permissions - if (!recordingAllowed()) { - lStatus = PERMISSION_DENIED; - goto Exit; - } - - // add client to list - { // scope for mLock - Mutex::Autolock _l(mLock); - thread = checkRecordThread_l(input); - if (thread == NULL) { - lStatus = BAD_VALUE; - goto Exit; - } - - wclient = mClients.valueFor(pid); - if (wclient != NULL) { - client = wclient.promote(); - } else { - client = new Client(this, pid); - mClients.add(pid, client); - } - - // If no audio session id is provided, create one here - if (sessionId != NULL && *sessionId != 0) { - lSessionId = *sessionId; - } else { - lSessionId = nextUniqueId(); - if (sessionId != NULL) { - *sessionId = lSessionId; - } - } - // create new record track. The record track uses one track in mHardwareMixerThread by convention. - recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate, - format, channelCount, frameCount, flags, lSessionId); - } - if (recordTrack->getCblk() == NULL) { - // remove local strong reference to Client before deleting the RecordTrack so that the Client - // destructor is called by the TrackBase destructor with mLock held - client.clear(); - recordTrack.clear(); - lStatus = NO_MEMORY; - goto Exit; - } - - // return to handle to client - recordHandle = new RecordHandle(recordTrack); - lStatus = NO_ERROR; - -Exit: - if (status) { - *status = lStatus; - } - return recordHandle; -} - -// ---------------------------------------------------------------------------- - -AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) - : BnAudioRecord(), - mRecordTrack(recordTrack) -{ -} - -AudioFlinger::RecordHandle::~RecordHandle() { - stop(); -} - -status_t AudioFlinger::RecordHandle::start() { - LOGV("RecordHandle::start()"); - return mRecordTrack->start(); -} - -void AudioFlinger::RecordHandle::stop() { - LOGV("RecordHandle::stop()"); - mRecordTrack->stop(); -} - -sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { - return mRecordTrack->getCblk(); -} - -status_t AudioFlinger::RecordHandle::onTransact( - uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) -{ - return BnAudioRecord::onTransact(code, data, reply, flags); -} - -// ---------------------------------------------------------------------------- - -AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) : - ThreadBase(audioFlinger, id), - mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) -{ - mReqChannelCount = AudioSystem::popCount(channels); - mReqSampleRate = sampleRate; - readInputParameters(); -} - - -AudioFlinger::RecordThread::~RecordThread() -{ - delete[] mRsmpInBuffer; - if (mResampler != 0) { - delete mResampler; - delete[] mRsmpOutBuffer; - } -} - -void AudioFlinger::RecordThread::onFirstRef() -{ - const size_t SIZE = 256; - char buffer[SIZE]; - - snprintf(buffer, SIZE, "Record Thread %p", this); - - run(buffer, PRIORITY_URGENT_AUDIO); -} - -bool AudioFlinger::RecordThread::threadLoop() -{ - AudioBufferProvider::Buffer buffer; - sp<RecordTrack> activeTrack; - - // start recording - while (!exitPending()) { - - processConfigEvents(); - - { // scope for mLock - Mutex::Autolock _l(mLock); - checkForNewParameters_l(); - if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { - if (!mStandby) { - mInput->standby(); - mStandby = true; - } - - if (exitPending()) break; - - LOGV("RecordThread: loop stopping"); - // go to sleep - mWaitWorkCV.wait(mLock); - LOGV("RecordThread: loop starting"); - continue; - } - if (mActiveTrack != 0) { - if (mActiveTrack->mState == TrackBase::PAUSING) { - if (!mStandby) { - mInput->standby(); - mStandby = true; - } - mActiveTrack.clear(); - mStartStopCond.broadcast(); - } else if (mActiveTrack->mState == TrackBase::RESUMING) { - if (mReqChannelCount != mActiveTrack->channelCount()) { - mActiveTrack.clear(); - mStartStopCond.broadcast(); - } else if (mBytesRead != 0) { - // record start succeeds only if first read from audio input - // succeeds - if (mBytesRead > 0) { - mActiveTrack->mState = TrackBase::ACTIVE; - } else { - mActiveTrack.clear(); - } - mStartStopCond.broadcast(); - } - mStandby = false; - } - } - } - - if (mActiveTrack != 0) { - if (mActiveTrack->mState != TrackBase::ACTIVE && - mActiveTrack->mState != TrackBase::RESUMING) { - usleep(5000); - continue; - } - buffer.frameCount = mFrameCount; - if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { - size_t framesOut = buffer.frameCount; - if (mResampler == 0) { - // no resampling - while (framesOut) { - size_t framesIn = mFrameCount - mRsmpInIndex; - if (framesIn) { - int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; - int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; - if (framesIn > framesOut) - framesIn = framesOut; - mRsmpInIndex += framesIn; - framesOut -= framesIn; - if ((int)mChannelCount == mReqChannelCount || - mFormat != AudioSystem::PCM_16_BIT) { - memcpy(dst, src, framesIn * mFrameSize); - } else { - int16_t *src16 = (int16_t *)src; - int16_t *dst16 = (int16_t *)dst; - if (mChannelCount == 1) { - while (framesIn--) { - *dst16++ = *src16; - *dst16++ = *src16++; - } - } else { - while (framesIn--) { - *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); - src16 += 2; - } - } - } - } - if (framesOut && mFrameCount == mRsmpInIndex) { - if (framesOut == mFrameCount && - ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) { - mBytesRead = mInput->read(buffer.raw, mInputBytes); - framesOut = 0; - } else { - mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); - mRsmpInIndex = 0; - } - if (mBytesRead < 0) { - LOGE("Error reading audio input"); - if (mActiveTrack->mState == TrackBase::ACTIVE) { - // Force input into standby so that it tries to - // recover at next read attempt - mInput->standby(); - usleep(5000); - } - mRsmpInIndex = mFrameCount; - framesOut = 0; - buffer.frameCount = 0; - } - } - } - } else { - // resampling - - memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); - // alter output frame count as if we were expecting stereo samples - if (mChannelCount == 1 && mReqChannelCount == 1) { - framesOut >>= 1; - } - mResampler->resample(mRsmpOutBuffer, framesOut, this); - // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() - // are 32 bit aligned which should be always true. - if (mChannelCount == 2 && mReqChannelCount == 1) { - AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); - // the resampler always outputs stereo samples: do post stereo to mono conversion - int16_t *src = (int16_t *)mRsmpOutBuffer; - int16_t *dst = buffer.i16; - while (framesOut--) { - *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); - src += 2; - } - } else { - AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); - } - - } - mActiveTrack->releaseBuffer(&buffer); - mActiveTrack->overflow(); - } - // client isn't retrieving buffers fast enough - else { - if (!mActiveTrack->setOverflow()) - LOGW("RecordThread: buffer overflow"); - // Release the processor for a while before asking for a new buffer. - // This will give the application more chance to read from the buffer and - // clear the overflow. - usleep(5000); - } - } - } - - if (!mStandby) { - mInput->standby(); - } - mActiveTrack.clear(); - - mStartStopCond.broadcast(); - - LOGV("RecordThread %p exiting", this); - return false; -} - -status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) -{ - LOGV("RecordThread::start"); - sp <ThreadBase> strongMe = this; - status_t status = NO_ERROR; - { - AutoMutex lock(&mLock); - if (mActiveTrack != 0) { - if (recordTrack != mActiveTrack.get()) { - status = -EBUSY; - } else if (mActiveTrack->mState == TrackBase::PAUSING) { - mActiveTrack->mState = TrackBase::ACTIVE; - } - return status; - } - - recordTrack->mState = TrackBase::IDLE; - mActiveTrack = recordTrack; - mLock.unlock(); - status_t status = AudioSystem::startInput(mId); - mLock.lock(); - if (status != NO_ERROR) { - mActiveTrack.clear(); - return status; - } - mActiveTrack->mState = TrackBase::RESUMING; - mRsmpInIndex = mFrameCount; - mBytesRead = 0; - // signal thread to start - LOGV("Signal record thread"); - mWaitWorkCV.signal(); - // do not wait for mStartStopCond if exiting - if (mExiting) { - mActiveTrack.clear(); - status = INVALID_OPERATION; - goto startError; - } - mStartStopCond.wait(mLock); - if (mActiveTrack == 0) { - LOGV("Record failed to start"); - status = BAD_VALUE; - goto startError; - } - LOGV("Record started OK"); - return status; - } -startError: - AudioSystem::stopInput(mId); - return status; -} - -void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { - LOGV("RecordThread::stop"); - sp <ThreadBase> strongMe = this; - { - AutoMutex lock(&mLock); - if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { - mActiveTrack->mState = TrackBase::PAUSING; - // do not wait for mStartStopCond if exiting - if (mExiting) { - return; - } - mStartStopCond.wait(mLock); - // if we have been restarted, recordTrack == mActiveTrack.get() here - if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { - mLock.unlock(); - AudioSystem::stopInput(mId); - mLock.lock(); - LOGV("Record stopped OK"); - } - } - } -} - -status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - pid_t pid = 0; - - snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); - result.append(buffer); - - if (mActiveTrack != 0) { - result.append("Active Track:\n"); - result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n"); - mActiveTrack->dump(buffer, SIZE); - result.append(buffer); - - snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); - result.append(buffer); - snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); - result.append(buffer); - snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); - result.append(buffer); - snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); - result.append(buffer); - snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); - result.append(buffer); - - - } else { - result.append("No record client\n"); - } - write(fd, result.string(), result.size()); - - dumpBase(fd, args); - - return NO_ERROR; -} - -status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) -{ - size_t framesReq = buffer->frameCount; - size_t framesReady = mFrameCount - mRsmpInIndex; - int channelCount; - - if (framesReady == 0) { - mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); - if (mBytesRead < 0) { - LOGE("RecordThread::getNextBuffer() Error reading audio input"); - if (mActiveTrack->mState == TrackBase::ACTIVE) { - // Force input into standby so that it tries to - // recover at next read attempt - mInput->standby(); - usleep(5000); - } - buffer->raw = 0; - buffer->frameCount = 0; - return NOT_ENOUGH_DATA; - } - mRsmpInIndex = 0; - framesReady = mFrameCount; - } - - if (framesReq > framesReady) { - framesReq = framesReady; - } - - if (mChannelCount == 1 && mReqChannelCount == 2) { - channelCount = 1; - } else { - channelCount = 2; - } - buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; - buffer->frameCount = framesReq; - return NO_ERROR; -} - -void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) -{ - mRsmpInIndex += buffer->frameCount; - buffer->frameCount = 0; -} - -bool AudioFlinger::RecordThread::checkForNewParameters_l() -{ - bool reconfig = false; - - while (!mNewParameters.isEmpty()) { - status_t status = NO_ERROR; - String8 keyValuePair = mNewParameters[0]; - AudioParameter param = AudioParameter(keyValuePair); - int value; - int reqFormat = mFormat; - int reqSamplingRate = mReqSampleRate; - int reqChannelCount = mReqChannelCount; - - if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { - reqSamplingRate = value; - reconfig = true; - } - if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { - reqFormat = value; - reconfig = true; - } - if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { - reqChannelCount = AudioSystem::popCount(value); - reconfig = true; - } - if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { - // do not accept frame count changes if tracks are open as the track buffer - // size depends on frame count and correct behavior would not be garantied - // if frame count is changed after track creation - if (mActiveTrack != 0) { - status = INVALID_OPERATION; - } else { - reconfig = true; - } - } - if (status == NO_ERROR) { - status = mInput->setParameters(keyValuePair); - if (status == INVALID_OPERATION) { - mInput->standby(); - status = mInput->setParameters(keyValuePair); - } - if (reconfig) { - if (status == BAD_VALUE && - reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT && - ((int)mInput->sampleRate() <= 2 * reqSamplingRate) && - (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) { - status = NO_ERROR; - } - if (status == NO_ERROR) { - readInputParameters(); - sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); - } - } - } - - mNewParameters.removeAt(0); - - mParamStatus = status; - mParamCond.signal(); - mWaitWorkCV.wait(mLock); - } - return reconfig; -} - -String8 AudioFlinger::RecordThread::getParameters(const String8& keys) -{ - return mInput->getParameters(keys); -} - -void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { - AudioSystem::OutputDescriptor desc; - void *param2 = 0; - - switch (event) { - case AudioSystem::INPUT_OPENED: - case AudioSystem::INPUT_CONFIG_CHANGED: - desc.channels = mChannels; - desc.samplingRate = mSampleRate; - desc.format = mFormat; - desc.frameCount = mFrameCount; - desc.latency = 0; - param2 = &desc; - break; - - case AudioSystem::INPUT_CLOSED: - default: - break; - } - mAudioFlinger->audioConfigChanged_l(event, mId, param2); -} - -void AudioFlinger::RecordThread::readInputParameters() -{ - if (mRsmpInBuffer) delete mRsmpInBuffer; - if (mRsmpOutBuffer) delete mRsmpOutBuffer; - if (mResampler) delete mResampler; - mResampler = 0; - - mSampleRate = mInput->sampleRate(); - mChannels = mInput->channels(); - mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); - mFormat = mInput->format(); - mFrameSize = (uint16_t)mInput->frameSize(); - mInputBytes = mInput->bufferSize(); - mFrameCount = mInputBytes / mFrameSize; - mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; - - if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) - { - int channelCount; - // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid - // stereo to mono post process as the resampler always outputs stereo. - if (mChannelCount == 1 && mReqChannelCount == 2) { - channelCount = 1; - } else { - channelCount = 2; - } - mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); - mResampler->setSampleRate(mSampleRate); - mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); - mRsmpOutBuffer = new int32_t[mFrameCount * 2]; - - // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples - if (mChannelCount == 1 && mReqChannelCount == 1) { - mFrameCount >>= 1; - } - - } - mRsmpInIndex = mFrameCount; -} - -unsigned int AudioFlinger::RecordThread::getInputFramesLost() -{ - return mInput->getInputFramesLost(); -} - -// ---------------------------------------------------------------------------- - -int AudioFlinger::openOutput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t *pLatencyMs, - uint32_t flags) -{ - status_t status; - PlaybackThread *thread = NULL; - mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; - uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; - uint32_t format = pFormat ? *pFormat : 0; - uint32_t channels = pChannels ? *pChannels : 0; - uint32_t latency = pLatencyMs ? *pLatencyMs : 0; - - LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", - pDevices ? *pDevices : 0, - samplingRate, - format, - channels, - flags); - - if (pDevices == NULL || *pDevices == 0) { - return 0; - } - Mutex::Autolock _l(mLock); - - AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices, - (int *)&format, - &channels, - &samplingRate, - &status); - LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", - output, - samplingRate, - format, - channels, - status); - - mHardwareStatus = AUDIO_HW_IDLE; - if (output != 0) { - int id = nextUniqueId(); - if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || - (format != AudioSystem::PCM_16_BIT) || - (channels != AudioSystem::CHANNEL_OUT_STEREO)) { - thread = new DirectOutputThread(this, output, id, *pDevices); - LOGV("openOutput() created direct output: ID %d thread %p", id, thread); - } else { - thread = new MixerThread(this, output, id, *pDevices); - LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); - -#ifdef LVMX - unsigned bitsPerSample = - (format == AudioSystem::PCM_16_BIT) ? 16 : - ((format == AudioSystem::PCM_8_BIT) ? 8 : 0); - unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1; - int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id()); - - LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount); - LifeVibes::setDevice(audioOutputType, *pDevices); -#endif - - } - mPlaybackThreads.add(id, thread); - - if (pSamplingRate) *pSamplingRate = samplingRate; - if (pFormat) *pFormat = format; - if (pChannels) *pChannels = channels; - if (pLatencyMs) *pLatencyMs = thread->latency(); - - // notify client processes of the new output creation - thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); - return id; - } - - return 0; -} - -int AudioFlinger::openDuplicateOutput(int output1, int output2) -{ - Mutex::Autolock _l(mLock); - MixerThread *thread1 = checkMixerThread_l(output1); - MixerThread *thread2 = checkMixerThread_l(output2); - - if (thread1 == NULL || thread2 == NULL) { - LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); - return 0; - } - - int id = nextUniqueId(); - DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); - thread->addOutputTrack(thread2); - mPlaybackThreads.add(id, thread); - // notify client processes of the new output creation - thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); - return id; -} - -status_t AudioFlinger::closeOutput(int output) -{ - // keep strong reference on the playback thread so that - // it is not destroyed while exit() is executed - sp <PlaybackThread> thread; - { - Mutex::Autolock _l(mLock); - thread = checkPlaybackThread_l(output); - if (thread == NULL) { - return BAD_VALUE; - } - - LOGV("closeOutput() %d", output); - - if (thread->type() == PlaybackThread::MIXER) { - for (size_t i = 0; i < mPlaybackThreads.size(); i++) { - if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) { - DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); - dupThread->removeOutputTrack((MixerThread *)thread.get()); - } - } - } - void *param2 = 0; - audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); - mPlaybackThreads.removeItem(output); - } - thread->exit(); - - if (thread->type() != PlaybackThread::DUPLICATING) { - mAudioHardware->closeOutputStream(thread->getOutput()); - } - return NO_ERROR; -} - -status_t AudioFlinger::suspendOutput(int output) -{ - Mutex::Autolock _l(mLock); - PlaybackThread *thread = checkPlaybackThread_l(output); - - if (thread == NULL) { - return BAD_VALUE; - } - - LOGV("suspendOutput() %d", output); - thread->suspend(); - - return NO_ERROR; -} - -status_t AudioFlinger::restoreOutput(int output) -{ - Mutex::Autolock _l(mLock); - PlaybackThread *thread = checkPlaybackThread_l(output); - - if (thread == NULL) { - return BAD_VALUE; - } - - LOGV("restoreOutput() %d", output); - - thread->restore(); - - return NO_ERROR; -} - -int AudioFlinger::openInput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t acoustics) -{ - status_t status; - RecordThread *thread = NULL; - uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; - uint32_t format = pFormat ? *pFormat : 0; - uint32_t channels = pChannels ? *pChannels : 0; - uint32_t reqSamplingRate = samplingRate; - uint32_t reqFormat = format; - uint32_t reqChannels = channels; - - if (pDevices == NULL || *pDevices == 0) { - return 0; - } - Mutex::Autolock _l(mLock); - - AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices, - (int *)&format, - &channels, - &samplingRate, - &status, - (AudioSystem::audio_in_acoustics)acoustics); - LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", - input, - samplingRate, - format, - channels, - acoustics, - status); - - // If the input could not be opened with the requested parameters and we can handle the conversion internally, - // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo - // or stereo to mono conversions on 16 bit PCM inputs. - if (input == 0 && status == BAD_VALUE && - reqFormat == format && format == AudioSystem::PCM_16_BIT && - (samplingRate <= 2 * reqSamplingRate) && - (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) { - LOGV("openInput() reopening with proposed sampling rate and channels"); - input = mAudioHardware->openInputStream(*pDevices, - (int *)&format, - &channels, - &samplingRate, - &status, - (AudioSystem::audio_in_acoustics)acoustics); - } - - if (input != 0) { - int id = nextUniqueId(); - // Start record thread - thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id); - mRecordThreads.add(id, thread); - LOGV("openInput() created record thread: ID %d thread %p", id, thread); - if (pSamplingRate) *pSamplingRate = reqSamplingRate; - if (pFormat) *pFormat = format; - if (pChannels) *pChannels = reqChannels; - - input->standby(); - - // notify client processes of the new input creation - thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); - return id; - } - - return 0; -} - -status_t AudioFlinger::closeInput(int input) -{ - // keep strong reference on the record thread so that - // it is not destroyed while exit() is executed - sp <RecordThread> thread; - { - Mutex::Autolock _l(mLock); - thread = checkRecordThread_l(input); - if (thread == NULL) { - return BAD_VALUE; - } - - LOGV("closeInput() %d", input); - void *param2 = 0; - audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); - mRecordThreads.removeItem(input); - } - thread->exit(); - - mAudioHardware->closeInputStream(thread->getInput()); - - return NO_ERROR; -} - -status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) -{ - Mutex::Autolock _l(mLock); - MixerThread *dstThread = checkMixerThread_l(output); - if (dstThread == NULL) { - LOGW("setStreamOutput() bad output id %d", output); - return BAD_VALUE; - } - - LOGV("setStreamOutput() stream %d to output %d", stream, output); - audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); - - for (size_t i = 0; i < mPlaybackThreads.size(); i++) { - PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); - if (thread != dstThread && - thread->type() != PlaybackThread::DIRECT) { - MixerThread *srcThread = (MixerThread *)thread; - srcThread->invalidateTracks(stream); - } - } - - return NO_ERROR; -} - - -int AudioFlinger::newAudioSessionId() -{ - return nextUniqueId(); -} - -// checkPlaybackThread_l() must be called with AudioFlinger::mLock held -AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const -{ - PlaybackThread *thread = NULL; - if (mPlaybackThreads.indexOfKey(output) >= 0) { - thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); - } - return thread; -} - -// checkMixerThread_l() must be called with AudioFlinger::mLock held -AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const -{ - PlaybackThread *thread = checkPlaybackThread_l(output); - if (thread != NULL) { - if (thread->type() == PlaybackThread::DIRECT) { - thread = NULL; - } - } - return (MixerThread *)thread; -} - -// checkRecordThread_l() must be called with AudioFlinger::mLock held -AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const -{ - RecordThread *thread = NULL; - if (mRecordThreads.indexOfKey(input) >= 0) { - thread = (RecordThread *)mRecordThreads.valueFor(input).get(); - } - return thread; -} - -int AudioFlinger::nextUniqueId() -{ - return android_atomic_inc(&mNextUniqueId); -} - -// ---------------------------------------------------------------------------- -// Effect management -// ---------------------------------------------------------------------------- - - -status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle) -{ - Mutex::Autolock _l(mLock); - return EffectLoadLibrary(libPath, handle); -} - -status_t AudioFlinger::unloadEffectLibrary(int handle) -{ - Mutex::Autolock _l(mLock); - return EffectUnloadLibrary(handle); -} - -status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) -{ - Mutex::Autolock _l(mLock); - return EffectQueryNumberEffects(numEffects); -} - -status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) -{ - Mutex::Autolock _l(mLock); - return EffectQueryEffect(index, descriptor); -} - -status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) -{ - Mutex::Autolock _l(mLock); - return EffectGetDescriptor(pUuid, descriptor); -} - - -// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp -static const effect_uuid_t VISUALIZATION_UUID_ = - {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; - -sp<IEffect> AudioFlinger::createEffect(pid_t pid, - effect_descriptor_t *pDesc, - const sp<IEffectClient>& effectClient, - int32_t priority, - int output, - int sessionId, - status_t *status, - int *id, - int *enabled) -{ - status_t lStatus = NO_ERROR; - sp<EffectHandle> handle; - effect_interface_t itfe; - effect_descriptor_t desc; - sp<Client> client; - wp<Client> wclient; - - LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d", pid, effectClient.get(), priority, sessionId, output); - - if (pDesc == NULL) { - lStatus = BAD_VALUE; - goto Exit; - } - - { - Mutex::Autolock _l(mLock); - - // check recording permission for visualizer - if (memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 || - memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) { - if (!recordingAllowed()) { - lStatus = PERMISSION_DENIED; - goto Exit; - } - } - - if (!EffectIsNullUuid(&pDesc->uuid)) { - // if uuid is specified, request effect descriptor - lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); - if (lStatus < 0) { - LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); - goto Exit; - } - } else { - // if uuid is not specified, look for an available implementation - // of the required type in effect factory - if (EffectIsNullUuid(&pDesc->type)) { - LOGW("createEffect() no effect type"); - lStatus = BAD_VALUE; - goto Exit; - } - uint32_t numEffects = 0; - effect_descriptor_t d; - bool found = false; - - lStatus = EffectQueryNumberEffects(&numEffects); - if (lStatus < 0) { - LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); - goto Exit; - } - for (uint32_t i = 0; i < numEffects; i++) { - lStatus = EffectQueryEffect(i, &desc); - if (lStatus < 0) { - LOGW("createEffect() error %d from EffectQueryEffect", lStatus); - continue; - } - if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { - // If matching type found save effect descriptor. If the session is - // 0 and the effect is not auxiliary, continue enumeration in case - // an auxiliary version of this effect type is available - found = true; - memcpy(&d, &desc, sizeof(effect_descriptor_t)); - if (sessionId != 0 || - (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - break; - } - } - } - if (!found) { - lStatus = BAD_VALUE; - LOGW("createEffect() effect not found"); - goto Exit; - } - // For same effect type, chose auxiliary version over insert version if - // connect to output mix (Compliance to OpenSL ES) - if (sessionId == 0 && - (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { - memcpy(&desc, &d, sizeof(effect_descriptor_t)); - } - } - - // Do not allow auxiliary effects on a session different from 0 (output mix) - if (sessionId != 0 && - (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - lStatus = INVALID_OPERATION; - goto Exit; - } - - // Session -1 is reserved for output stage effects that can only be created - // by audio policy manager (running in same process) - if (sessionId == -1 && getpid() != IPCThreadState::self()->getCallingPid()) { - lStatus = INVALID_OPERATION; - goto Exit; - } - - // return effect descriptor - memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); - - // If output is not specified try to find a matching audio session ID in one of the - // output threads. - // TODO: allow attachment of effect to inputs - if (output == 0) { - if (sessionId <= 0) { - // default to first output - // TODO: define criteria to choose output when not specified. Or - // receive output from audio policy manager - if (mPlaybackThreads.size() != 0) { - output = mPlaybackThreads.keyAt(0); - } - } else { - // look for the thread where the specified audio session is present - for (size_t i = 0; i < mPlaybackThreads.size(); i++) { - if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId)) { - output = mPlaybackThreads.keyAt(i); - break; - } - } - } - } - PlaybackThread *thread = checkPlaybackThread_l(output); - if (thread == NULL) { - LOGE("unknown output thread"); - lStatus = BAD_VALUE; - goto Exit; - } - - wclient = mClients.valueFor(pid); - - if (wclient != NULL) { - client = wclient.promote(); - } else { - client = new Client(this, pid); - mClients.add(pid, client); - } - - // create effect on selected output trhead - handle = thread->createEffect_l(client, effectClient, priority, sessionId, &desc, enabled, &lStatus); - if (handle != 0 && id != NULL) { - *id = handle->id(); - } - } - -Exit: - if(status) { - *status = lStatus; - } - return handle; -} - -status_t AudioFlinger::registerEffectResource_l(effect_descriptor_t *desc) { - if (mTotalEffectsCpuLoad + desc->cpuLoad > MAX_EFFECTS_CPU_LOAD) { - LOGW("registerEffectResource() CPU Load limit exceeded for Fx %s, CPU %f MIPS", - desc->name, (float)desc->cpuLoad/10); - return INVALID_OPERATION; - } - if (mTotalEffectsMemory + desc->memoryUsage > MAX_EFFECTS_MEMORY) { - LOGW("registerEffectResource() memory limit exceeded for Fx %s, Memory %d KB", - desc->name, desc->memoryUsage); - return INVALID_OPERATION; - } - mTotalEffectsCpuLoad += desc->cpuLoad; - mTotalEffectsMemory += desc->memoryUsage; - LOGV("registerEffectResource_l() effect %s, CPU %d, memory %d", - desc->name, desc->cpuLoad, desc->memoryUsage); - LOGV(" total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory); - return NO_ERROR; -} - -void AudioFlinger::unregisterEffectResource_l(effect_descriptor_t *desc) { - mTotalEffectsCpuLoad -= desc->cpuLoad; - mTotalEffectsMemory -= desc->memoryUsage; - LOGV("unregisterEffectResource_l() effect %s, CPU %d, memory %d", - desc->name, desc->cpuLoad, desc->memoryUsage); - LOGV(" total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory); -} - -// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held -sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l( - const sp<AudioFlinger::Client>& client, - const sp<IEffectClient>& effectClient, - int32_t priority, - int sessionId, - effect_descriptor_t *desc, - int *enabled, - status_t *status - ) -{ - sp<EffectModule> effect; - sp<EffectHandle> handle; - status_t lStatus; - sp<Track> track; - sp<EffectChain> chain; - bool effectCreated = false; - bool effectRegistered = false; - - if (mOutput == 0) { - LOGW("createEffect_l() Audio driver not initialized."); - lStatus = NO_INIT; - goto Exit; - } - - // Do not allow auxiliary effect on session other than 0 - if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY && - sessionId != 0) { - LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", desc->name, sessionId); - lStatus = BAD_VALUE; - goto Exit; - } - - // Do not allow effects with session ID 0 on direct output or duplicating threads - // TODO: add rule for hw accelerated effects on direct outputs with non PCM format - if (sessionId == 0 && mType != MIXER) { - LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", desc->name, sessionId); - lStatus = BAD_VALUE; - goto Exit; - } - - LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); - - { // scope for mLock - Mutex::Autolock _l(mLock); - - // check for existing effect chain with the requested audio session - chain = getEffectChain_l(sessionId); - if (chain == 0) { - // create a new chain for this session - LOGV("createEffect_l() new effect chain for session %d", sessionId); - chain = new EffectChain(this, sessionId); - addEffectChain_l(chain); - } else { - effect = chain->getEffectFromDesc(desc); - } - - LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); - - if (effect == 0) { - // Check CPU and memory usage - lStatus = mAudioFlinger->registerEffectResource_l(desc); - if (lStatus != NO_ERROR) { - goto Exit; - } - effectRegistered = true; - // create a new effect module if none present in the chain - effect = new EffectModule(this, chain, desc, mAudioFlinger->nextUniqueId(), sessionId); - lStatus = effect->status(); - if (lStatus != NO_ERROR) { - goto Exit; - } - lStatus = chain->addEffect(effect); - if (lStatus != NO_ERROR) { - goto Exit; - } - effectCreated = true; - - effect->setDevice(mDevice); - effect->setMode(mAudioFlinger->getMode()); - } - // create effect handle and connect it to effect module - handle = new EffectHandle(effect, client, effectClient, priority); - lStatus = effect->addHandle(handle); - if (enabled) { - *enabled = (int)effect->isEnabled(); - } - } - -Exit: - if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { - if (effectCreated) { - if (chain->removeEffect(effect) == 0) { - removeEffectChain_l(chain); - } - } - if (effectRegistered) { - mAudioFlinger->unregisterEffectResource_l(desc); - } - handle.clear(); - } - - if(status) { - *status = lStatus; - } - return handle; -} - -void AudioFlinger::PlaybackThread::disconnectEffect(const sp< EffectModule>& effect, - const wp<EffectHandle>& handle) { - effect_descriptor_t desc = effect->desc(); - Mutex::Autolock _l(mLock); - // delete the effect module if removing last handle on it - if (effect->removeHandle(handle) == 0) { - if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - detachAuxEffect_l(effect->id()); - } - sp<EffectChain> chain = effect->chain().promote(); - if (chain != 0) { - // remove effect chain if remove last effect - if (chain->removeEffect(effect) == 0) { - removeEffectChain_l(chain); - } - } - mLock.unlock(); - mAudioFlinger->mLock.lock(); - mAudioFlinger->unregisterEffectResource_l(&desc); - mAudioFlinger->mLock.unlock(); - } -} - -status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) -{ - int session = chain->sessionId(); - int16_t *buffer = mMixBuffer; - bool ownsBuffer = false; - - LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); - if (session > 0) { - // Only one effect chain can be present in direct output thread and it uses - // the mix buffer as input - if (mType != DIRECT) { - size_t numSamples = mFrameCount * mChannelCount; - buffer = new int16_t[numSamples]; - memset(buffer, 0, numSamples * sizeof(int16_t)); - LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); - ownsBuffer = true; - } - - // Attach all tracks with same session ID to this chain. - for (size_t i = 0; i < mTracks.size(); ++i) { - sp<Track> track = mTracks[i]; - if (session == track->sessionId()) { - LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); - track->setMainBuffer(buffer); - } - } - - // indicate all active tracks in the chain - for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { - sp<Track> track = mActiveTracks[i].promote(); - if (track == 0) continue; - if (session == track->sessionId()) { - LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); - chain->startTrack(); - } - } - } - - chain->setInBuffer(buffer, ownsBuffer); - chain->setOutBuffer(mMixBuffer); - // Effect chain for session -1 is inserted at end of effect chains list - // in order to be processed last as it contains output stage effects - // Effect chain for session 0 is inserted before session -1 to be processed - // after track specific effects and before output stage - // Effect chain for session other than 0 is inserted at beginning of effect - // chains list to be processed before output mix effects. Relative order between - // sessions other than 0 is not important - size_t size = mEffectChains.size(); - size_t i = 0; - for (i = 0; i < size; i++) { - if (mEffectChains[i]->sessionId() < session) break; - } - mEffectChains.insertAt(chain, i); - - return NO_ERROR; -} - -size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) -{ - int session = chain->sessionId(); - - LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); - - for (size_t i = 0; i < mEffectChains.size(); i++) { - if (chain == mEffectChains[i]) { - mEffectChains.removeAt(i); - // detach all tracks with same session ID from this chain - for (size_t i = 0; i < mTracks.size(); ++i) { - sp<Track> track = mTracks[i]; - if (session == track->sessionId()) { - track->setMainBuffer(mMixBuffer); - } - } - } - } - return mEffectChains.size(); -} - -void AudioFlinger::PlaybackThread::lockEffectChains_l() -{ - for (size_t i = 0; i < mEffectChains.size(); i++) { - mEffectChains[i]->lock(); - } -} - -void AudioFlinger::PlaybackThread::unlockEffectChains() -{ - Mutex::Autolock _l(mLock); - for (size_t i = 0; i < mEffectChains.size(); i++) { - mEffectChains[i]->unlock(); - } -} - -sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId) -{ - sp<EffectModule> effect; - - sp<EffectChain> chain = getEffectChain_l(sessionId); - if (chain != 0) { - effect = chain->getEffectFromId(effectId); - } - return effect; -} - -status_t AudioFlinger::PlaybackThread::attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) -{ - Mutex::Autolock _l(mLock); - return attachAuxEffect_l(track, EffectId); -} - -status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) -{ - status_t status = NO_ERROR; - - if (EffectId == 0) { - track->setAuxBuffer(0, NULL); - } else { - // Auxiliary effects are always in audio session 0 - sp<EffectModule> effect = getEffect_l(0, EffectId); - if (effect != 0) { - if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); - } else { - status = INVALID_OPERATION; - } - } else { - status = BAD_VALUE; - } - } - return status; -} - -void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) -{ - for (size_t i = 0; i < mTracks.size(); ++i) { - sp<Track> track = mTracks[i]; - if (track->auxEffectId() == effectId) { - attachAuxEffect_l(track, 0); - } - } -} - -// ---------------------------------------------------------------------------- -// EffectModule implementation -// ---------------------------------------------------------------------------- - -#undef LOG_TAG -#define LOG_TAG "AudioFlinger::EffectModule" - -AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, - const wp<AudioFlinger::EffectChain>& chain, - effect_descriptor_t *desc, - int id, - int sessionId) - : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), - mStatus(NO_INIT), mState(IDLE) -{ - LOGV("Constructor %p", this); - int lStatus; - sp<ThreadBase> thread = mThread.promote(); - if (thread == 0) { - return; - } - PlaybackThread *p = (PlaybackThread *)thread.get(); - - memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); - - // create effect engine from effect factory - mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface); - - if (mStatus != NO_ERROR) { - return; - } - lStatus = init(); - if (lStatus < 0) { - mStatus = lStatus; - goto Error; - } - - LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); - return; -Error: - EffectRelease(mEffectInterface); - mEffectInterface = NULL; - LOGV("Constructor Error %d", mStatus); -} - -AudioFlinger::EffectModule::~EffectModule() -{ - LOGV("Destructor %p", this); - if (mEffectInterface != NULL) { - // release effect engine - EffectRelease(mEffectInterface); - } -} - -status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) -{ - status_t status; - - Mutex::Autolock _l(mLock); - // First handle in mHandles has highest priority and controls the effect module - int priority = handle->priority(); - size_t size = mHandles.size(); - sp<EffectHandle> h; - size_t i; - for (i = 0; i < size; i++) { - h = mHandles[i].promote(); - if (h == 0) continue; - if (h->priority() <= priority) break; - } - // if inserted in first place, move effect control from previous owner to this handle - if (i == 0) { - if (h != 0) { - h->setControl(false, true); - } - handle->setControl(true, false); - status = NO_ERROR; - } else { - status = ALREADY_EXISTS; - } - mHandles.insertAt(handle, i); - return status; -} - -size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) -{ - Mutex::Autolock _l(mLock); - size_t size = mHandles.size(); - size_t i; - for (i = 0; i < size; i++) { - if (mHandles[i] == handle) break; - } - if (i == size) { - return size; - } - mHandles.removeAt(i); - size = mHandles.size(); - // if removed from first place, move effect control from this handle to next in line - if (i == 0 && size != 0) { - sp<EffectHandle> h = mHandles[0].promote(); - if (h != 0) { - h->setControl(true, true); - } - } - - return size; -} - -void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle) -{ - // keep a strong reference on this EffectModule to avoid calling the - // destructor before we exit - sp<EffectModule> keep(this); - { - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); - playbackThread->disconnectEffect(keep, handle); - } - } -} - -void AudioFlinger::EffectModule::updateState() { - Mutex::Autolock _l(mLock); - - switch (mState) { - case RESTART: - reset_l(); - // FALL THROUGH - - case STARTING: - // clear auxiliary effect input buffer for next accumulation - if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - memset(mConfig.inputCfg.buffer.raw, - 0, - mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); - } - start_l(); - mState = ACTIVE; - break; - case STOPPING: - stop_l(); - mDisableWaitCnt = mMaxDisableWaitCnt; - mState = STOPPED; - break; - case STOPPED: - // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the - // turn off sequence. - if (--mDisableWaitCnt == 0) { - reset_l(); - mState = IDLE; - } - break; - default: //IDLE , ACTIVE - break; - } -} - -void AudioFlinger::EffectModule::process() -{ - Mutex::Autolock _l(mLock); - - if (mEffectInterface == NULL || - mConfig.inputCfg.buffer.raw == NULL || - mConfig.outputCfg.buffer.raw == NULL) { - return; - } - - if (mState == ACTIVE || mState == STOPPING || mState == STOPPED) { - // do 32 bit to 16 bit conversion for auxiliary effect input buffer - if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, - mConfig.inputCfg.buffer.s32, - mConfig.inputCfg.buffer.frameCount); - } - - // do the actual processing in the effect engine - int ret = (*mEffectInterface)->process(mEffectInterface, - &mConfig.inputCfg.buffer, - &mConfig.outputCfg.buffer); - - // force transition to IDLE state when engine is ready - if (mState == STOPPED && ret == -ENODATA) { - mDisableWaitCnt = 1; - } - - // clear auxiliary effect input buffer for next accumulation - if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); - } - } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && - mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw){ - // If an insert effect is idle and input buffer is different from output buffer, copy input to - // output - sp<EffectChain> chain = mChain.promote(); - if (chain != 0 && chain->activeTracks() != 0) { - size_t size = mConfig.inputCfg.buffer.frameCount * sizeof(int16_t); - if (mConfig.inputCfg.channels == CHANNEL_STEREO) { - size *= 2; - } - memcpy(mConfig.outputCfg.buffer.raw, mConfig.inputCfg.buffer.raw, size); - } - } -} - -void AudioFlinger::EffectModule::reset_l() -{ - if (mEffectInterface == NULL) { - return; - } - (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); -} - -status_t AudioFlinger::EffectModule::configure() -{ - uint32_t channels; - if (mEffectInterface == NULL) { - return NO_INIT; - } - - sp<ThreadBase> thread = mThread.promote(); - if (thread == 0) { - return DEAD_OBJECT; - } - - // TODO: handle configuration of effects replacing track process - if (thread->channelCount() == 1) { - channels = CHANNEL_MONO; - } else { - channels = CHANNEL_STEREO; - } - - if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - mConfig.inputCfg.channels = CHANNEL_MONO; - } else { - mConfig.inputCfg.channels = channels; - } - mConfig.outputCfg.channels = channels; - mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15; - mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15; - mConfig.inputCfg.samplingRate = thread->sampleRate(); - mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; - mConfig.inputCfg.bufferProvider.cookie = NULL; - mConfig.inputCfg.bufferProvider.getBuffer = NULL; - mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; - mConfig.outputCfg.bufferProvider.cookie = NULL; - mConfig.outputCfg.bufferProvider.getBuffer = NULL; - mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; - mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; - // Insert effect: - // - in session 0 or -1, always overwrites output buffer: input buffer == output buffer - // - in other sessions: - // last effect in the chain accumulates in output buffer: input buffer != output buffer - // other effect: overwrites output buffer: input buffer == output buffer - // Auxiliary effect: - // accumulates in output buffer: input buffer != output buffer - // Therefore: accumulate <=> input buffer != output buffer - if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { - mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; - } else { - mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; - } - mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; - mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; - mConfig.inputCfg.buffer.frameCount = thread->frameCount(); - mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; - - status_t cmdStatus; - int size = sizeof(int); - status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_CONFIGURE, sizeof(effect_config_t), &mConfig, &size, &cmdStatus); - if (status == 0) { - status = cmdStatus; - } - - mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / - (1000 * mConfig.outputCfg.buffer.frameCount); - - return status; -} - -status_t AudioFlinger::EffectModule::init() -{ - Mutex::Autolock _l(mLock); - if (mEffectInterface == NULL) { - return NO_INIT; - } - status_t cmdStatus; - int size = sizeof(status_t); - status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_INIT, 0, NULL, &size, &cmdStatus); - if (status == 0) { - status = cmdStatus; - } - return status; -} - -status_t AudioFlinger::EffectModule::start_l() -{ - if (mEffectInterface == NULL) { - return NO_INIT; - } - status_t cmdStatus; - int size = sizeof(status_t); - status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_ENABLE, 0, NULL, &size, &cmdStatus); - if (status == 0) { - status = cmdStatus; - } - return status; -} - -status_t AudioFlinger::EffectModule::stop_l() -{ - if (mEffectInterface == NULL) { - return NO_INIT; - } - status_t cmdStatus; - int size = sizeof(status_t); - status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_DISABLE, 0, NULL, &size, &cmdStatus); - if (status == 0) { - status = cmdStatus; - } - return status; -} - -status_t AudioFlinger::EffectModule::command(int cmdCode, int cmdSize, void *pCmdData, int *replySize, void *pReplyData) -{ - Mutex::Autolock _l(mLock); -// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); - - if (mEffectInterface == NULL) { - return NO_INIT; - } - status_t status = (*mEffectInterface)->command(mEffectInterface, cmdCode, cmdSize, pCmdData, replySize, pReplyData); - if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { - int size = (replySize == NULL) ? 0 : *replySize; - for (size_t i = 1; i < mHandles.size(); i++) { - sp<EffectHandle> h = mHandles[i].promote(); - if (h != 0) { - h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); - } - } - } - return status; -} - -status_t AudioFlinger::EffectModule::setEnabled(bool enabled) -{ - Mutex::Autolock _l(mLock); - LOGV("setEnabled %p enabled %d", this, enabled); - - if (enabled != isEnabled()) { - switch (mState) { - // going from disabled to enabled - case IDLE: - mState = STARTING; - break; - case STOPPED: - mState = RESTART; - break; - case STOPPING: - mState = ACTIVE; - break; - - // going from enabled to disabled - case RESTART: - case STARTING: - mState = IDLE; - break; - case ACTIVE: - mState = STOPPING; - break; - } - for (size_t i = 1; i < mHandles.size(); i++) { - sp<EffectHandle> h = mHandles[i].promote(); - if (h != 0) { - h->setEnabled(enabled); - } - } - } - return NO_ERROR; -} - -bool AudioFlinger::EffectModule::isEnabled() -{ - switch (mState) { - case RESTART: - case STARTING: - case ACTIVE: - return true; - case IDLE: - case STOPPING: - case STOPPED: - default: - return false; - } -} - -status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) -{ - Mutex::Autolock _l(mLock); - status_t status = NO_ERROR; - - // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume - // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) - if ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) & (EFFECT_FLAG_VOLUME_CTRL|EFFECT_FLAG_VOLUME_IND)) { - status_t cmdStatus; - uint32_t volume[2]; - uint32_t *pVolume = NULL; - int size = sizeof(volume); - volume[0] = *left; - volume[1] = *right; - if (controller) { - pVolume = volume; - } - status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_VOLUME, size, volume, &size, pVolume); - if (controller && status == NO_ERROR && size == sizeof(volume)) { - *left = volume[0]; - *right = volume[1]; - } - } - return status; -} - -status_t AudioFlinger::EffectModule::setDevice(uint32_t device) -{ - Mutex::Autolock _l(mLock); - status_t status = NO_ERROR; - if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { - // convert device bit field from AudioSystem to EffectApi format. - device = deviceAudioSystemToEffectApi(device); - if (device == 0) { - return BAD_VALUE; - } - status_t cmdStatus; - int size = sizeof(status_t); - status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_DEVICE, sizeof(uint32_t), &device, &size, &cmdStatus); - if (status == NO_ERROR) { - status = cmdStatus; - } - } - return status; -} - -status_t AudioFlinger::EffectModule::setMode(uint32_t mode) -{ - Mutex::Autolock _l(mLock); - status_t status = NO_ERROR; - if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { - // convert audio mode from AudioSystem to EffectApi format. - int effectMode = modeAudioSystemToEffectApi(mode); - if (effectMode < 0) { - return BAD_VALUE; - } - status_t cmdStatus; - int size = sizeof(status_t); - status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_AUDIO_MODE, sizeof(int), &effectMode, &size, &cmdStatus); - if (status == NO_ERROR) { - status = cmdStatus; - } - } - return status; -} - -// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified -const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = { - DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE - DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER - DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET - DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE - DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO - DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET - DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT - DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP - DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES - DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER - DEVICE_AUX_DIGITAL // AudioSystem::DEVICE_OUT_AUX_DIGITAL -}; - -uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device) -{ - uint32_t deviceOut = 0; - while (device) { - const uint32_t i = 31 - __builtin_clz(device); - device &= ~(1 << i); - if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) { - LOGE("device convertion error for AudioSystem device 0x%08x", device); - return 0; - } - deviceOut |= (uint32_t)sDeviceConvTable[i]; - } - return deviceOut; -} - -// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified -const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = { - AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL - AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE - AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_CALL -}; - -int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode) -{ - int modeOut = -1; - if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) { - modeOut = (int)sModeConvTable[mode]; - } - return modeOut; -} - -status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); - result.append(buffer); - - bool locked = tryLock(mLock); - // failed to lock - AudioFlinger is probably deadlocked - if (!locked) { - result.append("\t\tCould not lock Fx mutex:\n"); - } - - result.append("\t\tSession Status State Engine:\n"); - snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", - mSessionId, mStatus, mState, (uint32_t)mEffectInterface); - result.append(buffer); - - result.append("\t\tDescriptor:\n"); - snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", - mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, - mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], - mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); - result.append(buffer); - snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", - mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, - mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], - mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); - result.append(buffer); - snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n", - mDescriptor.apiVersion, - mDescriptor.flags); - result.append(buffer); - snprintf(buffer, SIZE, "\t\t- name: %s\n", - mDescriptor.name); - result.append(buffer); - snprintf(buffer, SIZE, "\t\t- implementor: %s\n", - mDescriptor.implementor); - result.append(buffer); - - result.append("\t\t- Input configuration:\n"); - result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); - snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", - (uint32_t)mConfig.inputCfg.buffer.raw, - mConfig.inputCfg.buffer.frameCount, - mConfig.inputCfg.samplingRate, - mConfig.inputCfg.channels, - mConfig.inputCfg.format); - result.append(buffer); - - result.append("\t\t- Output configuration:\n"); - result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); - snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", - (uint32_t)mConfig.outputCfg.buffer.raw, - mConfig.outputCfg.buffer.frameCount, - mConfig.outputCfg.samplingRate, - mConfig.outputCfg.channels, - mConfig.outputCfg.format); - result.append(buffer); - - snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); - result.append(buffer); - result.append("\t\t\tPid Priority Ctrl Locked client server\n"); - for (size_t i = 0; i < mHandles.size(); ++i) { - sp<EffectHandle> handle = mHandles[i].promote(); - if (handle != 0) { - handle->dump(buffer, SIZE); - result.append(buffer); - } - } - - result.append("\n"); - - write(fd, result.string(), result.length()); - - if (locked) { - mLock.unlock(); - } - - return NO_ERROR; -} - -// ---------------------------------------------------------------------------- -// EffectHandle implementation -// ---------------------------------------------------------------------------- - -#undef LOG_TAG -#define LOG_TAG "AudioFlinger::EffectHandle" - -AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, - const sp<AudioFlinger::Client>& client, - const sp<IEffectClient>& effectClient, - int32_t priority) - : BnEffect(), - mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false) -{ - LOGV("constructor %p", this); - - int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); - mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); - if (mCblkMemory != 0) { - mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); - - if (mCblk) { - new(mCblk) effect_param_cblk_t(); - mBuffer = (uint8_t *)mCblk + bufOffset; - } - } else { - LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); - return; - } -} - -AudioFlinger::EffectHandle::~EffectHandle() -{ - LOGV("Destructor %p", this); - disconnect(); -} - -status_t AudioFlinger::EffectHandle::enable() -{ - if (!mHasControl) return INVALID_OPERATION; - if (mEffect == 0) return DEAD_OBJECT; - - return mEffect->setEnabled(true); -} - -status_t AudioFlinger::EffectHandle::disable() -{ - if (!mHasControl) return INVALID_OPERATION; - if (mEffect == NULL) return DEAD_OBJECT; - - return mEffect->setEnabled(false); -} - -void AudioFlinger::EffectHandle::disconnect() -{ - if (mEffect == 0) { - return; - } - mEffect->disconnect(this); - // release sp on module => module destructor can be called now - mEffect.clear(); - if (mCblk) { - mCblk->~effect_param_cblk_t(); // destroy our shared-structure. - } - mCblkMemory.clear(); // and free the shared memory - if (mClient != 0) { - Mutex::Autolock _l(mClient->audioFlinger()->mLock); - mClient.clear(); - } -} - -status_t AudioFlinger::EffectHandle::command(int cmdCode, int cmdSize, void *pCmdData, int *replySize, void *pReplyData) -{ -// LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); - - // only get parameter command is permitted for applications not controlling the effect - if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { - return INVALID_OPERATION; - } - if (mEffect == 0) return DEAD_OBJECT; - - // handle commands that are not forwarded transparently to effect engine - if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { - // No need to trylock() here as this function is executed in the binder thread serving a particular client process: - // no risk to block the whole media server process or mixer threads is we are stuck here - Mutex::Autolock _l(mCblk->lock); - if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || - mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { - mCblk->serverIndex = 0; - mCblk->clientIndex = 0; - return BAD_VALUE; - } - status_t status = NO_ERROR; - while (mCblk->serverIndex < mCblk->clientIndex) { - int reply; - int rsize = sizeof(int); - int *p = (int *)(mBuffer + mCblk->serverIndex); - int size = *p++; - if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { - LOGW("command(): invalid parameter block size"); - break; - } - effect_param_t *param = (effect_param_t *)p; - if (param->psize == 0 || param->vsize == 0) { - LOGW("command(): null parameter or value size"); - mCblk->serverIndex += size; - continue; - } - int psize = sizeof(effect_param_t) + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize; - status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, psize, p, &rsize, &reply); - if (ret == NO_ERROR) { - if (reply != NO_ERROR) { - status = reply; - } - } else { - status = ret; - } - mCblk->serverIndex += size; - } - mCblk->serverIndex = 0; - mCblk->clientIndex = 0; - return status; - } else if (cmdCode == EFFECT_CMD_ENABLE) { - return enable(); - } else if (cmdCode == EFFECT_CMD_DISABLE) { - return disable(); - } - - return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); -} - -sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { - return mCblkMemory; -} - -void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal) -{ - LOGV("setControl %p control %d", this, hasControl); - - mHasControl = hasControl; - if (signal && mEffectClient != 0) { - mEffectClient->controlStatusChanged(hasControl); - } -} - -void AudioFlinger::EffectHandle::commandExecuted(int cmdCode, int cmdSize, void *pCmdData, int replySize, void *pReplyData) -{ - if (mEffectClient != 0) { - mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); - } -} - - - -void AudioFlinger::EffectHandle::setEnabled(bool enabled) -{ - if (mEffectClient != 0) { - mEffectClient->enableStatusChanged(enabled); - } -} - -status_t AudioFlinger::EffectHandle::onTransact( - uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) -{ - return BnEffect::onTransact(code, data, reply, flags); -} - - -void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) -{ - bool locked = tryLock(mCblk->lock); - - snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", - (mClient == NULL) ? getpid() : mClient->pid(), - mPriority, - mHasControl, - !locked, - mCblk->clientIndex, - mCblk->serverIndex - ); - - if (locked) { - mCblk->lock.unlock(); - } -} - -#undef LOG_TAG -#define LOG_TAG "AudioFlinger::EffectChain" - -AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, - int sessionId) - : mThread(wThread), mSessionId(sessionId), mVolumeCtrlIdx(-1), mActiveTrackCnt(0), mOwnInBuffer(false) -{ - -} - -AudioFlinger::EffectChain::~EffectChain() -{ - if (mOwnInBuffer) { - delete mInBuffer; - } - -} - -sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc(effect_descriptor_t *descriptor) -{ - sp<EffectModule> effect; - size_t size = mEffects.size(); - - for (size_t i = 0; i < size; i++) { - if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { - effect = mEffects[i]; - break; - } - } - return effect; -} - -sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId(int id) -{ - sp<EffectModule> effect; - size_t size = mEffects.size(); - - for (size_t i = 0; i < size; i++) { - if (mEffects[i]->id() == id) { - effect = mEffects[i]; - break; - } - } - return effect; -} - -// Must be called with EffectChain::mLock locked -void AudioFlinger::EffectChain::process_l() -{ - size_t size = mEffects.size(); - for (size_t i = 0; i < size; i++) { - mEffects[i]->process(); - } - for (size_t i = 0; i < size; i++) { - mEffects[i]->updateState(); - } - // if no track is active, input buffer must be cleared here as the mixer process - // will not do it - if (mSessionId > 0 && activeTracks() == 0) { - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - size_t numSamples = thread->frameCount() * thread->channelCount(); - memset(mInBuffer, 0, numSamples * sizeof(int16_t)); - } - } -} - -status_t AudioFlinger::EffectChain::addEffect(sp<EffectModule>& effect) -{ - effect_descriptor_t desc = effect->desc(); - uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; - - Mutex::Autolock _l(mLock); - - if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - // Auxiliary effects are inserted at the beginning of mEffects vector as - // they are processed first and accumulated in chain input buffer - mEffects.insertAt(effect, 0); - sp<ThreadBase> thread = mThread.promote(); - if (thread == 0) { - return NO_INIT; - } - // the input buffer for auxiliary effect contains mono samples in - // 32 bit format. This is to avoid saturation in AudoMixer - // accumulation stage. Saturation is done in EffectModule::process() before - // calling the process in effect engine - size_t numSamples = thread->frameCount(); - int32_t *buffer = new int32_t[numSamples]; - memset(buffer, 0, numSamples * sizeof(int32_t)); - effect->setInBuffer((int16_t *)buffer); - // auxiliary effects output samples to chain input buffer for further processing - // by insert effects - effect->setOutBuffer(mInBuffer); - } else { - // Insert effects are inserted at the end of mEffects vector as they are processed - // after track and auxiliary effects. - // Insert effect order as a function of indicated preference: - // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if - // another effect is present - // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the - // last effect claiming first position - // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the - // first effect claiming last position - // else if EFFECT_FLAG_INSERT_ANY insert after first or before last - // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is - // already present - - int size = (int)mEffects.size(); - int idx_insert = size; - int idx_insert_first = -1; - int idx_insert_last = -1; - - for (int i = 0; i < size; i++) { - effect_descriptor_t d = mEffects[i]->desc(); - uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; - uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; - if (iMode == EFFECT_FLAG_TYPE_INSERT) { - // check invalid effect chaining combinations - if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || - iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { - LOGW("addEffect() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); - return INVALID_OPERATION; - } - // remember position of first insert effect and by default - // select this as insert position for new effect - if (idx_insert == size) { - idx_insert = i; - } - // remember position of last insert effect claiming - // first position - if (iPref == EFFECT_FLAG_INSERT_FIRST) { - idx_insert_first = i; - } - // remember position of first insert effect claiming - // last position - if (iPref == EFFECT_FLAG_INSERT_LAST && - idx_insert_last == -1) { - idx_insert_last = i; - } - } - } - - // modify idx_insert from first position if needed - if (insertPref == EFFECT_FLAG_INSERT_LAST) { - if (idx_insert_last != -1) { - idx_insert = idx_insert_last; - } else { - idx_insert = size; - } - } else { - if (idx_insert_first != -1) { - idx_insert = idx_insert_first + 1; - } - } - - // always read samples from chain input buffer - effect->setInBuffer(mInBuffer); - - // if last effect in the chain, output samples to chain - // output buffer, otherwise to chain input buffer - if (idx_insert == size) { - if (idx_insert != 0) { - mEffects[idx_insert-1]->setOutBuffer(mInBuffer); - mEffects[idx_insert-1]->configure(); - } - effect->setOutBuffer(mOutBuffer); - } else { - effect->setOutBuffer(mInBuffer); - } - mEffects.insertAt(effect, idx_insert); - // Always give volume control to last effect in chain with volume control capability - if (((desc.flags & EFFECT_FLAG_VOLUME_MASK) & EFFECT_FLAG_VOLUME_CTRL) && - mVolumeCtrlIdx < idx_insert) { - mVolumeCtrlIdx = idx_insert; - } - - LOGV("addEffect() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); - } - effect->configure(); - return NO_ERROR; -} - -size_t AudioFlinger::EffectChain::removeEffect(const sp<EffectModule>& effect) -{ - Mutex::Autolock _l(mLock); - - int size = (int)mEffects.size(); - int i; - uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; - - for (i = 0; i < size; i++) { - if (effect == mEffects[i]) { - if (type == EFFECT_FLAG_TYPE_AUXILIARY) { - delete[] effect->inBuffer(); - } else { - if (i == size - 1 && i != 0) { - mEffects[i - 1]->setOutBuffer(mOutBuffer); - mEffects[i - 1]->configure(); - } - } - mEffects.removeAt(i); - LOGV("removeEffect() effect %p, removed from chain %p at rank %d", effect.get(), this, i); - break; - } - } - // Return volume control to last effect in chain with volume control capability - if (mVolumeCtrlIdx == i) { - size = (int)mEffects.size(); - for (i = size; i > 0; i--) { - if ((mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) & EFFECT_FLAG_VOLUME_CTRL) { - break; - } - } - // mVolumeCtrlIdx reset to -1 if no effect found with volume control flag set - mVolumeCtrlIdx = i - 1; - } - - return mEffects.size(); -} - -void AudioFlinger::EffectChain::setDevice(uint32_t device) -{ - size_t size = mEffects.size(); - for (size_t i = 0; i < size; i++) { - mEffects[i]->setDevice(device); - } -} - -void AudioFlinger::EffectChain::setMode(uint32_t mode) -{ - size_t size = mEffects.size(); - for (size_t i = 0; i < size; i++) { - mEffects[i]->setMode(mode); - } -} - -bool AudioFlinger::EffectChain::setVolume(uint32_t *left, uint32_t *right) -{ - uint32_t newLeft = *left; - uint32_t newRight = *right; - bool hasControl = false; - - // first get volume update from volume controller - if (mVolumeCtrlIdx >= 0) { - mEffects[mVolumeCtrlIdx]->setVolume(&newLeft, &newRight, true); - hasControl = true; - } - // then indicate volume to all other effects in chain. - // Pass altered volume to effects before volume controller - // and requested volume to effects after controller - uint32_t lVol = newLeft; - uint32_t rVol = newRight; - size_t size = mEffects.size(); - for (size_t i = 0; i < size; i++) { - if ((int)i == mVolumeCtrlIdx) continue; - // this also works for mVolumeCtrlIdx == -1 when there is no volume controller - if ((int)i > mVolumeCtrlIdx) { - lVol = *left; - rVol = *right; - } - mEffects[i]->setVolume(&lVol, &rVol, false); - } - *left = newLeft; - *right = newRight; - - return hasControl; -} - -sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getVolumeController() -{ - sp<EffectModule> effect; - if (mVolumeCtrlIdx >= 0) { - effect = mEffects[mVolumeCtrlIdx]; - } - return effect; -} - - -status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); - result.append(buffer); - - bool locked = tryLock(mLock); - // failed to lock - AudioFlinger is probably deadlocked - if (!locked) { - result.append("\tCould not lock mutex:\n"); - } - - result.append("\tNum fx In buffer Out buffer Vol ctrl Active tracks:\n"); - snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %02d %d\n", - mEffects.size(), - (uint32_t)mInBuffer, - (uint32_t)mOutBuffer, - (mVolumeCtrlIdx == -1) ? 0 : mEffects[mVolumeCtrlIdx]->id(), - mActiveTrackCnt); - result.append(buffer); - write(fd, result.string(), result.size()); - - for (size_t i = 0; i < mEffects.size(); ++i) { - sp<EffectModule> effect = mEffects[i]; - if (effect != 0) { - effect->dump(fd, args); - } - } - - if (locked) { - mLock.unlock(); - } - - return NO_ERROR; -} - -#undef LOG_TAG -#define LOG_TAG "AudioFlinger" - -// ---------------------------------------------------------------------------- - -status_t AudioFlinger::onTransact( - uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) -{ - return BnAudioFlinger::onTransact(code, data, reply, flags); -} - -// ---------------------------------------------------------------------------- - -void AudioFlinger::instantiate() { - defaultServiceManager()->addService( - String16("media.audio_flinger"), new AudioFlinger()); -} - -}; // namespace android diff --git a/libs/audioflinger/AudioFlinger.h b/libs/audioflinger/AudioFlinger.h deleted file mode 100644 index 507c9ac..0000000 --- a/libs/audioflinger/AudioFlinger.h +++ /dev/null @@ -1,1148 +0,0 @@ -/* //device/include/server/AudioFlinger/AudioFlinger.h -** -** Copyright 2007, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#ifndef ANDROID_AUDIO_FLINGER_H -#define ANDROID_AUDIO_FLINGER_H - -#include <stdint.h> -#include <sys/types.h> -#include <limits.h> - -#include <media/IAudioFlinger.h> -#include <media/IAudioFlingerClient.h> -#include <media/IAudioTrack.h> -#include <media/IAudioRecord.h> -#include <media/AudioTrack.h> - -#include <utils/Atomic.h> -#include <utils/Errors.h> -#include <utils/threads.h> -#include <binder/MemoryDealer.h> -#include <utils/SortedVector.h> -#include <utils/Vector.h> - -#include <hardware_legacy/AudioHardwareInterface.h> - -#include "AudioBufferProvider.h" - -namespace android { - -class audio_track_cblk_t; -class effect_param_cblk_t; -class AudioMixer; -class AudioBuffer; -class AudioResampler; - - -// ---------------------------------------------------------------------------- - -#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true )) -#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false )) - - -// ---------------------------------------------------------------------------- - -static const nsecs_t kStandbyTimeInNsecs = seconds(3); - -class AudioFlinger : public BnAudioFlinger -{ -public: - static void instantiate(); - - virtual status_t dump(int fd, const Vector<String16>& args); - - // IAudioFlinger interface - virtual sp<IAudioTrack> createTrack( - pid_t pid, - int streamType, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount, - uint32_t flags, - const sp<IMemory>& sharedBuffer, - int output, - int *sessionId, - status_t *status); - - virtual uint32_t sampleRate(int output) const; - virtual int channelCount(int output) const; - virtual int format(int output) const; - virtual size_t frameCount(int output) const; - virtual uint32_t latency(int output) const; - - virtual status_t setMasterVolume(float value); - virtual status_t setMasterMute(bool muted); - - virtual float masterVolume() const; - virtual bool masterMute() const; - - virtual status_t setStreamVolume(int stream, float value, int output); - virtual status_t setStreamMute(int stream, bool muted); - - virtual float streamVolume(int stream, int output) const; - virtual bool streamMute(int stream) const; - - virtual status_t setMode(int mode); - - virtual status_t setMicMute(bool state); - virtual bool getMicMute() const; - - virtual bool isStreamActive(int stream) const; - - virtual status_t setParameters(int ioHandle, const String8& keyValuePairs); - virtual String8 getParameters(int ioHandle, const String8& keys); - - virtual void registerClient(const sp<IAudioFlingerClient>& client); - - virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount); - virtual unsigned int getInputFramesLost(int ioHandle); - - virtual int openOutput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t *pLatencyMs, - uint32_t flags); - - virtual int openDuplicateOutput(int output1, int output2); - - virtual status_t closeOutput(int output); - - virtual status_t suspendOutput(int output); - - virtual status_t restoreOutput(int output); - - virtual int openInput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t acoustics); - - virtual status_t closeInput(int input); - - virtual status_t setStreamOutput(uint32_t stream, int output); - - virtual status_t setVoiceVolume(float volume); - - virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output); - - virtual int newAudioSessionId(); - - virtual status_t loadEffectLibrary(const char *libPath, int *handle); - - virtual status_t unloadEffectLibrary(int handle); - - virtual status_t queryNumberEffects(uint32_t *numEffects); - - virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor); - - virtual status_t getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor); - - virtual sp<IEffect> createEffect(pid_t pid, - effect_descriptor_t *pDesc, - const sp<IEffectClient>& effectClient, - int32_t priority, - int output, - int sessionId, - status_t *status, - int *id, - int *enabled); - - status_t registerEffectResource_l(effect_descriptor_t *desc); - void unregisterEffectResource_l(effect_descriptor_t *desc); - - enum hardware_call_state { - AUDIO_HW_IDLE = 0, - AUDIO_HW_INIT, - AUDIO_HW_OUTPUT_OPEN, - AUDIO_HW_OUTPUT_CLOSE, - AUDIO_HW_INPUT_OPEN, - AUDIO_HW_INPUT_CLOSE, - AUDIO_HW_STANDBY, - AUDIO_HW_SET_MASTER_VOLUME, - AUDIO_HW_GET_ROUTING, - AUDIO_HW_SET_ROUTING, - AUDIO_HW_GET_MODE, - AUDIO_HW_SET_MODE, - AUDIO_HW_GET_MIC_MUTE, - AUDIO_HW_SET_MIC_MUTE, - AUDIO_SET_VOICE_VOLUME, - AUDIO_SET_PARAMETER, - }; - - // record interface - virtual sp<IAudioRecord> openRecord( - pid_t pid, - int input, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount, - uint32_t flags, - int *sessionId, - status_t *status); - - virtual status_t onTransact( - uint32_t code, - const Parcel& data, - Parcel* reply, - uint32_t flags); - - uint32_t getMode() { return mMode; } - -private: - AudioFlinger(); - virtual ~AudioFlinger(); - - - // Internal dump utilites. - status_t dumpPermissionDenial(int fd, const Vector<String16>& args); - status_t dumpClients(int fd, const Vector<String16>& args); - status_t dumpInternals(int fd, const Vector<String16>& args); - - // --- Client --- - class Client : public RefBase { - public: - Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); - virtual ~Client(); - const sp<MemoryDealer>& heap() const; - pid_t pid() const { return mPid; } - sp<AudioFlinger> audioFlinger() { return mAudioFlinger; } - - private: - Client(const Client&); - Client& operator = (const Client&); - sp<AudioFlinger> mAudioFlinger; - sp<MemoryDealer> mMemoryDealer; - pid_t mPid; - }; - - // --- Notification Client --- - class NotificationClient : public IBinder::DeathRecipient { - public: - NotificationClient(const sp<AudioFlinger>& audioFlinger, - const sp<IAudioFlingerClient>& client, - pid_t pid); - virtual ~NotificationClient(); - - sp<IAudioFlingerClient> client() { return mClient; } - - // IBinder::DeathRecipient - virtual void binderDied(const wp<IBinder>& who); - - private: - NotificationClient(const NotificationClient&); - NotificationClient& operator = (const NotificationClient&); - - sp<AudioFlinger> mAudioFlinger; - pid_t mPid; - sp<IAudioFlingerClient> mClient; - }; - - class TrackHandle; - class RecordHandle; - class RecordThread; - class PlaybackThread; - class MixerThread; - class DirectOutputThread; - class DuplicatingThread; - class Track; - class RecordTrack; - class EffectModule; - class EffectHandle; - class EffectChain; - - class ThreadBase : public Thread { - public: - ThreadBase (const sp<AudioFlinger>& audioFlinger, int id); - virtual ~ThreadBase(); - - status_t dumpBase(int fd, const Vector<String16>& args); - - // base for record and playback - class TrackBase : public AudioBufferProvider, public RefBase { - - public: - enum track_state { - IDLE, - TERMINATED, - STOPPED, - RESUMING, - ACTIVE, - PAUSING, - PAUSED - }; - - enum track_flags { - STEPSERVER_FAILED = 0x01, // StepServer could not acquire cblk->lock mutex - SYSTEM_FLAGS_MASK = 0x0000ffffUL, - // The upper 16 bits are used for track-specific flags. - }; - - TrackBase(const wp<ThreadBase>& thread, - const sp<Client>& client, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount, - uint32_t flags, - const sp<IMemory>& sharedBuffer, - int sessionId); - ~TrackBase(); - - virtual status_t start() = 0; - virtual void stop() = 0; - sp<IMemory> getCblk() const; - audio_track_cblk_t* cblk() const { return mCblk; } - int sessionId() { return mSessionId; } - - protected: - friend class ThreadBase; - friend class RecordHandle; - friend class PlaybackThread; - friend class RecordThread; - friend class MixerThread; - friend class DirectOutputThread; - - TrackBase(const TrackBase&); - TrackBase& operator = (const TrackBase&); - - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0; - virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); - - int format() const { - return mFormat; - } - - int channelCount() const ; - - int sampleRate() const; - - void* getBuffer(uint32_t offset, uint32_t frames) const; - - bool isStopped() const { - return mState == STOPPED; - } - - bool isTerminated() const { - return mState == TERMINATED; - } - - bool step(); - void reset(); - - wp<ThreadBase> mThread; - sp<Client> mClient; - sp<IMemory> mCblkMemory; - audio_track_cblk_t* mCblk; - void* mBuffer; - void* mBufferEnd; - uint32_t mFrameCount; - // we don't really need a lock for these - int mState; - int mClientTid; - uint8_t mFormat; - uint32_t mFlags; - int mSessionId; - }; - - class ConfigEvent { - public: - ConfigEvent() : mEvent(0), mParam(0) {} - - int mEvent; - int mParam; - }; - - uint32_t sampleRate() const; - int channelCount() const; - int format() const; - size_t frameCount() const; - void wakeUp() { mWaitWorkCV.broadcast(); } - void exit(); - virtual bool checkForNewParameters_l() = 0; - virtual status_t setParameters(const String8& keyValuePairs); - virtual String8 getParameters(const String8& keys) = 0; - virtual void audioConfigChanged_l(int event, int param = 0) = 0; - void sendConfigEvent(int event, int param = 0); - void sendConfigEvent_l(int event, int param = 0); - void processConfigEvents(); - int id() const { return mId;} - bool standby() { return mStandby; } - - mutable Mutex mLock; - - protected: - - friend class Track; - friend class TrackBase; - friend class PlaybackThread; - friend class MixerThread; - friend class DirectOutputThread; - friend class DuplicatingThread; - friend class RecordThread; - friend class RecordTrack; - - Condition mWaitWorkCV; - sp<AudioFlinger> mAudioFlinger; - uint32_t mSampleRate; - size_t mFrameCount; - uint32_t mChannels; - uint16_t mChannelCount; - uint16_t mFrameSize; - int mFormat; - Condition mParamCond; - Vector<String8> mNewParameters; - status_t mParamStatus; - Vector<ConfigEvent *> mConfigEvents; - bool mStandby; - int mId; - bool mExiting; - }; - - // --- PlaybackThread --- - class PlaybackThread : public ThreadBase { - public: - - enum type { - MIXER, - DIRECT, - DUPLICATING - }; - - enum mixer_state { - MIXER_IDLE, - MIXER_TRACKS_ENABLED, - MIXER_TRACKS_READY - }; - - // playback track - class Track : public TrackBase { - public: - Track( const wp<ThreadBase>& thread, - const sp<Client>& client, - int streamType, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId); - ~Track(); - - void dump(char* buffer, size_t size); - virtual status_t start(); - virtual void stop(); - void pause(); - - void flush(); - void destroy(); - void mute(bool); - void setVolume(float left, float right); - int name() const { - return mName; - } - - int type() const { - return mStreamType; - } - status_t attachAuxEffect(int EffectId); - void setAuxBuffer(int EffectId, int32_t *buffer); - int32_t *auxBuffer() { return mAuxBuffer; } - void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } - int16_t *mainBuffer() { return mMainBuffer; } - int auxEffectId() { return mAuxEffectId; } - - - protected: - friend class ThreadBase; - friend class AudioFlinger; - friend class TrackHandle; - friend class PlaybackThread; - friend class MixerThread; - friend class DirectOutputThread; - - Track(const Track&); - Track& operator = (const Track&); - - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); - bool isMuted() { return mMute; } - bool isPausing() const { - return mState == PAUSING; - } - bool isPaused() const { - return mState == PAUSED; - } - bool isReady() const; - void setPaused() { mState = PAUSED; } - void reset(); - - bool isOutputTrack() const { - return (mStreamType == AudioSystem::NUM_STREAM_TYPES); - } - - // we don't really need a lock for these - float mVolume[2]; - volatile bool mMute; - // FILLED state is used for suppressing volume ramp at begin of playing - enum {FS_FILLING, FS_FILLED, FS_ACTIVE}; - mutable uint8_t mFillingUpStatus; - int8_t mRetryCount; - sp<IMemory> mSharedBuffer; - bool mResetDone; - int mStreamType; - int mName; - int16_t *mMainBuffer; - int32_t *mAuxBuffer; - int mAuxEffectId; - }; // end of Track - - - // playback track - class OutputTrack : public Track { - public: - - class Buffer: public AudioBufferProvider::Buffer { - public: - int16_t *mBuffer; - }; - - OutputTrack( const wp<ThreadBase>& thread, - DuplicatingThread *sourceThread, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount); - ~OutputTrack(); - - virtual status_t start(); - virtual void stop(); - bool write(int16_t* data, uint32_t frames); - bool bufferQueueEmpty() { return (mBufferQueue.size() == 0) ? true : false; } - bool isActive() { return mActive; } - wp<ThreadBase>& thread() { return mThread; } - - private: - - status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); - void clearBufferQueue(); - - // Maximum number of pending buffers allocated by OutputTrack::write() - static const uint8_t kMaxOverFlowBuffers = 10; - - Vector < Buffer* > mBufferQueue; - AudioBufferProvider::Buffer mOutBuffer; - bool mActive; - DuplicatingThread* mSourceThread; - }; // end of OutputTrack - - PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device); - virtual ~PlaybackThread(); - - virtual status_t dump(int fd, const Vector<String16>& args); - - // Thread virtuals - virtual status_t readyToRun(); - virtual void onFirstRef(); - - virtual uint32_t latency() const; - - virtual status_t setMasterVolume(float value); - virtual status_t setMasterMute(bool muted); - - virtual float masterVolume() const; - virtual bool masterMute() const; - - virtual status_t setStreamVolume(int stream, float value); - virtual status_t setStreamMute(int stream, bool muted); - - virtual float streamVolume(int stream) const; - virtual bool streamMute(int stream) const; - - bool isStreamActive(int stream) const; - - sp<Track> createTrack_l( - const sp<AudioFlinger::Client>& client, - int streamType, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId, - status_t *status); - - AudioStreamOut* getOutput() { return mOutput; } - - virtual int type() const { return mType; } - void suspend() { mSuspended++; } - void restore() { if (mSuspended) mSuspended--; } - bool isSuspended() { return (mSuspended != 0); } - virtual String8 getParameters(const String8& keys); - virtual void audioConfigChanged_l(int event, int param = 0); - virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); - int16_t *mixBuffer() { return mMixBuffer; }; - - sp<EffectHandle> createEffect_l( - const sp<AudioFlinger::Client>& client, - const sp<IEffectClient>& effectClient, - int32_t priority, - int sessionId, - effect_descriptor_t *desc, - int *enabled, - status_t *status); - void disconnectEffect(const sp< EffectModule>& effect, - const wp<EffectHandle>& handle); - - bool hasAudioSession(int sessionId); - sp<EffectChain> getEffectChain(int sessionId); - sp<EffectChain> getEffectChain_l(int sessionId); - status_t addEffectChain_l(const sp<EffectChain>& chain); - size_t removeEffectChain_l(const sp<EffectChain>& chain); - void lockEffectChains_l(); - void unlockEffectChains(); - - sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); - void detachAuxEffect_l(int effectId); - status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId); - status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId); - void setMode(uint32_t mode); - - struct stream_type_t { - stream_type_t() - : volume(1.0f), - mute(false) - { - } - float volume; - bool mute; - }; - - protected: - int mType; - int16_t* mMixBuffer; - int mSuspended; - int mBytesWritten; - bool mMasterMute; - SortedVector< wp<Track> > mActiveTracks; - - virtual int getTrackName_l() = 0; - virtual void deleteTrackName_l(int name) = 0; - virtual uint32_t activeSleepTimeUs() = 0; - virtual uint32_t idleSleepTimeUs() = 0; - - private: - - friend class AudioFlinger; - friend class OutputTrack; - friend class Track; - friend class TrackBase; - friend class MixerThread; - friend class DirectOutputThread; - friend class DuplicatingThread; - - PlaybackThread(const Client&); - PlaybackThread& operator = (const PlaybackThread&); - - status_t addTrack_l(const sp<Track>& track); - void destroyTrack_l(const sp<Track>& track); - - void readOutputParameters(); - - uint32_t device() { return mDevice; } - - virtual status_t dumpInternals(int fd, const Vector<String16>& args); - status_t dumpTracks(int fd, const Vector<String16>& args); - status_t dumpEffectChains(int fd, const Vector<String16>& args); - - SortedVector< sp<Track> > mTracks; - // mStreamTypes[] uses 1 additionnal stream type internally for the OutputTrack used by DuplicatingThread - stream_type_t mStreamTypes[AudioSystem::NUM_STREAM_TYPES + 1]; - AudioStreamOut* mOutput; - float mMasterVolume; - nsecs_t mLastWriteTime; - int mNumWrites; - int mNumDelayedWrites; - bool mInWrite; - Vector< sp<EffectChain> > mEffectChains; - uint32_t mDevice; - }; - - class MixerThread : public PlaybackThread { - public: - MixerThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device); - virtual ~MixerThread(); - - // Thread virtuals - virtual bool threadLoop(); - - void invalidateTracks(int streamType); - virtual bool checkForNewParameters_l(); - virtual status_t dumpInternals(int fd, const Vector<String16>& args); - - protected: - uint32_t prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove); - virtual int getTrackName_l(); - virtual void deleteTrackName_l(int name); - virtual uint32_t activeSleepTimeUs(); - virtual uint32_t idleSleepTimeUs(); - - AudioMixer* mAudioMixer; - }; - - class DirectOutputThread : public PlaybackThread { - public: - - DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device); - ~DirectOutputThread(); - - // Thread virtuals - virtual bool threadLoop(); - - virtual bool checkForNewParameters_l(); - - protected: - virtual int getTrackName_l(); - virtual void deleteTrackName_l(int name); - virtual uint32_t activeSleepTimeUs(); - virtual uint32_t idleSleepTimeUs(); - - private: - void applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp); - - float mLeftVolFloat; - float mRightVolFloat; - uint16_t mLeftVolShort; - uint16_t mRightVolShort; - }; - - class DuplicatingThread : public MixerThread { - public: - DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, int id); - ~DuplicatingThread(); - - // Thread virtuals - virtual bool threadLoop(); - void addOutputTrack(MixerThread* thread); - void removeOutputTrack(MixerThread* thread); - uint32_t waitTimeMs() { return mWaitTimeMs; } - protected: - virtual uint32_t activeSleepTimeUs(); - - private: - bool outputsReady(SortedVector< sp<OutputTrack> > &outputTracks); - void updateWaitTime(); - - SortedVector < sp<OutputTrack> > mOutputTracks; - uint32_t mWaitTimeMs; - }; - - PlaybackThread *checkPlaybackThread_l(int output) const; - MixerThread *checkMixerThread_l(int output) const; - RecordThread *checkRecordThread_l(int input) const; - float streamVolumeInternal(int stream) const { return mStreamTypes[stream].volume; } - void audioConfigChanged_l(int event, int ioHandle, void *param2); - - int nextUniqueId(); - - friend class AudioBuffer; - - class TrackHandle : public android::BnAudioTrack { - public: - TrackHandle(const sp<PlaybackThread::Track>& track); - virtual ~TrackHandle(); - virtual status_t start(); - virtual void stop(); - virtual void flush(); - virtual void mute(bool); - virtual void pause(); - virtual void setVolume(float left, float right); - virtual sp<IMemory> getCblk() const; - virtual status_t attachAuxEffect(int effectId); - virtual status_t onTransact( - uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); - private: - sp<PlaybackThread::Track> mTrack; - }; - - friend class Client; - friend class PlaybackThread::Track; - - - void removeClient_l(pid_t pid); - void removeNotificationClient(pid_t pid); - - - // record thread - class RecordThread : public ThreadBase, public AudioBufferProvider - { - public: - - // record track - class RecordTrack : public TrackBase { - public: - RecordTrack(const wp<ThreadBase>& thread, - const sp<Client>& client, - uint32_t sampleRate, - int format, - int channelCount, - int frameCount, - uint32_t flags, - int sessionId); - ~RecordTrack(); - - virtual status_t start(); - virtual void stop(); - - bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; } - bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } - - void dump(char* buffer, size_t size); - private: - friend class AudioFlinger; - friend class RecordThread; - - RecordTrack(const RecordTrack&); - RecordTrack& operator = (const RecordTrack&); - - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); - - bool mOverflow; - }; - - - RecordThread(const sp<AudioFlinger>& audioFlinger, - AudioStreamIn *input, - uint32_t sampleRate, - uint32_t channels, - int id); - ~RecordThread(); - - virtual bool threadLoop(); - virtual status_t readyToRun() { return NO_ERROR; } - virtual void onFirstRef(); - - status_t start(RecordTrack* recordTrack); - void stop(RecordTrack* recordTrack); - status_t dump(int fd, const Vector<String16>& args); - AudioStreamIn* getInput() { return mInput; } - - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); - virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); - virtual bool checkForNewParameters_l(); - virtual String8 getParameters(const String8& keys); - virtual void audioConfigChanged_l(int event, int param = 0); - void readInputParameters(); - virtual unsigned int getInputFramesLost(); - - private: - RecordThread(); - AudioStreamIn *mInput; - sp<RecordTrack> mActiveTrack; - Condition mStartStopCond; - AudioResampler *mResampler; - int32_t *mRsmpOutBuffer; - int16_t *mRsmpInBuffer; - size_t mRsmpInIndex; - size_t mInputBytes; - int mReqChannelCount; - uint32_t mReqSampleRate; - ssize_t mBytesRead; - }; - - class RecordHandle : public android::BnAudioRecord { - public: - RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); - virtual ~RecordHandle(); - virtual status_t start(); - virtual void stop(); - virtual sp<IMemory> getCblk() const; - virtual status_t onTransact( - uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); - private: - sp<RecordThread::RecordTrack> mRecordTrack; - }; - - //--- Audio Effect Management - - // EffectModule and EffectChain classes both have their own mutex to protect - // state changes or resource modifications. Always respect the following order - // if multiple mutexes must be acquired to avoid cross deadlock: - // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule - - // The EffectModule class is a wrapper object controlling the effect engine implementation - // in the effect library. It prevents concurrent calls to process() and command() functions - // from different client threads. It keeps a list of EffectHandle objects corresponding - // to all client applications using this effect and notifies applications of effect state, - // control or parameter changes. It manages the activation state machine to send appropriate - // reset, enable, disable commands to effect engine and provide volume - // ramping when effects are activated/deactivated. - // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by - // the attached track(s) to accumulate their auxiliary channel. - class EffectModule: public RefBase { - public: - EffectModule(const wp<ThreadBase>& wThread, - const wp<AudioFlinger::EffectChain>& chain, - effect_descriptor_t *desc, - int id, - int sessionId); - ~EffectModule(); - - enum effect_state { - IDLE, - RESTART, - STARTING, - ACTIVE, - STOPPING, - STOPPED - }; - - int id() { return mId; } - void process(); - void updateState(); - status_t command(int cmdCode, int cmdSize, void *pCmdData, int *replySize, void *pReplyData); - - void reset_l(); - status_t configure(); - status_t init(); - uint32_t state() { - return mState; - } - uint32_t status() { - return mStatus; - } - status_t setEnabled(bool enabled); - bool isEnabled(); - - void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } - int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } - void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } - int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } - - status_t addHandle(sp<EffectHandle>& handle); - void disconnect(const wp<EffectHandle>& handle); - size_t removeHandle (const wp<EffectHandle>& handle); - - effect_descriptor_t& desc() { return mDescriptor; } - wp<EffectChain>& chain() { return mChain; } - - status_t setDevice(uint32_t device); - status_t setVolume(uint32_t *left, uint32_t *right, bool controller); - status_t setMode(uint32_t mode); - - status_t dump(int fd, const Vector<String16>& args); - - protected: - - // Maximum time allocated to effect engines to complete the turn off sequence - static const uint32_t MAX_DISABLE_TIME_MS = 10000; - - EffectModule(const EffectModule&); - EffectModule& operator = (const EffectModule&); - - status_t start_l(); - status_t stop_l(); - - // update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified - static const uint32_t sDeviceConvTable[]; - static uint32_t deviceAudioSystemToEffectApi(uint32_t device); - - // update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified - static const uint32_t sModeConvTable[]; - static int modeAudioSystemToEffectApi(uint32_t mode); - - Mutex mLock; // mutex for process, commands and handles list protection - wp<ThreadBase> mThread; // parent thread - wp<EffectChain> mChain; // parent effect chain - int mId; // this instance unique ID - int mSessionId; // audio session ID - effect_descriptor_t mDescriptor;// effect descriptor received from effect engine - effect_config_t mConfig; // input and output audio configuration - effect_interface_t mEffectInterface; // Effect module C API - status_t mStatus; // initialization status - uint32_t mState; // current activation state (effect_state) - Vector< wp<EffectHandle> > mHandles; // list of client handles - uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after - // sending disable command. - uint32_t mDisableWaitCnt; // current process() calls count during disable period. - }; - - // The EffectHandle class implements the IEffect interface. It provides resources - // to receive parameter updates, keeps track of effect control - // ownership and state and has a pointer to the EffectModule object it is controlling. - // There is one EffectHandle object for each application controlling (or using) - // an effect module. - // The EffectHandle is obtained by calling AudioFlinger::createEffect(). - class EffectHandle: public android::BnEffect { - public: - - EffectHandle(const sp<EffectModule>& effect, - const sp<AudioFlinger::Client>& client, - const sp<IEffectClient>& effectClient, - int32_t priority); - virtual ~EffectHandle(); - - // IEffect - virtual status_t enable(); - virtual status_t disable(); - virtual status_t command(int cmdCode, int cmdSize, void *pCmdData, int *replySize, void *pReplyData); - virtual void disconnect(); - virtual sp<IMemory> getCblk() const; - virtual status_t onTransact(uint32_t code, const Parcel& data, - Parcel* reply, uint32_t flags); - - - // Give or take control of effect module - void setControl(bool hasControl, bool signal); - void commandExecuted(int cmdCode, int cmdSize, void *pCmdData, int replySize, void *pReplyData); - void setEnabled(bool enabled); - - // Getters - int id() { return mEffect->id(); } - int priority() { return mPriority; } - bool hasControl() { return mHasControl; } - sp<EffectModule> effect() { return mEffect; } - - void dump(char* buffer, size_t size); - - protected: - - EffectHandle(const EffectHandle&); - EffectHandle& operator =(const EffectHandle&); - - sp<EffectModule> mEffect; // pointer to controlled EffectModule - sp<IEffectClient> mEffectClient; // callback interface for client notifications - sp<Client> mClient; // client for shared memory allocation - sp<IMemory> mCblkMemory; // shared memory for control block - effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory - uint8_t* mBuffer; // pointer to parameter area in shared memory - int mPriority; // client application priority to control the effect - bool mHasControl; // true if this handle is controlling the effect - }; - - // the EffectChain class represents a group of effects associated to one audio session. - // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). - // The EffecChain with session ID 0 contains global effects applied to the output mix. - // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) - // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding - // in the effect process order. When attached to a track (session ID != 0), it also provide it's own - // input buffer used by the track as accumulation buffer. - class EffectChain: public RefBase { - public: - EffectChain(const wp<ThreadBase>& wThread, int sessionId); - ~EffectChain(); - - void process_l(); - - void lock() { - mLock.lock(); - } - void unlock() { - mLock.unlock(); - } - - status_t addEffect(sp<EffectModule>& handle); - size_t removeEffect(const sp<EffectModule>& handle); - - int sessionId() { - return mSessionId; - } - sp<EffectModule> getEffectFromDesc(effect_descriptor_t *descriptor); - sp<EffectModule> getEffectFromId(int id); - sp<EffectModule> getVolumeController(); - bool setVolume(uint32_t *left, uint32_t *right); - void setDevice(uint32_t device); - void setMode(uint32_t mode); - - - void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { - mInBuffer = buffer; - mOwnInBuffer = ownsBuffer; - } - int16_t *inBuffer() { - return mInBuffer; - } - void setOutBuffer(int16_t *buffer) { - mOutBuffer = buffer; - } - int16_t *outBuffer() { - return mOutBuffer; - } - - void startTrack() {mActiveTrackCnt++;} - void stopTrack() {mActiveTrackCnt--;} - int activeTracks() { return mActiveTrackCnt;} - - status_t dump(int fd, const Vector<String16>& args); - - protected: - - EffectChain(const EffectChain&); - EffectChain& operator =(const EffectChain&); - - wp<ThreadBase> mThread; // parent mixer thread - Mutex mLock; // mutex protecting effect list - Vector<sp<EffectModule> > mEffects; // list of effect modules - int mSessionId; // audio session ID - int16_t *mInBuffer; // chain input buffer - int16_t *mOutBuffer; // chain output buffer - int mVolumeCtrlIdx; // index of insert effect having control over volume - int mActiveTrackCnt; // number of active tracks connected - bool mOwnInBuffer; // true if the chain owns its input buffer - }; - - friend class RecordThread; - friend class PlaybackThread; - - - mutable Mutex mLock; - - DefaultKeyedVector< pid_t, wp<Client> > mClients; - - mutable Mutex mHardwareLock; - AudioHardwareInterface* mAudioHardware; - mutable int mHardwareStatus; - - - DefaultKeyedVector< int, sp<PlaybackThread> > mPlaybackThreads; - PlaybackThread::stream_type_t mStreamTypes[AudioSystem::NUM_STREAM_TYPES]; - float mMasterVolume; - bool mMasterMute; - - DefaultKeyedVector< int, sp<RecordThread> > mRecordThreads; - - DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; - volatile int32_t mNextUniqueId; -#ifdef LVMX - int mLifeVibesClientPid; -#endif - uint32_t mMode; - - // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units - static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000; - // Maximum memory allocated to audio effects in KB - static const uint32_t MAX_EFFECTS_MEMORY = 512; - uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects - uint32_t mTotalEffectsMemory; // current memory used by effects -}; - -// ---------------------------------------------------------------------------- - -}; // namespace android - -#endif // ANDROID_AUDIO_FLINGER_H diff --git a/libs/audioflinger/AudioHardwareGeneric.cpp b/libs/audioflinger/AudioHardwareGeneric.cpp deleted file mode 100644 index d63c031..0000000 --- a/libs/audioflinger/AudioHardwareGeneric.cpp +++ /dev/null @@ -1,411 +0,0 @@ -/* -** -** Copyright 2007, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#include <stdint.h> -#include <sys/types.h> - -#include <stdlib.h> -#include <stdio.h> -#include <unistd.h> -#include <sched.h> -#include <fcntl.h> -#include <sys/ioctl.h> - -#define LOG_TAG "AudioHardware" -#include <utils/Log.h> -#include <utils/String8.h> - -#include "AudioHardwareGeneric.h" -#include <media/AudioRecord.h> - -namespace android { - -// ---------------------------------------------------------------------------- - -static char const * const kAudioDeviceName = "/dev/eac"; - -// ---------------------------------------------------------------------------- - -AudioHardwareGeneric::AudioHardwareGeneric() - : mOutput(0), mInput(0), mFd(-1), mMicMute(false) -{ - mFd = ::open(kAudioDeviceName, O_RDWR); -} - -AudioHardwareGeneric::~AudioHardwareGeneric() -{ - if (mFd >= 0) ::close(mFd); - closeOutputStream((AudioStreamOut *)mOutput); - closeInputStream((AudioStreamIn *)mInput); -} - -status_t AudioHardwareGeneric::initCheck() -{ - if (mFd >= 0) { - if (::access(kAudioDeviceName, O_RDWR) == NO_ERROR) - return NO_ERROR; - } - return NO_INIT; -} - -AudioStreamOut* AudioHardwareGeneric::openOutputStream( - uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) -{ - AutoMutex lock(mLock); - - // only one output stream allowed - if (mOutput) { - if (status) { - *status = INVALID_OPERATION; - } - return 0; - } - - // create new output stream - AudioStreamOutGeneric* out = new AudioStreamOutGeneric(); - status_t lStatus = out->set(this, mFd, devices, format, channels, sampleRate); - if (status) { - *status = lStatus; - } - if (lStatus == NO_ERROR) { - mOutput = out; - } else { - delete out; - } - return mOutput; -} - -void AudioHardwareGeneric::closeOutputStream(AudioStreamOut* out) { - if (mOutput && out == mOutput) { - delete mOutput; - mOutput = 0; - } -} - -AudioStreamIn* AudioHardwareGeneric::openInputStream( - uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, - status_t *status, AudioSystem::audio_in_acoustics acoustics) -{ - // check for valid input source - if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) { - return 0; - } - - AutoMutex lock(mLock); - - // only one input stream allowed - if (mInput) { - if (status) { - *status = INVALID_OPERATION; - } - return 0; - } - - // create new output stream - AudioStreamInGeneric* in = new AudioStreamInGeneric(); - status_t lStatus = in->set(this, mFd, devices, format, channels, sampleRate, acoustics); - if (status) { - *status = lStatus; - } - if (lStatus == NO_ERROR) { - mInput = in; - } else { - delete in; - } - return mInput; -} - -void AudioHardwareGeneric::closeInputStream(AudioStreamIn* in) { - if (mInput && in == mInput) { - delete mInput; - mInput = 0; - } -} - -status_t AudioHardwareGeneric::setVoiceVolume(float v) -{ - // Implement: set voice volume - return NO_ERROR; -} - -status_t AudioHardwareGeneric::setMasterVolume(float v) -{ - // Implement: set master volume - // return error - software mixer will handle it - return INVALID_OPERATION; -} - -status_t AudioHardwareGeneric::setMicMute(bool state) -{ - mMicMute = state; - return NO_ERROR; -} - -status_t AudioHardwareGeneric::getMicMute(bool* state) -{ - *state = mMicMute; - return NO_ERROR; -} - -status_t AudioHardwareGeneric::dumpInternals(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - result.append("AudioHardwareGeneric::dumpInternals\n"); - snprintf(buffer, SIZE, "\tmFd: %d mMicMute: %s\n", mFd, mMicMute? "true": "false"); - result.append(buffer); - ::write(fd, result.string(), result.size()); - return NO_ERROR; -} - -status_t AudioHardwareGeneric::dump(int fd, const Vector<String16>& args) -{ - dumpInternals(fd, args); - if (mInput) { - mInput->dump(fd, args); - } - if (mOutput) { - mOutput->dump(fd, args); - } - return NO_ERROR; -} - -// ---------------------------------------------------------------------------- - -status_t AudioStreamOutGeneric::set( - AudioHardwareGeneric *hw, - int fd, - uint32_t devices, - int *pFormat, - uint32_t *pChannels, - uint32_t *pRate) -{ - int lFormat = pFormat ? *pFormat : 0; - uint32_t lChannels = pChannels ? *pChannels : 0; - uint32_t lRate = pRate ? *pRate : 0; - - // fix up defaults - if (lFormat == 0) lFormat = format(); - if (lChannels == 0) lChannels = channels(); - if (lRate == 0) lRate = sampleRate(); - - // check values - if ((lFormat != format()) || - (lChannels != channels()) || - (lRate != sampleRate())) { - if (pFormat) *pFormat = format(); - if (pChannels) *pChannels = channels(); - if (pRate) *pRate = sampleRate(); - return BAD_VALUE; - } - - if (pFormat) *pFormat = lFormat; - if (pChannels) *pChannels = lChannels; - if (pRate) *pRate = lRate; - - mAudioHardware = hw; - mFd = fd; - mDevice = devices; - return NO_ERROR; -} - -AudioStreamOutGeneric::~AudioStreamOutGeneric() -{ -} - -ssize_t AudioStreamOutGeneric::write(const void* buffer, size_t bytes) -{ - Mutex::Autolock _l(mLock); - return ssize_t(::write(mFd, buffer, bytes)); -} - -status_t AudioStreamOutGeneric::standby() -{ - // Implement: audio hardware to standby mode - return NO_ERROR; -} - -status_t AudioStreamOutGeneric::dump(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - snprintf(buffer, SIZE, "AudioStreamOutGeneric::dump\n"); - result.append(buffer); - snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate()); - result.append(buffer); - snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize()); - result.append(buffer); - snprintf(buffer, SIZE, "\tchannels: %d\n", channels()); - result.append(buffer); - snprintf(buffer, SIZE, "\tformat: %d\n", format()); - result.append(buffer); - snprintf(buffer, SIZE, "\tdevice: %d\n", mDevice); - result.append(buffer); - snprintf(buffer, SIZE, "\tmAudioHardware: %p\n", mAudioHardware); - result.append(buffer); - snprintf(buffer, SIZE, "\tmFd: %d\n", mFd); - result.append(buffer); - ::write(fd, result.string(), result.size()); - return NO_ERROR; -} - -status_t AudioStreamOutGeneric::setParameters(const String8& keyValuePairs) -{ - AudioParameter param = AudioParameter(keyValuePairs); - String8 key = String8(AudioParameter::keyRouting); - status_t status = NO_ERROR; - int device; - LOGV("setParameters() %s", keyValuePairs.string()); - - if (param.getInt(key, device) == NO_ERROR) { - mDevice = device; - param.remove(key); - } - - if (param.size()) { - status = BAD_VALUE; - } - return status; -} - -String8 AudioStreamOutGeneric::getParameters(const String8& keys) -{ - AudioParameter param = AudioParameter(keys); - String8 value; - String8 key = String8(AudioParameter::keyRouting); - - if (param.get(key, value) == NO_ERROR) { - param.addInt(key, (int)mDevice); - } - - LOGV("getParameters() %s", param.toString().string()); - return param.toString(); -} - -status_t AudioStreamOutGeneric::getRenderPosition(uint32_t *dspFrames) -{ - return INVALID_OPERATION; -} - -// ---------------------------------------------------------------------------- - -// record functions -status_t AudioStreamInGeneric::set( - AudioHardwareGeneric *hw, - int fd, - uint32_t devices, - int *pFormat, - uint32_t *pChannels, - uint32_t *pRate, - AudioSystem::audio_in_acoustics acoustics) -{ - if (pFormat == 0 || pChannels == 0 || pRate == 0) return BAD_VALUE; - LOGV("AudioStreamInGeneric::set(%p, %d, %d, %d, %u)", hw, fd, *pFormat, *pChannels, *pRate); - // check values - if ((*pFormat != format()) || - (*pChannels != channels()) || - (*pRate != sampleRate())) { - LOGE("Error opening input channel"); - *pFormat = format(); - *pChannels = channels(); - *pRate = sampleRate(); - return BAD_VALUE; - } - - mAudioHardware = hw; - mFd = fd; - mDevice = devices; - return NO_ERROR; -} - -AudioStreamInGeneric::~AudioStreamInGeneric() -{ -} - -ssize_t AudioStreamInGeneric::read(void* buffer, ssize_t bytes) -{ - AutoMutex lock(mLock); - if (mFd < 0) { - LOGE("Attempt to read from unopened device"); - return NO_INIT; - } - return ::read(mFd, buffer, bytes); -} - -status_t AudioStreamInGeneric::dump(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - snprintf(buffer, SIZE, "AudioStreamInGeneric::dump\n"); - result.append(buffer); - snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate()); - result.append(buffer); - snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize()); - result.append(buffer); - snprintf(buffer, SIZE, "\tchannels: %d\n", channels()); - result.append(buffer); - snprintf(buffer, SIZE, "\tformat: %d\n", format()); - result.append(buffer); - snprintf(buffer, SIZE, "\tdevice: %d\n", mDevice); - result.append(buffer); - snprintf(buffer, SIZE, "\tmAudioHardware: %p\n", mAudioHardware); - result.append(buffer); - snprintf(buffer, SIZE, "\tmFd: %d\n", mFd); - result.append(buffer); - ::write(fd, result.string(), result.size()); - return NO_ERROR; -} - -status_t AudioStreamInGeneric::setParameters(const String8& keyValuePairs) -{ - AudioParameter param = AudioParameter(keyValuePairs); - String8 key = String8(AudioParameter::keyRouting); - status_t status = NO_ERROR; - int device; - LOGV("setParameters() %s", keyValuePairs.string()); - - if (param.getInt(key, device) == NO_ERROR) { - mDevice = device; - param.remove(key); - } - - if (param.size()) { - status = BAD_VALUE; - } - return status; -} - -String8 AudioStreamInGeneric::getParameters(const String8& keys) -{ - AudioParameter param = AudioParameter(keys); - String8 value; - String8 key = String8(AudioParameter::keyRouting); - - if (param.get(key, value) == NO_ERROR) { - param.addInt(key, (int)mDevice); - } - - LOGV("getParameters() %s", param.toString().string()); - return param.toString(); -} - -// ---------------------------------------------------------------------------- - -}; // namespace android diff --git a/libs/audioflinger/AudioHardwareGeneric.h b/libs/audioflinger/AudioHardwareGeneric.h deleted file mode 100644 index aa4e78d..0000000 --- a/libs/audioflinger/AudioHardwareGeneric.h +++ /dev/null @@ -1,151 +0,0 @@ -/* -** -** Copyright 2007, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#ifndef ANDROID_AUDIO_HARDWARE_GENERIC_H -#define ANDROID_AUDIO_HARDWARE_GENERIC_H - -#include <stdint.h> -#include <sys/types.h> - -#include <utils/threads.h> - -#include <hardware_legacy/AudioHardwareBase.h> - -namespace android { - -// ---------------------------------------------------------------------------- - -class AudioHardwareGeneric; - -class AudioStreamOutGeneric : public AudioStreamOut { -public: - AudioStreamOutGeneric() : mAudioHardware(0), mFd(-1) {} - virtual ~AudioStreamOutGeneric(); - - virtual status_t set( - AudioHardwareGeneric *hw, - int mFd, - uint32_t devices, - int *pFormat, - uint32_t *pChannels, - uint32_t *pRate); - - virtual uint32_t sampleRate() const { return 44100; } - virtual size_t bufferSize() const { return 4096; } - virtual uint32_t channels() const { return AudioSystem::CHANNEL_OUT_STEREO; } - virtual int format() const { return AudioSystem::PCM_16_BIT; } - virtual uint32_t latency() const { return 20; } - virtual status_t setVolume(float left, float right) { return INVALID_OPERATION; } - virtual ssize_t write(const void* buffer, size_t bytes); - virtual status_t standby(); - virtual status_t dump(int fd, const Vector<String16>& args); - virtual status_t setParameters(const String8& keyValuePairs); - virtual String8 getParameters(const String8& keys); - virtual status_t getRenderPosition(uint32_t *dspFrames); - -private: - AudioHardwareGeneric *mAudioHardware; - Mutex mLock; - int mFd; - uint32_t mDevice; -}; - -class AudioStreamInGeneric : public AudioStreamIn { -public: - AudioStreamInGeneric() : mAudioHardware(0), mFd(-1) {} - virtual ~AudioStreamInGeneric(); - - virtual status_t set( - AudioHardwareGeneric *hw, - int mFd, - uint32_t devices, - int *pFormat, - uint32_t *pChannels, - uint32_t *pRate, - AudioSystem::audio_in_acoustics acoustics); - - virtual uint32_t sampleRate() const { return 8000; } - virtual size_t bufferSize() const { return 320; } - virtual uint32_t channels() const { return AudioSystem::CHANNEL_IN_MONO; } - virtual int format() const { return AudioSystem::PCM_16_BIT; } - virtual status_t setGain(float gain) { return INVALID_OPERATION; } - virtual ssize_t read(void* buffer, ssize_t bytes); - virtual status_t dump(int fd, const Vector<String16>& args); - virtual status_t standby() { return NO_ERROR; } - virtual status_t setParameters(const String8& keyValuePairs); - virtual String8 getParameters(const String8& keys); - virtual unsigned int getInputFramesLost() const { return 0; } - -private: - AudioHardwareGeneric *mAudioHardware; - Mutex mLock; - int mFd; - uint32_t mDevice; -}; - - -class AudioHardwareGeneric : public AudioHardwareBase -{ -public: - AudioHardwareGeneric(); - virtual ~AudioHardwareGeneric(); - virtual status_t initCheck(); - virtual status_t setVoiceVolume(float volume); - virtual status_t setMasterVolume(float volume); - - // mic mute - virtual status_t setMicMute(bool state); - virtual status_t getMicMute(bool* state); - - // create I/O streams - virtual AudioStreamOut* openOutputStream( - uint32_t devices, - int *format=0, - uint32_t *channels=0, - uint32_t *sampleRate=0, - status_t *status=0); - virtual void closeOutputStream(AudioStreamOut* out); - - virtual AudioStreamIn* openInputStream( - uint32_t devices, - int *format, - uint32_t *channels, - uint32_t *sampleRate, - status_t *status, - AudioSystem::audio_in_acoustics acoustics); - virtual void closeInputStream(AudioStreamIn* in); - - void closeOutputStream(AudioStreamOutGeneric* out); - void closeInputStream(AudioStreamInGeneric* in); -protected: - virtual status_t dump(int fd, const Vector<String16>& args); - -private: - status_t dumpInternals(int fd, const Vector<String16>& args); - - Mutex mLock; - AudioStreamOutGeneric *mOutput; - AudioStreamInGeneric *mInput; - int mFd; - bool mMicMute; -}; - -// ---------------------------------------------------------------------------- - -}; // namespace android - -#endif // ANDROID_AUDIO_HARDWARE_GENERIC_H diff --git a/libs/audioflinger/AudioHardwareInterface.cpp b/libs/audioflinger/AudioHardwareInterface.cpp deleted file mode 100644 index 9a4a7f9..0000000 --- a/libs/audioflinger/AudioHardwareInterface.cpp +++ /dev/null @@ -1,182 +0,0 @@ -/* -** -** Copyright 2007, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#include <cutils/properties.h> -#include <string.h> -#include <unistd.h> -//#define LOG_NDEBUG 0 - -#define LOG_TAG "AudioHardwareInterface" -#include <utils/Log.h> -#include <utils/String8.h> - -#include "AudioHardwareStub.h" -#include "AudioHardwareGeneric.h" -#ifdef WITH_A2DP -#include "A2dpAudioInterface.h" -#endif - -#ifdef ENABLE_AUDIO_DUMP -#include "AudioDumpInterface.h" -#endif - - -// change to 1 to log routing calls -#define LOG_ROUTING_CALLS 1 - -namespace android { - -#if LOG_ROUTING_CALLS -static const char* routingModeStrings[] = -{ - "OUT OF RANGE", - "INVALID", - "CURRENT", - "NORMAL", - "RINGTONE", - "IN_CALL" -}; - -static const char* routeNone = "NONE"; - -static const char* displayMode(int mode) -{ - if ((mode < -2) || (mode > 2)) - return routingModeStrings[0]; - return routingModeStrings[mode+3]; -} -#endif - -// ---------------------------------------------------------------------------- - -AudioHardwareInterface* AudioHardwareInterface::create() -{ - /* - * FIXME: This code needs to instantiate the correct audio device - * interface. For now - we use compile-time switches. - */ - AudioHardwareInterface* hw = 0; - char value[PROPERTY_VALUE_MAX]; - -#ifdef GENERIC_AUDIO - hw = new AudioHardwareGeneric(); -#else - // if running in emulation - use the emulator driver - if (property_get("ro.kernel.qemu", value, 0)) { - LOGD("Running in emulation - using generic audio driver"); - hw = new AudioHardwareGeneric(); - } - else { - LOGV("Creating Vendor Specific AudioHardware"); - hw = createAudioHardware(); - } -#endif - if (hw->initCheck() != NO_ERROR) { - LOGW("Using stubbed audio hardware. No sound will be produced."); - delete hw; - hw = new AudioHardwareStub(); - } - -#ifdef WITH_A2DP - hw = new A2dpAudioInterface(hw); -#endif - -#ifdef ENABLE_AUDIO_DUMP - // This code adds a record of buffers in a file to write calls made by AudioFlinger. - // It replaces the current AudioHardwareInterface object by an intermediate one which - // will record buffers in a file (after sending them to hardware) for testing purpose. - // This feature is enabled by defining symbol ENABLE_AUDIO_DUMP. - // The output file is set with setParameters("test_cmd_file_name=<name>"). Pause are not recorded in the file. - LOGV("opening PCM dump interface"); - hw = new AudioDumpInterface(hw); // replace interface -#endif - return hw; -} - -AudioStreamOut::~AudioStreamOut() -{ -} - -AudioStreamIn::~AudioStreamIn() {} - -AudioHardwareBase::AudioHardwareBase() -{ - mMode = 0; -} - -status_t AudioHardwareBase::setMode(int mode) -{ -#if LOG_ROUTING_CALLS - LOGD("setMode(%s)", displayMode(mode)); -#endif - if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) - return BAD_VALUE; - if (mMode == mode) - return ALREADY_EXISTS; - mMode = mode; - return NO_ERROR; -} - -// default implementation -status_t AudioHardwareBase::setParameters(const String8& keyValuePairs) -{ - return NO_ERROR; -} - -// default implementation -String8 AudioHardwareBase::getParameters(const String8& keys) -{ - AudioParameter param = AudioParameter(keys); - return param.toString(); -} - -// default implementation -size_t AudioHardwareBase::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) -{ - if (sampleRate != 8000) { - LOGW("getInputBufferSize bad sampling rate: %d", sampleRate); - return 0; - } - if (format != AudioSystem::PCM_16_BIT) { - LOGW("getInputBufferSize bad format: %d", format); - return 0; - } - if (channelCount != 1) { - LOGW("getInputBufferSize bad channel count: %d", channelCount); - return 0; - } - - return 320; -} - -status_t AudioHardwareBase::dumpState(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - snprintf(buffer, SIZE, "AudioHardwareBase::dumpState\n"); - result.append(buffer); - snprintf(buffer, SIZE, "\tmMode: %d\n", mMode); - result.append(buffer); - ::write(fd, result.string(), result.size()); - dump(fd, args); // Dump the state of the concrete child. - return NO_ERROR; -} - -// ---------------------------------------------------------------------------- - -}; // namespace android diff --git a/libs/audioflinger/AudioHardwareStub.cpp b/libs/audioflinger/AudioHardwareStub.cpp deleted file mode 100644 index d481150..0000000 --- a/libs/audioflinger/AudioHardwareStub.cpp +++ /dev/null @@ -1,209 +0,0 @@ -/* //device/servers/AudioFlinger/AudioHardwareStub.cpp -** -** Copyright 2007, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#include <stdint.h> -#include <sys/types.h> - -#include <stdlib.h> -#include <unistd.h> -#include <utils/String8.h> - -#include "AudioHardwareStub.h" -#include <media/AudioRecord.h> - -namespace android { - -// ---------------------------------------------------------------------------- - -AudioHardwareStub::AudioHardwareStub() : mMicMute(false) -{ -} - -AudioHardwareStub::~AudioHardwareStub() -{ -} - -status_t AudioHardwareStub::initCheck() -{ - return NO_ERROR; -} - -AudioStreamOut* AudioHardwareStub::openOutputStream( - uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) -{ - AudioStreamOutStub* out = new AudioStreamOutStub(); - status_t lStatus = out->set(format, channels, sampleRate); - if (status) { - *status = lStatus; - } - if (lStatus == NO_ERROR) - return out; - delete out; - return 0; -} - -void AudioHardwareStub::closeOutputStream(AudioStreamOut* out) -{ - delete out; -} - -AudioStreamIn* AudioHardwareStub::openInputStream( - uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, - status_t *status, AudioSystem::audio_in_acoustics acoustics) -{ - // check for valid input source - if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) { - return 0; - } - - AudioStreamInStub* in = new AudioStreamInStub(); - status_t lStatus = in->set(format, channels, sampleRate, acoustics); - if (status) { - *status = lStatus; - } - if (lStatus == NO_ERROR) - return in; - delete in; - return 0; -} - -void AudioHardwareStub::closeInputStream(AudioStreamIn* in) -{ - delete in; -} - -status_t AudioHardwareStub::setVoiceVolume(float volume) -{ - return NO_ERROR; -} - -status_t AudioHardwareStub::setMasterVolume(float volume) -{ - return NO_ERROR; -} - -status_t AudioHardwareStub::dumpInternals(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - result.append("AudioHardwareStub::dumpInternals\n"); - snprintf(buffer, SIZE, "\tmMicMute: %s\n", mMicMute? "true": "false"); - result.append(buffer); - ::write(fd, result.string(), result.size()); - return NO_ERROR; -} - -status_t AudioHardwareStub::dump(int fd, const Vector<String16>& args) -{ - dumpInternals(fd, args); - return NO_ERROR; -} - -// ---------------------------------------------------------------------------- - -status_t AudioStreamOutStub::set(int *pFormat, uint32_t *pChannels, uint32_t *pRate) -{ - if (pFormat) *pFormat = format(); - if (pChannels) *pChannels = channels(); - if (pRate) *pRate = sampleRate(); - - return NO_ERROR; -} - -ssize_t AudioStreamOutStub::write(const void* buffer, size_t bytes) -{ - // fake timing for audio output - usleep(bytes * 1000000 / sizeof(int16_t) / AudioSystem::popCount(channels()) / sampleRate()); - return bytes; -} - -status_t AudioStreamOutStub::standby() -{ - return NO_ERROR; -} - -status_t AudioStreamOutStub::dump(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - snprintf(buffer, SIZE, "AudioStreamOutStub::dump\n"); - snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate()); - snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize()); - snprintf(buffer, SIZE, "\tchannels: %d\n", channels()); - snprintf(buffer, SIZE, "\tformat: %d\n", format()); - result.append(buffer); - ::write(fd, result.string(), result.size()); - return NO_ERROR; -} - -String8 AudioStreamOutStub::getParameters(const String8& keys) -{ - AudioParameter param = AudioParameter(keys); - return param.toString(); -} - -status_t AudioStreamOutStub::getRenderPosition(uint32_t *dspFrames) -{ - return INVALID_OPERATION; -} - -// ---------------------------------------------------------------------------- - -status_t AudioStreamInStub::set(int *pFormat, uint32_t *pChannels, uint32_t *pRate, - AudioSystem::audio_in_acoustics acoustics) -{ - return NO_ERROR; -} - -ssize_t AudioStreamInStub::read(void* buffer, ssize_t bytes) -{ - // fake timing for audio input - usleep(bytes * 1000000 / sizeof(int16_t) / AudioSystem::popCount(channels()) / sampleRate()); - memset(buffer, 0, bytes); - return bytes; -} - -status_t AudioStreamInStub::dump(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - snprintf(buffer, SIZE, "AudioStreamInStub::dump\n"); - result.append(buffer); - snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate()); - result.append(buffer); - snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize()); - result.append(buffer); - snprintf(buffer, SIZE, "\tchannels: %d\n", channels()); - result.append(buffer); - snprintf(buffer, SIZE, "\tformat: %d\n", format()); - result.append(buffer); - ::write(fd, result.string(), result.size()); - return NO_ERROR; -} - -String8 AudioStreamInStub::getParameters(const String8& keys) -{ - AudioParameter param = AudioParameter(keys); - return param.toString(); -} - -// ---------------------------------------------------------------------------- - -}; // namespace android diff --git a/libs/audioflinger/AudioHardwareStub.h b/libs/audioflinger/AudioHardwareStub.h deleted file mode 100644 index 06a29de..0000000 --- a/libs/audioflinger/AudioHardwareStub.h +++ /dev/null @@ -1,106 +0,0 @@ -/* //device/servers/AudioFlinger/AudioHardwareStub.h -** -** Copyright 2007, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#ifndef ANDROID_AUDIO_HARDWARE_STUB_H -#define ANDROID_AUDIO_HARDWARE_STUB_H - -#include <stdint.h> -#include <sys/types.h> - -#include <hardware_legacy/AudioHardwareBase.h> - -namespace android { - -// ---------------------------------------------------------------------------- - -class AudioStreamOutStub : public AudioStreamOut { -public: - virtual status_t set(int *pFormat, uint32_t *pChannels, uint32_t *pRate); - virtual uint32_t sampleRate() const { return 44100; } - virtual size_t bufferSize() const { return 4096; } - virtual uint32_t channels() const { return AudioSystem::CHANNEL_OUT_STEREO; } - virtual int format() const { return AudioSystem::PCM_16_BIT; } - virtual uint32_t latency() const { return 0; } - virtual status_t setVolume(float left, float right) { return NO_ERROR; } - virtual ssize_t write(const void* buffer, size_t bytes); - virtual status_t standby(); - virtual status_t dump(int fd, const Vector<String16>& args); - virtual status_t setParameters(const String8& keyValuePairs) { return NO_ERROR;} - virtual String8 getParameters(const String8& keys); - virtual status_t getRenderPosition(uint32_t *dspFrames); -}; - -class AudioStreamInStub : public AudioStreamIn { -public: - virtual status_t set(int *pFormat, uint32_t *pChannels, uint32_t *pRate, AudioSystem::audio_in_acoustics acoustics); - virtual uint32_t sampleRate() const { return 8000; } - virtual size_t bufferSize() const { return 320; } - virtual uint32_t channels() const { return AudioSystem::CHANNEL_IN_MONO; } - virtual int format() const { return AudioSystem::PCM_16_BIT; } - virtual status_t setGain(float gain) { return NO_ERROR; } - virtual ssize_t read(void* buffer, ssize_t bytes); - virtual status_t dump(int fd, const Vector<String16>& args); - virtual status_t standby() { return NO_ERROR; } - virtual status_t setParameters(const String8& keyValuePairs) { return NO_ERROR;} - virtual String8 getParameters(const String8& keys); - virtual unsigned int getInputFramesLost() const { return 0; } -}; - -class AudioHardwareStub : public AudioHardwareBase -{ -public: - AudioHardwareStub(); - virtual ~AudioHardwareStub(); - virtual status_t initCheck(); - virtual status_t setVoiceVolume(float volume); - virtual status_t setMasterVolume(float volume); - - // mic mute - virtual status_t setMicMute(bool state) { mMicMute = state; return NO_ERROR; } - virtual status_t getMicMute(bool* state) { *state = mMicMute ; return NO_ERROR; } - - // create I/O streams - virtual AudioStreamOut* openOutputStream( - uint32_t devices, - int *format=0, - uint32_t *channels=0, - uint32_t *sampleRate=0, - status_t *status=0); - virtual void closeOutputStream(AudioStreamOut* out); - - virtual AudioStreamIn* openInputStream( - uint32_t devices, - int *format, - uint32_t *channels, - uint32_t *sampleRate, - status_t *status, - AudioSystem::audio_in_acoustics acoustics); - virtual void closeInputStream(AudioStreamIn* in); - -protected: - virtual status_t dump(int fd, const Vector<String16>& args); - - bool mMicMute; -private: - status_t dumpInternals(int fd, const Vector<String16>& args); -}; - -// ---------------------------------------------------------------------------- - -}; // namespace android - -#endif // ANDROID_AUDIO_HARDWARE_STUB_H diff --git a/libs/audioflinger/AudioMixer.cpp b/libs/audioflinger/AudioMixer.cpp deleted file mode 100644 index 8aaa325..0000000 --- a/libs/audioflinger/AudioMixer.cpp +++ /dev/null @@ -1,1195 +0,0 @@ -/* //device/include/server/AudioFlinger/AudioMixer.cpp -** -** Copyright 2007, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#define LOG_TAG "AudioMixer" -//#define LOG_NDEBUG 0 - -#include <stdint.h> -#include <string.h> -#include <stdlib.h> -#include <sys/types.h> - -#include <utils/Errors.h> -#include <utils/Log.h> - -#include "AudioMixer.h" - -namespace android { -// ---------------------------------------------------------------------------- - -static inline int16_t clamp16(int32_t sample) -{ - if ((sample>>15) ^ (sample>>31)) - sample = 0x7FFF ^ (sample>>31); - return sample; -} - -// ---------------------------------------------------------------------------- - -AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate) - : mActiveTrack(0), mTrackNames(0), mSampleRate(sampleRate) -{ - mState.enabledTracks= 0; - mState.needsChanged = 0; - mState.frameCount = frameCount; - mState.outputTemp = 0; - mState.resampleTemp = 0; - mState.hook = process__nop; - track_t* t = mState.tracks; - for (int i=0 ; i<32 ; i++) { - t->needs = 0; - t->volume[0] = UNITY_GAIN; - t->volume[1] = UNITY_GAIN; - t->volumeInc[0] = 0; - t->volumeInc[1] = 0; - t->auxLevel = 0; - t->auxInc = 0; - t->channelCount = 2; - t->enabled = 0; - t->format = 16; - t->buffer.raw = 0; - t->bufferProvider = 0; - t->hook = 0; - t->resampler = 0; - t->sampleRate = mSampleRate; - t->in = 0; - t->mainBuffer = NULL; - t->auxBuffer = NULL; - t++; - } -} - - AudioMixer::~AudioMixer() - { - track_t* t = mState.tracks; - for (int i=0 ; i<32 ; i++) { - delete t->resampler; - t++; - } - delete [] mState.outputTemp; - delete [] mState.resampleTemp; - } - - int AudioMixer::getTrackName() - { - uint32_t names = mTrackNames; - uint32_t mask = 1; - int n = 0; - while (names & mask) { - mask <<= 1; - n++; - } - if (mask) { - LOGV("add track (%d)", n); - mTrackNames |= mask; - return TRACK0 + n; - } - return -1; - } - - void AudioMixer::invalidateState(uint32_t mask) - { - if (mask) { - mState.needsChanged |= mask; - mState.hook = process__validate; - } - } - - void AudioMixer::deleteTrackName(int name) - { - name -= TRACK0; - if (uint32_t(name) < MAX_NUM_TRACKS) { - LOGV("deleteTrackName(%d)", name); - track_t& track(mState.tracks[ name ]); - if (track.enabled != 0) { - track.enabled = 0; - invalidateState(1<<name); - } - if (track.resampler) { - // delete the resampler - delete track.resampler; - track.resampler = 0; - track.sampleRate = mSampleRate; - invalidateState(1<<name); - } - track.volumeInc[0] = 0; - track.volumeInc[1] = 0; - mTrackNames &= ~(1<<name); - } - } - -status_t AudioMixer::enable(int name) -{ - switch (name) { - case MIXING: { - if (mState.tracks[ mActiveTrack ].enabled != 1) { - mState.tracks[ mActiveTrack ].enabled = 1; - LOGV("enable(%d)", mActiveTrack); - invalidateState(1<<mActiveTrack); - } - } break; - default: - return NAME_NOT_FOUND; - } - return NO_ERROR; -} - -status_t AudioMixer::disable(int name) -{ - switch (name) { - case MIXING: { - if (mState.tracks[ mActiveTrack ].enabled != 0) { - mState.tracks[ mActiveTrack ].enabled = 0; - LOGV("disable(%d)", mActiveTrack); - invalidateState(1<<mActiveTrack); - } - } break; - default: - return NAME_NOT_FOUND; - } - return NO_ERROR; -} - -status_t AudioMixer::setActiveTrack(int track) -{ - if (uint32_t(track-TRACK0) >= MAX_NUM_TRACKS) { - return BAD_VALUE; - } - mActiveTrack = track - TRACK0; - return NO_ERROR; -} - -status_t AudioMixer::setParameter(int target, int name, void *value) -{ - int valueInt = (int)value; - int32_t *valueBuf = (int32_t *)value; - - switch (target) { - case TRACK: - if (name == CHANNEL_COUNT) { - if ((uint32_t(valueInt) <= MAX_NUM_CHANNELS) && (valueInt)) { - if (mState.tracks[ mActiveTrack ].channelCount != valueInt) { - mState.tracks[ mActiveTrack ].channelCount = valueInt; - LOGV("setParameter(TRACK, CHANNEL_COUNT, %d)", valueInt); - invalidateState(1<<mActiveTrack); - } - return NO_ERROR; - } - } - if (name == MAIN_BUFFER) { - if (mState.tracks[ mActiveTrack ].mainBuffer != valueBuf) { - mState.tracks[ mActiveTrack ].mainBuffer = valueBuf; - LOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf); - invalidateState(1<<mActiveTrack); - } - return NO_ERROR; - } - if (name == AUX_BUFFER) { - if (mState.tracks[ mActiveTrack ].auxBuffer != valueBuf) { - mState.tracks[ mActiveTrack ].auxBuffer = valueBuf; - LOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf); - invalidateState(1<<mActiveTrack); - } - return NO_ERROR; - } - - break; - case RESAMPLE: - if (name == SAMPLE_RATE) { - if (valueInt > 0) { - track_t& track = mState.tracks[ mActiveTrack ]; - if (track.setResampler(uint32_t(valueInt), mSampleRate)) { - LOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)", - uint32_t(valueInt)); - invalidateState(1<<mActiveTrack); - } - return NO_ERROR; - } - } - break; - case RAMP_VOLUME: - case VOLUME: - if ((uint32_t(name-VOLUME0) < MAX_NUM_CHANNELS)) { - track_t& track = mState.tracks[ mActiveTrack ]; - if (track.volume[name-VOLUME0] != valueInt) { - LOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt); - track.prevVolume[name-VOLUME0] = track.volume[name-VOLUME0] << 16; - track.volume[name-VOLUME0] = valueInt; - if (target == VOLUME) { - track.prevVolume[name-VOLUME0] = valueInt << 16; - track.volumeInc[name-VOLUME0] = 0; - } else { - int32_t d = (valueInt<<16) - track.prevVolume[name-VOLUME0]; - int32_t volInc = d / int32_t(mState.frameCount); - track.volumeInc[name-VOLUME0] = volInc; - if (volInc == 0) { - track.prevVolume[name-VOLUME0] = valueInt << 16; - } - } - invalidateState(1<<mActiveTrack); - } - return NO_ERROR; - } else if (name == AUXLEVEL) { - track_t& track = mState.tracks[ mActiveTrack ]; - if (track.auxLevel != valueInt) { - LOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt); - track.prevAuxLevel = track.auxLevel << 16; - track.auxLevel = valueInt; - if (target == VOLUME) { - track.prevAuxLevel = valueInt << 16; - track.auxInc = 0; - } else { - int32_t d = (valueInt<<16) - track.prevAuxLevel; - int32_t volInc = d / int32_t(mState.frameCount); - track.auxInc = volInc; - if (volInc == 0) { - track.prevAuxLevel = valueInt << 16; - } - } - invalidateState(1<<mActiveTrack); - } - return NO_ERROR; - } - break; - } - return BAD_VALUE; -} - -bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate) -{ - if (value!=devSampleRate || resampler) { - if (sampleRate != value) { - sampleRate = value; - if (resampler == 0) { - resampler = AudioResampler::create( - format, channelCount, devSampleRate); - } - return true; - } - } - return false; -} - -bool AudioMixer::track_t::doesResample() const -{ - return resampler != 0; -} - -inline -void AudioMixer::track_t::adjustVolumeRamp(bool aux) -{ - for (int i=0 ; i<2 ; i++) { - if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) || - ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) { - volumeInc[i] = 0; - prevVolume[i] = volume[i]<<16; - } - } - if (aux) { - if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) || - ((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) { - auxInc = 0; - prevAuxLevel = auxLevel<<16; - } - } -} - - -status_t AudioMixer::setBufferProvider(AudioBufferProvider* buffer) -{ - mState.tracks[ mActiveTrack ].bufferProvider = buffer; - return NO_ERROR; -} - - - -void AudioMixer::process() -{ - mState.hook(&mState); -} - - -void AudioMixer::process__validate(state_t* state) -{ - LOGW_IF(!state->needsChanged, - "in process__validate() but nothing's invalid"); - - uint32_t changed = state->needsChanged; - state->needsChanged = 0; // clear the validation flag - - // recompute which tracks are enabled / disabled - uint32_t enabled = 0; - uint32_t disabled = 0; - while (changed) { - const int i = 31 - __builtin_clz(changed); - const uint32_t mask = 1<<i; - changed &= ~mask; - track_t& t = state->tracks[i]; - (t.enabled ? enabled : disabled) |= mask; - } - state->enabledTracks &= ~disabled; - state->enabledTracks |= enabled; - - // compute everything we need... - int countActiveTracks = 0; - int all16BitsStereoNoResample = 1; - int resampling = 0; - int volumeRamp = 0; - uint32_t en = state->enabledTracks; - while (en) { - const int i = 31 - __builtin_clz(en); - en &= ~(1<<i); - - countActiveTracks++; - track_t& t = state->tracks[i]; - uint32_t n = 0; - n |= NEEDS_CHANNEL_1 + t.channelCount - 1; - n |= NEEDS_FORMAT_16; - n |= t.doesResample() ? NEEDS_RESAMPLE_ENABLED : NEEDS_RESAMPLE_DISABLED; - if (t.auxLevel != 0 && t.auxBuffer != NULL) { - n |= NEEDS_AUX_ENABLED; - } - - if (t.volumeInc[0]|t.volumeInc[1]) { - volumeRamp = 1; - } else if (!t.doesResample() && t.volumeRL == 0) { - n |= NEEDS_MUTE_ENABLED; - } - t.needs = n; - - if ((n & NEEDS_MUTE__MASK) == NEEDS_MUTE_ENABLED) { - t.hook = track__nop; - } else { - if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) { - all16BitsStereoNoResample = 0; - } - if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) { - all16BitsStereoNoResample = 0; - resampling = 1; - t.hook = track__genericResample; - } else { - if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){ - t.hook = track__16BitsMono; - all16BitsStereoNoResample = 0; - } - if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_2){ - t.hook = track__16BitsStereo; - } - } - } - } - - // select the processing hooks - state->hook = process__nop; - if (countActiveTracks) { - if (resampling) { - if (!state->outputTemp) { - state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; - } - if (!state->resampleTemp) { - state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; - } - state->hook = process__genericResampling; - } else { - if (state->outputTemp) { - delete [] state->outputTemp; - state->outputTemp = 0; - } - if (state->resampleTemp) { - delete [] state->resampleTemp; - state->resampleTemp = 0; - } - state->hook = process__genericNoResampling; - if (all16BitsStereoNoResample && !volumeRamp) { - if (countActiveTracks == 1) { - state->hook = process__OneTrack16BitsStereoNoResampling; - } - } - } - } - - LOGV("mixer configuration change: %d activeTracks (%08x) " - "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d", - countActiveTracks, state->enabledTracks, - all16BitsStereoNoResample, resampling, volumeRamp); - - state->hook(state); - - // Now that the volume ramp has been done, set optimal state and - // track hooks for subsequent mixer process - if (countActiveTracks) { - int allMuted = 1; - uint32_t en = state->enabledTracks; - while (en) { - const int i = 31 - __builtin_clz(en); - en &= ~(1<<i); - track_t& t = state->tracks[i]; - if (!t.doesResample() && t.volumeRL == 0) - { - t.needs |= NEEDS_MUTE_ENABLED; - t.hook = track__nop; - } else { - allMuted = 0; - } - } - if (allMuted) { - state->hook = process__nop; - } else if (all16BitsStereoNoResample) { - if (countActiveTracks == 1) { - state->hook = process__OneTrack16BitsStereoNoResampling; - } - } - } -} - -static inline -int32_t mulAdd(int16_t in, int16_t v, int32_t a) -{ -#if defined(__arm__) && !defined(__thumb__) - int32_t out; - asm( "smlabb %[out], %[in], %[v], %[a] \n" - : [out]"=r"(out) - : [in]"%r"(in), [v]"r"(v), [a]"r"(a) - : ); - return out; -#else - return a + in * int32_t(v); -#endif -} - -static inline -int32_t mul(int16_t in, int16_t v) -{ -#if defined(__arm__) && !defined(__thumb__) - int32_t out; - asm( "smulbb %[out], %[in], %[v] \n" - : [out]"=r"(out) - : [in]"%r"(in), [v]"r"(v) - : ); - return out; -#else - return in * int32_t(v); -#endif -} - -static inline -int32_t mulAddRL(int left, uint32_t inRL, uint32_t vRL, int32_t a) -{ -#if defined(__arm__) && !defined(__thumb__) - int32_t out; - if (left) { - asm( "smlabb %[out], %[inRL], %[vRL], %[a] \n" - : [out]"=r"(out) - : [inRL]"%r"(inRL), [vRL]"r"(vRL), [a]"r"(a) - : ); - } else { - asm( "smlatt %[out], %[inRL], %[vRL], %[a] \n" - : [out]"=r"(out) - : [inRL]"%r"(inRL), [vRL]"r"(vRL), [a]"r"(a) - : ); - } - return out; -#else - if (left) { - return a + int16_t(inRL&0xFFFF) * int16_t(vRL&0xFFFF); - } else { - return a + int16_t(inRL>>16) * int16_t(vRL>>16); - } -#endif -} - -static inline -int32_t mulRL(int left, uint32_t inRL, uint32_t vRL) -{ -#if defined(__arm__) && !defined(__thumb__) - int32_t out; - if (left) { - asm( "smulbb %[out], %[inRL], %[vRL] \n" - : [out]"=r"(out) - : [inRL]"%r"(inRL), [vRL]"r"(vRL) - : ); - } else { - asm( "smultt %[out], %[inRL], %[vRL] \n" - : [out]"=r"(out) - : [inRL]"%r"(inRL), [vRL]"r"(vRL) - : ); - } - return out; -#else - if (left) { - return int16_t(inRL&0xFFFF) * int16_t(vRL&0xFFFF); - } else { - return int16_t(inRL>>16) * int16_t(vRL>>16); - } -#endif -} - - -void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux) -{ - t->resampler->setSampleRate(t->sampleRate); - - // ramp gain - resample to temp buffer and scale/mix in 2nd step - if (aux != NULL) { - // always resample with unity gain when sending to auxiliary buffer to be able - // to apply send level after resampling - // TODO: modify each resampler to support aux channel? - t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN); - memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t)); - t->resampler->resample(temp, outFrameCount, t->bufferProvider); - if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) { - volumeRampStereo(t, out, outFrameCount, temp, aux); - } else { - volumeStereo(t, out, outFrameCount, temp, aux); - } - } else { - if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) { - t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN); - memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t)); - t->resampler->resample(temp, outFrameCount, t->bufferProvider); - volumeRampStereo(t, out, outFrameCount, temp, aux); - } - - // constant gain - else { - t->resampler->setVolume(t->volume[0], t->volume[1]); - t->resampler->resample(out, outFrameCount, t->bufferProvider); - } - } -} - -void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux) -{ -} - -void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) -{ - int32_t vl = t->prevVolume[0]; - int32_t vr = t->prevVolume[1]; - const int32_t vlInc = t->volumeInc[0]; - const int32_t vrInc = t->volumeInc[1]; - - //LOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", - // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], - // (vl + vlInc*frameCount)/65536.0f, frameCount); - - // ramp volume - if UNLIKELY(aux != NULL) { - int32_t va = t->prevAuxLevel; - const int32_t vaInc = t->auxInc; - int32_t l; - int32_t r; - - do { - l = (*temp++ >> 12); - r = (*temp++ >> 12); - *out++ += (vl >> 16) * l; - *out++ += (vr >> 16) * r; - *aux++ += (va >> 17) * (l + r); - vl += vlInc; - vr += vrInc; - va += vaInc; - } while (--frameCount); - t->prevAuxLevel = va; - } else { - do { - *out++ += (vl >> 16) * (*temp++ >> 12); - *out++ += (vr >> 16) * (*temp++ >> 12); - vl += vlInc; - vr += vrInc; - } while (--frameCount); - } - t->prevVolume[0] = vl; - t->prevVolume[1] = vr; - t->adjustVolumeRamp((aux != NULL)); -} - -void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) -{ - const int16_t vl = t->volume[0]; - const int16_t vr = t->volume[1]; - - if UNLIKELY(aux != NULL) { - const int16_t va = (int16_t)t->auxLevel; - do { - int16_t l = (int16_t)(*temp++ >> 12); - int16_t r = (int16_t)(*temp++ >> 12); - out[0] = mulAdd(l, vl, out[0]); - int16_t a = (int16_t)(((int32_t)l + r) >> 1); - out[1] = mulAdd(r, vr, out[1]); - out += 2; - aux[0] = mulAdd(a, va, aux[0]); - aux++; - } while (--frameCount); - } else { - do { - int16_t l = (int16_t)(*temp++ >> 12); - int16_t r = (int16_t)(*temp++ >> 12); - out[0] = mulAdd(l, vl, out[0]); - out[1] = mulAdd(r, vr, out[1]); - out += 2; - } while (--frameCount); - } -} - -void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) -{ - int16_t const *in = static_cast<int16_t const *>(t->in); - - if UNLIKELY(aux != NULL) { - int32_t l; - int32_t r; - // ramp gain - if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) { - int32_t vl = t->prevVolume[0]; - int32_t vr = t->prevVolume[1]; - int32_t va = t->prevAuxLevel; - const int32_t vlInc = t->volumeInc[0]; - const int32_t vrInc = t->volumeInc[1]; - const int32_t vaInc = t->auxInc; - // LOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", - // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], - // (vl + vlInc*frameCount)/65536.0f, frameCount); - - do { - l = (int32_t)*in++; - r = (int32_t)*in++; - *out++ += (vl >> 16) * l; - *out++ += (vr >> 16) * r; - *aux++ += (va >> 17) * (l + r); - vl += vlInc; - vr += vrInc; - va += vaInc; - } while (--frameCount); - - t->prevVolume[0] = vl; - t->prevVolume[1] = vr; - t->prevAuxLevel = va; - t->adjustVolumeRamp(true); - } - - // constant gain - else { - const uint32_t vrl = t->volumeRL; - const int16_t va = (int16_t)t->auxLevel; - do { - uint32_t rl = *reinterpret_cast<uint32_t const *>(in); - int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1); - in += 2; - out[0] = mulAddRL(1, rl, vrl, out[0]); - out[1] = mulAddRL(0, rl, vrl, out[1]); - out += 2; - aux[0] = mulAdd(a, va, aux[0]); - aux++; - } while (--frameCount); - } - } else { - // ramp gain - if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) { - int32_t vl = t->prevVolume[0]; - int32_t vr = t->prevVolume[1]; - const int32_t vlInc = t->volumeInc[0]; - const int32_t vrInc = t->volumeInc[1]; - - // LOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", - // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], - // (vl + vlInc*frameCount)/65536.0f, frameCount); - - do { - *out++ += (vl >> 16) * (int32_t) *in++; - *out++ += (vr >> 16) * (int32_t) *in++; - vl += vlInc; - vr += vrInc; - } while (--frameCount); - - t->prevVolume[0] = vl; - t->prevVolume[1] = vr; - t->adjustVolumeRamp(false); - } - - // constant gain - else { - const uint32_t vrl = t->volumeRL; - do { - uint32_t rl = *reinterpret_cast<uint32_t const *>(in); - in += 2; - out[0] = mulAddRL(1, rl, vrl, out[0]); - out[1] = mulAddRL(0, rl, vrl, out[1]); - out += 2; - } while (--frameCount); - } - } - t->in = in; -} - -void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) -{ - int16_t const *in = static_cast<int16_t const *>(t->in); - - if UNLIKELY(aux != NULL) { - // ramp gain - if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) { - int32_t vl = t->prevVolume[0]; - int32_t vr = t->prevVolume[1]; - int32_t va = t->prevAuxLevel; - const int32_t vlInc = t->volumeInc[0]; - const int32_t vrInc = t->volumeInc[1]; - const int32_t vaInc = t->auxInc; - - // LOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", - // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], - // (vl + vlInc*frameCount)/65536.0f, frameCount); - - do { - int32_t l = *in++; - *out++ += (vl >> 16) * l; - *out++ += (vr >> 16) * l; - *aux++ += (va >> 16) * l; - vl += vlInc; - vr += vrInc; - va += vaInc; - } while (--frameCount); - - t->prevVolume[0] = vl; - t->prevVolume[1] = vr; - t->prevAuxLevel = va; - t->adjustVolumeRamp(true); - } - // constant gain - else { - const int16_t vl = t->volume[0]; - const int16_t vr = t->volume[1]; - const int16_t va = (int16_t)t->auxLevel; - do { - int16_t l = *in++; - out[0] = mulAdd(l, vl, out[0]); - out[1] = mulAdd(l, vr, out[1]); - out += 2; - aux[0] = mulAdd(l, va, aux[0]); - aux++; - } while (--frameCount); - } - } else { - // ramp gain - if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) { - int32_t vl = t->prevVolume[0]; - int32_t vr = t->prevVolume[1]; - const int32_t vlInc = t->volumeInc[0]; - const int32_t vrInc = t->volumeInc[1]; - - // LOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", - // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], - // (vl + vlInc*frameCount)/65536.0f, frameCount); - - do { - int32_t l = *in++; - *out++ += (vl >> 16) * l; - *out++ += (vr >> 16) * l; - vl += vlInc; - vr += vrInc; - } while (--frameCount); - - t->prevVolume[0] = vl; - t->prevVolume[1] = vr; - t->adjustVolumeRamp(false); - } - // constant gain - else { - const int16_t vl = t->volume[0]; - const int16_t vr = t->volume[1]; - do { - int16_t l = *in++; - out[0] = mulAdd(l, vl, out[0]); - out[1] = mulAdd(l, vr, out[1]); - out += 2; - } while (--frameCount); - } - } - t->in = in; -} - -void AudioMixer::ditherAndClamp(int32_t* out, int32_t const *sums, size_t c) -{ - for (size_t i=0 ; i<c ; i++) { - int32_t l = *sums++; - int32_t r = *sums++; - int32_t nl = l >> 12; - int32_t nr = r >> 12; - l = clamp16(nl); - r = clamp16(nr); - *out++ = (r<<16) | (l & 0xFFFF); - } -} - -// no-op case -void AudioMixer::process__nop(state_t* state) -{ - uint32_t e0 = state->enabledTracks; - size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS; - while (e0) { - // process by group of tracks with same output buffer to - // avoid multiple memset() on same buffer - uint32_t e1 = e0, e2 = e0; - int i = 31 - __builtin_clz(e1); - track_t& t1 = state->tracks[i]; - e2 &= ~(1<<i); - while (e2) { - i = 31 - __builtin_clz(e2); - e2 &= ~(1<<i); - track_t& t2 = state->tracks[i]; - if UNLIKELY(t2.mainBuffer != t1.mainBuffer) { - e1 &= ~(1<<i); - } - } - e0 &= ~(e1); - - memset(t1.mainBuffer, 0, bufSize); - - while (e1) { - i = 31 - __builtin_clz(e1); - e1 &= ~(1<<i); - t1 = state->tracks[i]; - size_t outFrames = state->frameCount; - while (outFrames) { - t1.buffer.frameCount = outFrames; - t1.bufferProvider->getNextBuffer(&t1.buffer); - if (!t1.buffer.raw) break; - outFrames -= t1.buffer.frameCount; - t1.bufferProvider->releaseBuffer(&t1.buffer); - } - } - } -} - -// generic code without resampling -void AudioMixer::process__genericNoResampling(state_t* state) -{ - int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32))); - - // acquire each track's buffer - uint32_t enabledTracks = state->enabledTracks; - uint32_t e0 = enabledTracks; - while (e0) { - const int i = 31 - __builtin_clz(e0); - e0 &= ~(1<<i); - track_t& t = state->tracks[i]; - t.buffer.frameCount = state->frameCount; - t.bufferProvider->getNextBuffer(&t.buffer); - t.frameCount = t.buffer.frameCount; - t.in = t.buffer.raw; - // t.in == NULL can happen if the track was flushed just after having - // been enabled for mixing. - if (t.in == NULL) - enabledTracks &= ~(1<<i); - } - - e0 = enabledTracks; - while (e0) { - // process by group of tracks with same output buffer to - // optimize cache use - uint32_t e1 = e0, e2 = e0; - int j = 31 - __builtin_clz(e1); - track_t& t1 = state->tracks[j]; - e2 &= ~(1<<j); - while (e2) { - j = 31 - __builtin_clz(e2); - e2 &= ~(1<<j); - track_t& t2 = state->tracks[j]; - if UNLIKELY(t2.mainBuffer != t1.mainBuffer) { - e1 &= ~(1<<j); - } - } - e0 &= ~(e1); - // this assumes output 16 bits stereo, no resampling - int32_t *out = t1.mainBuffer; - size_t numFrames = 0; - do { - memset(outTemp, 0, sizeof(outTemp)); - e2 = e1; - while (e2) { - const int i = 31 - __builtin_clz(e2); - e2 &= ~(1<<i); - track_t& t = state->tracks[i]; - size_t outFrames = BLOCKSIZE; - int32_t *aux = NULL; - if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) { - aux = t.auxBuffer + numFrames; - } - while (outFrames) { - size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount; - if (inFrames) { - (t.hook)(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux); - t.frameCount -= inFrames; - outFrames -= inFrames; - if UNLIKELY(aux != NULL) { - aux += inFrames; - } - } - if (t.frameCount == 0 && outFrames) { - t.bufferProvider->releaseBuffer(&t.buffer); - t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames); - t.bufferProvider->getNextBuffer(&t.buffer); - t.in = t.buffer.raw; - if (t.in == NULL) { - enabledTracks &= ~(1<<i); - e1 &= ~(1<<i); - break; - } - t.frameCount = t.buffer.frameCount; - } - } - } - ditherAndClamp(out, outTemp, BLOCKSIZE); - out += BLOCKSIZE; - numFrames += BLOCKSIZE; - } while (numFrames < state->frameCount); - } - - // release each track's buffer - e0 = enabledTracks; - while (e0) { - const int i = 31 - __builtin_clz(e0); - e0 &= ~(1<<i); - track_t& t = state->tracks[i]; - t.bufferProvider->releaseBuffer(&t.buffer); - } -} - - - // generic code with resampling -void AudioMixer::process__genericResampling(state_t* state) -{ - int32_t* const outTemp = state->outputTemp; - const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount; - memset(outTemp, 0, size); - - size_t numFrames = state->frameCount; - - uint32_t e0 = state->enabledTracks; - while (e0) { - // process by group of tracks with same output buffer - // to optimize cache use - uint32_t e1 = e0, e2 = e0; - int j = 31 - __builtin_clz(e1); - track_t& t1 = state->tracks[j]; - e2 &= ~(1<<j); - while (e2) { - j = 31 - __builtin_clz(e2); - e2 &= ~(1<<j); - track_t& t2 = state->tracks[j]; - if UNLIKELY(t2.mainBuffer != t1.mainBuffer) { - e1 &= ~(1<<j); - } - } - e0 &= ~(e1); - int32_t *out = t1.mainBuffer; - while (e1) { - const int i = 31 - __builtin_clz(e1); - e1 &= ~(1<<i); - track_t& t = state->tracks[i]; - int32_t *aux = NULL; - if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) { - aux = t.auxBuffer; - } - - // this is a little goofy, on the resampling case we don't - // acquire/release the buffers because it's done by - // the resampler. - if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) { - (t.hook)(&t, outTemp, numFrames, state->resampleTemp, aux); - } else { - - size_t outFrames = 0; - - while (outFrames < numFrames) { - t.buffer.frameCount = numFrames - outFrames; - t.bufferProvider->getNextBuffer(&t.buffer); - t.in = t.buffer.raw; - // t.in == NULL can happen if the track was flushed just after having - // been enabled for mixing. - if (t.in == NULL) break; - - if UNLIKELY(aux != NULL) { - aux += outFrames; - } - (t.hook)(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux); - outFrames += t.buffer.frameCount; - t.bufferProvider->releaseBuffer(&t.buffer); - } - } - } - ditherAndClamp(out, outTemp, numFrames); - } -} - -// one track, 16 bits stereo without resampling is the most common case -void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state) -{ - const int i = 31 - __builtin_clz(state->enabledTracks); - const track_t& t = state->tracks[i]; - - AudioBufferProvider::Buffer& b(t.buffer); - - int32_t* out = t.mainBuffer; - size_t numFrames = state->frameCount; - - const int16_t vl = t.volume[0]; - const int16_t vr = t.volume[1]; - const uint32_t vrl = t.volumeRL; - while (numFrames) { - b.frameCount = numFrames; - t.bufferProvider->getNextBuffer(&b); - int16_t const *in = b.i16; - - // in == NULL can happen if the track was flushed just after having - // been enabled for mixing. - if (in == NULL || ((unsigned long)in & 3)) { - memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t)); - LOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: buffer %p track %d, channels %d, needs %08x", - in, i, t.channelCount, t.needs); - return; - } - size_t outFrames = b.frameCount; - - if (UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) { - // volume is boosted, so we might need to clamp even though - // we process only one track. - do { - uint32_t rl = *reinterpret_cast<uint32_t const *>(in); - in += 2; - int32_t l = mulRL(1, rl, vrl) >> 12; - int32_t r = mulRL(0, rl, vrl) >> 12; - // clamping... - l = clamp16(l); - r = clamp16(r); - *out++ = (r<<16) | (l & 0xFFFF); - } while (--outFrames); - } else { - do { - uint32_t rl = *reinterpret_cast<uint32_t const *>(in); - in += 2; - int32_t l = mulRL(1, rl, vrl) >> 12; - int32_t r = mulRL(0, rl, vrl) >> 12; - *out++ = (r<<16) | (l & 0xFFFF); - } while (--outFrames); - } - numFrames -= b.frameCount; - t.bufferProvider->releaseBuffer(&b); - } -} - -// 2 tracks is also a common case -// NEVER used in current implementation of process__validate() -// only use if the 2 tracks have the same output buffer -void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state) -{ - int i; - uint32_t en = state->enabledTracks; - - i = 31 - __builtin_clz(en); - const track_t& t0 = state->tracks[i]; - AudioBufferProvider::Buffer& b0(t0.buffer); - - en &= ~(1<<i); - i = 31 - __builtin_clz(en); - const track_t& t1 = state->tracks[i]; - AudioBufferProvider::Buffer& b1(t1.buffer); - - int16_t const *in0; - const int16_t vl0 = t0.volume[0]; - const int16_t vr0 = t0.volume[1]; - size_t frameCount0 = 0; - - int16_t const *in1; - const int16_t vl1 = t1.volume[0]; - const int16_t vr1 = t1.volume[1]; - size_t frameCount1 = 0; - - //FIXME: only works if two tracks use same buffer - int32_t* out = t0.mainBuffer; - size_t numFrames = state->frameCount; - int16_t const *buff = NULL; - - - while (numFrames) { - - if (frameCount0 == 0) { - b0.frameCount = numFrames; - t0.bufferProvider->getNextBuffer(&b0); - if (b0.i16 == NULL) { - if (buff == NULL) { - buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount]; - } - in0 = buff; - b0.frameCount = numFrames; - } else { - in0 = b0.i16; - } - frameCount0 = b0.frameCount; - } - if (frameCount1 == 0) { - b1.frameCount = numFrames; - t1.bufferProvider->getNextBuffer(&b1); - if (b1.i16 == NULL) { - if (buff == NULL) { - buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount]; - } - in1 = buff; - b1.frameCount = numFrames; - } else { - in1 = b1.i16; - } - frameCount1 = b1.frameCount; - } - - size_t outFrames = frameCount0 < frameCount1?frameCount0:frameCount1; - - numFrames -= outFrames; - frameCount0 -= outFrames; - frameCount1 -= outFrames; - - do { - int32_t l0 = *in0++; - int32_t r0 = *in0++; - l0 = mul(l0, vl0); - r0 = mul(r0, vr0); - int32_t l = *in1++; - int32_t r = *in1++; - l = mulAdd(l, vl1, l0) >> 12; - r = mulAdd(r, vr1, r0) >> 12; - // clamping... - l = clamp16(l); - r = clamp16(r); - *out++ = (r<<16) | (l & 0xFFFF); - } while (--outFrames); - - if (frameCount0 == 0) { - t0.bufferProvider->releaseBuffer(&b0); - } - if (frameCount1 == 0) { - t1.bufferProvider->releaseBuffer(&b1); - } - } - - if (buff != NULL) { - delete [] buff; - } -} - -// ---------------------------------------------------------------------------- -}; // namespace android - diff --git a/libs/audioflinger/AudioMixer.h b/libs/audioflinger/AudioMixer.h deleted file mode 100644 index aee3e17..0000000 --- a/libs/audioflinger/AudioMixer.h +++ /dev/null @@ -1,207 +0,0 @@ -/* //device/include/server/AudioFlinger/AudioMixer.h -** -** Copyright 2007, The Android Open Source Project -** -** Licensed under the Apache License, Version 2.0 (the "License"); -** you may not use this file except in compliance with the License. -** You may obtain a copy of the License at -** -** http://www.apache.org/licenses/LICENSE-2.0 -** -** Unless required by applicable law or agreed to in writing, software -** distributed under the License is distributed on an "AS IS" BASIS, -** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -** See the License for the specific language governing permissions and -** limitations under the License. -*/ - -#ifndef ANDROID_AUDIO_MIXER_H -#define ANDROID_AUDIO_MIXER_H - -#include <stdint.h> -#include <sys/types.h> - -#include "AudioBufferProvider.h" -#include "AudioResampler.h" - -namespace android { - -// ---------------------------------------------------------------------------- - -#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true )) -#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false )) - -// ---------------------------------------------------------------------------- - -class AudioMixer -{ -public: - AudioMixer(size_t frameCount, uint32_t sampleRate); - - ~AudioMixer(); - - static const uint32_t MAX_NUM_TRACKS = 32; - static const uint32_t MAX_NUM_CHANNELS = 2; - - static const uint16_t UNITY_GAIN = 0x1000; - - enum { // names - - // track units (32 units) - TRACK0 = 0x1000, - - // enable/disable - MIXING = 0x2000, - - // setParameter targets - TRACK = 0x3000, - RESAMPLE = 0x3001, - RAMP_VOLUME = 0x3002, // ramp to new volume - VOLUME = 0x3003, // don't ramp - - // set Parameter names - // for target TRACK - CHANNEL_COUNT = 0x4000, - FORMAT = 0x4001, - MAIN_BUFFER = 0x4002, - AUX_BUFFER = 0x4003, - // for TARGET RESAMPLE - SAMPLE_RATE = 0x4100, - // for TARGET VOLUME (8 channels max) - VOLUME0 = 0x4200, - VOLUME1 = 0x4201, - AUXLEVEL = 0x4210, - }; - - - int getTrackName(); - void deleteTrackName(int name); - - status_t enable(int name); - status_t disable(int name); - - status_t setActiveTrack(int track); - status_t setParameter(int target, int name, void *value); - - status_t setBufferProvider(AudioBufferProvider* bufferProvider); - void process(); - - uint32_t trackNames() const { return mTrackNames; } - - static void ditherAndClamp(int32_t* out, int32_t const *sums, size_t c); - -private: - - enum { - NEEDS_CHANNEL_COUNT__MASK = 0x00000003, - NEEDS_FORMAT__MASK = 0x000000F0, - NEEDS_MUTE__MASK = 0x00000100, - NEEDS_RESAMPLE__MASK = 0x00001000, - NEEDS_AUX__MASK = 0x00010000, - }; - - enum { - NEEDS_CHANNEL_1 = 0x00000000, - NEEDS_CHANNEL_2 = 0x00000001, - - NEEDS_FORMAT_16 = 0x00000010, - - NEEDS_MUTE_DISABLED = 0x00000000, - NEEDS_MUTE_ENABLED = 0x00000100, - - NEEDS_RESAMPLE_DISABLED = 0x00000000, - NEEDS_RESAMPLE_ENABLED = 0x00001000, - - NEEDS_AUX_DISABLED = 0x00000000, - NEEDS_AUX_ENABLED = 0x00010000, - }; - - static inline int32_t applyVolume(int32_t in, int32_t v) { - return in * v; - } - - - struct state_t; - struct track_t; - - typedef void (*mix_t)(state_t* state); - typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux); - static const int BLOCKSIZE = 16; // 4 cache lines - - struct track_t { - uint32_t needs; - - union { - int16_t volume[2]; // [0]3.12 fixed point - int32_t volumeRL; - }; - - int32_t prevVolume[2]; - - int32_t volumeInc[2]; - int32_t auxLevel; - int32_t auxInc; - int32_t prevAuxLevel; - - uint16_t frameCount; - - uint8_t channelCount : 4; - uint8_t enabled : 1; - uint8_t reserved0 : 3; - uint8_t format; - - AudioBufferProvider* bufferProvider; - mutable AudioBufferProvider::Buffer buffer; - - hook_t hook; - void const* in; // current location in buffer - - AudioResampler* resampler; - uint32_t sampleRate; - int32_t* mainBuffer; - int32_t* auxBuffer; - - bool setResampler(uint32_t sampleRate, uint32_t devSampleRate); - bool doesResample() const; - void adjustVolumeRamp(bool aux); - }; - - // pad to 32-bytes to fill cache line - struct state_t { - uint32_t enabledTracks; - uint32_t needsChanged; - size_t frameCount; - mix_t hook; - int32_t *outputTemp; - int32_t *resampleTemp; - int32_t reserved[2]; - track_t tracks[32]; __attribute__((aligned(32))); - }; - - int mActiveTrack; - uint32_t mTrackNames; - const uint32_t mSampleRate; - - state_t mState __attribute__((aligned(32))); - - void invalidateState(uint32_t mask); - - static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); - static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); - static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); - static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); - static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); - static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); - - static void process__validate(state_t* state); - static void process__nop(state_t* state); - static void process__genericNoResampling(state_t* state); - static void process__genericResampling(state_t* state); - static void process__OneTrack16BitsStereoNoResampling(state_t* state); - static void process__TwoTracks16BitsStereoNoResampling(state_t* state); -}; - -// ---------------------------------------------------------------------------- -}; // namespace android - -#endif // ANDROID_AUDIO_MIXER_H diff --git a/libs/audioflinger/AudioPolicyManagerBase.cpp b/libs/audioflinger/AudioPolicyManagerBase.cpp deleted file mode 100644 index 381a958..0000000 --- a/libs/audioflinger/AudioPolicyManagerBase.cpp +++ /dev/null @@ -1,1973 +0,0 @@ -/* - * Copyright (C) 2009 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#define LOG_TAG "AudioPolicyManagerBase" -//#define LOG_NDEBUG 0 -#include <utils/Log.h> -#include <hardware_legacy/AudioPolicyManagerBase.h> -#include <media/mediarecorder.h> - -namespace android { - - -// ---------------------------------------------------------------------------- -// AudioPolicyInterface implementation -// ---------------------------------------------------------------------------- - - -status_t AudioPolicyManagerBase::setDeviceConnectionState(AudioSystem::audio_devices device, - AudioSystem::device_connection_state state, - const char *device_address) -{ - - LOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address); - - // connect/disconnect only 1 device at a time - if (AudioSystem::popCount(device) != 1) return BAD_VALUE; - - if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) { - LOGE("setDeviceConnectionState() invalid address: %s", device_address); - return BAD_VALUE; - } - - // handle output devices - if (AudioSystem::isOutputDevice(device)) { - -#ifndef WITH_A2DP - if (AudioSystem::isA2dpDevice(device)) { - LOGE("setDeviceConnectionState() invalid device: %x", device); - return BAD_VALUE; - } -#endif - - switch (state) - { - // handle output device connection - case AudioSystem::DEVICE_STATE_AVAILABLE: - if (mAvailableOutputDevices & device) { - LOGW("setDeviceConnectionState() device already connected: %x", device); - return INVALID_OPERATION; - } - LOGV("setDeviceConnectionState() connecting device %x", device); - - // register new device as available - mAvailableOutputDevices |= device; - -#ifdef WITH_A2DP - // handle A2DP device connection - if (AudioSystem::isA2dpDevice(device)) { - status_t status = handleA2dpConnection(device, device_address); - if (status != NO_ERROR) { - mAvailableOutputDevices &= ~device; - return status; - } - } else -#endif - { - if (AudioSystem::isBluetoothScoDevice(device)) { - LOGV("setDeviceConnectionState() BT SCO device, address %s", device_address); - // keep track of SCO device address - mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN); -#ifdef WITH_A2DP - if (mA2dpOutput != 0 && - mPhoneState != AudioSystem::MODE_NORMAL) { - mpClientInterface->suspendOutput(mA2dpOutput); - } -#endif - } - } - break; - // handle output device disconnection - case AudioSystem::DEVICE_STATE_UNAVAILABLE: { - if (!(mAvailableOutputDevices & device)) { - LOGW("setDeviceConnectionState() device not connected: %x", device); - return INVALID_OPERATION; - } - - - LOGV("setDeviceConnectionState() disconnecting device %x", device); - // remove device from available output devices - mAvailableOutputDevices &= ~device; - -#ifdef WITH_A2DP - // handle A2DP device disconnection - if (AudioSystem::isA2dpDevice(device)) { - status_t status = handleA2dpDisconnection(device, device_address); - if (status != NO_ERROR) { - mAvailableOutputDevices |= device; - return status; - } - } else -#endif - { - if (AudioSystem::isBluetoothScoDevice(device)) { - mScoDeviceAddress = ""; -#ifdef WITH_A2DP - if (mA2dpOutput != 0 && - mPhoneState != AudioSystem::MODE_NORMAL) { - mpClientInterface->restoreOutput(mA2dpOutput); - } -#endif - } - } - } break; - - default: - LOGE("setDeviceConnectionState() invalid state: %x", state); - return BAD_VALUE; - } - - // request routing change if necessary - uint32_t newDevice = getNewDevice(mHardwareOutput, false); -#ifdef WITH_A2DP - checkOutputForAllStrategies(newDevice); - // A2DP outputs must be closed after checkOutputForAllStrategies() is executed - if (state == AudioSystem::DEVICE_STATE_UNAVAILABLE && AudioSystem::isA2dpDevice(device)) { - closeA2dpOutputs(); - } -#endif - updateDeviceForStrategy(); - setOutputDevice(mHardwareOutput, newDevice); - - if (device == AudioSystem::DEVICE_OUT_WIRED_HEADSET) { - device = AudioSystem::DEVICE_IN_WIRED_HEADSET; - } else if (device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO || - device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET || - device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT) { - device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET; - } else { - return NO_ERROR; - } - } - // handle input devices - if (AudioSystem::isInputDevice(device)) { - - switch (state) - { - // handle input device connection - case AudioSystem::DEVICE_STATE_AVAILABLE: { - if (mAvailableInputDevices & device) { - LOGW("setDeviceConnectionState() device already connected: %d", device); - return INVALID_OPERATION; - } - mAvailableInputDevices |= device; - } - break; - - // handle input device disconnection - case AudioSystem::DEVICE_STATE_UNAVAILABLE: { - if (!(mAvailableInputDevices & device)) { - LOGW("setDeviceConnectionState() device not connected: %d", device); - return INVALID_OPERATION; - } - mAvailableInputDevices &= ~device; - } break; - - default: - LOGE("setDeviceConnectionState() invalid state: %x", state); - return BAD_VALUE; - } - - audio_io_handle_t activeInput = getActiveInput(); - if (activeInput != 0) { - AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput); - uint32_t newDevice = getDeviceForInputSource(inputDesc->mInputSource); - if (newDevice != inputDesc->mDevice) { - LOGV("setDeviceConnectionState() changing device from %x to %x for input %d", - inputDesc->mDevice, newDevice, activeInput); - inputDesc->mDevice = newDevice; - AudioParameter param = AudioParameter(); - param.addInt(String8(AudioParameter::keyRouting), (int)newDevice); - mpClientInterface->setParameters(activeInput, param.toString()); - } - } - - return NO_ERROR; - } - - LOGW("setDeviceConnectionState() invalid device: %x", device); - return BAD_VALUE; -} - -AudioSystem::device_connection_state AudioPolicyManagerBase::getDeviceConnectionState(AudioSystem::audio_devices device, - const char *device_address) -{ - AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE; - String8 address = String8(device_address); - if (AudioSystem::isOutputDevice(device)) { - if (device & mAvailableOutputDevices) { -#ifdef WITH_A2DP - if (AudioSystem::isA2dpDevice(device) && - address != "" && mA2dpDeviceAddress != address) { - return state; - } -#endif - if (AudioSystem::isBluetoothScoDevice(device) && - address != "" && mScoDeviceAddress != address) { - return state; - } - state = AudioSystem::DEVICE_STATE_AVAILABLE; - } - } else if (AudioSystem::isInputDevice(device)) { - if (device & mAvailableInputDevices) { - state = AudioSystem::DEVICE_STATE_AVAILABLE; - } - } - - return state; -} - -void AudioPolicyManagerBase::setPhoneState(int state) -{ - LOGV("setPhoneState() state %d", state); - uint32_t newDevice = 0; - if (state < 0 || state >= AudioSystem::NUM_MODES) { - LOGW("setPhoneState() invalid state %d", state); - return; - } - - if (state == mPhoneState ) { - LOGW("setPhoneState() setting same state %d", state); - return; - } - - // if leaving call state, handle special case of active streams - // pertaining to sonification strategy see handleIncallSonification() - if (mPhoneState == AudioSystem::MODE_IN_CALL) { - LOGV("setPhoneState() in call state management: new state is %d", state); - for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { - handleIncallSonification(stream, false, true); - } - } - - // store previous phone state for management of sonification strategy below - int oldState = mPhoneState; - mPhoneState = state; - bool force = false; - - // are we entering or starting a call - if ((oldState != AudioSystem::MODE_IN_CALL) && (state == AudioSystem::MODE_IN_CALL)) { - LOGV(" Entering call in setPhoneState()"); - // force routing command to audio hardware when starting a call - // even if no device change is needed - force = true; - } else if ((oldState == AudioSystem::MODE_IN_CALL) && (state != AudioSystem::MODE_IN_CALL)) { - LOGV(" Exiting call in setPhoneState()"); - // force routing command to audio hardware when exiting a call - // even if no device change is needed - force = true; - } - - // check for device and output changes triggered by new phone state - newDevice = getNewDevice(mHardwareOutput, false); -#ifdef WITH_A2DP - checkOutputForAllStrategies(newDevice); - // suspend A2DP output if a SCO device is present. - if (mA2dpOutput != 0 && mScoDeviceAddress != "") { - if (oldState == AudioSystem::MODE_NORMAL) { - mpClientInterface->suspendOutput(mA2dpOutput); - } else if (state == AudioSystem::MODE_NORMAL) { - mpClientInterface->restoreOutput(mA2dpOutput); - } - } -#endif - updateDeviceForStrategy(); - - AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput); - - // force routing command to audio hardware when ending call - // even if no device change is needed - if (oldState == AudioSystem::MODE_IN_CALL && newDevice == 0) { - newDevice = hwOutputDesc->device(); - } - - // when changing from ring tone to in call mode, mute the ringing tone - // immediately and delay the route change to avoid sending the ring tone - // tail into the earpiece or headset. - int delayMs = 0; - if (state == AudioSystem::MODE_IN_CALL && oldState == AudioSystem::MODE_RINGTONE) { - // delay the device change command by twice the output latency to have some margin - // and be sure that audio buffers not yet affected by the mute are out when - // we actually apply the route change - delayMs = hwOutputDesc->mLatency*2; - setStreamMute(AudioSystem::RING, true, mHardwareOutput); - } - - // change routing is necessary - setOutputDevice(mHardwareOutput, newDevice, force, delayMs); - - // if entering in call state, handle special case of active streams - // pertaining to sonification strategy see handleIncallSonification() - if (state == AudioSystem::MODE_IN_CALL) { - LOGV("setPhoneState() in call state management: new state is %d", state); - // unmute the ringing tone after a sufficient delay if it was muted before - // setting output device above - if (oldState == AudioSystem::MODE_RINGTONE) { - setStreamMute(AudioSystem::RING, false, mHardwareOutput, MUTE_TIME_MS); - } - for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { - handleIncallSonification(stream, true, true); - } - } - - // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE - if (state == AudioSystem::MODE_RINGTONE && - (hwOutputDesc->mRefCount[AudioSystem::MUSIC] || - (systemTime() - mMusicStopTime) < seconds(SONIFICATION_HEADSET_MUSIC_DELAY))) { - mLimitRingtoneVolume = true; - } else { - mLimitRingtoneVolume = false; - } -} - -void AudioPolicyManagerBase::setRingerMode(uint32_t mode, uint32_t mask) -{ - LOGV("setRingerMode() mode %x, mask %x", mode, mask); - - mRingerMode = mode; -} - -void AudioPolicyManagerBase::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) -{ - LOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState); - - bool forceVolumeReeval = false; - switch(usage) { - case AudioSystem::FOR_COMMUNICATION: - if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO && - config != AudioSystem::FORCE_NONE) { - LOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config); - return; - } - mForceUse[usage] = config; - break; - case AudioSystem::FOR_MEDIA: - if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP && - config != AudioSystem::FORCE_WIRED_ACCESSORY && config != AudioSystem::FORCE_NONE) { - LOGW("setForceUse() invalid config %d for FOR_MEDIA", config); - return; - } - mForceUse[usage] = config; - break; - case AudioSystem::FOR_RECORD: - if (config != AudioSystem::FORCE_BT_SCO && config != AudioSystem::FORCE_WIRED_ACCESSORY && - config != AudioSystem::FORCE_NONE) { - LOGW("setForceUse() invalid config %d for FOR_RECORD", config); - return; - } - mForceUse[usage] = config; - break; - case AudioSystem::FOR_DOCK: - if (config != AudioSystem::FORCE_NONE && config != AudioSystem::FORCE_BT_CAR_DOCK && - config != AudioSystem::FORCE_BT_DESK_DOCK && config != AudioSystem::FORCE_WIRED_ACCESSORY) { - LOGW("setForceUse() invalid config %d for FOR_DOCK", config); - } - forceVolumeReeval = true; - mForceUse[usage] = config; - break; - default: - LOGW("setForceUse() invalid usage %d", usage); - break; - } - - // check for device and output changes triggered by new phone state - uint32_t newDevice = getNewDevice(mHardwareOutput, false); -#ifdef WITH_A2DP - checkOutputForAllStrategies(newDevice); -#endif - updateDeviceForStrategy(); - setOutputDevice(mHardwareOutput, newDevice); - if (forceVolumeReeval) { - applyStreamVolumes(mHardwareOutput, newDevice); - } -} - -AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage) -{ - return mForceUse[usage]; -} - -void AudioPolicyManagerBase::setSystemProperty(const char* property, const char* value) -{ - LOGV("setSystemProperty() property %s, value %s", property, value); - if (strcmp(property, "ro.camera.sound.forced") == 0) { - if (atoi(value)) { - LOGV("ENFORCED_AUDIBLE cannot be muted"); - mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = false; - } else { - LOGV("ENFORCED_AUDIBLE can be muted"); - mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = true; - } - } -} - -audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream, - uint32_t samplingRate, - uint32_t format, - uint32_t channels, - AudioSystem::output_flags flags) -{ - audio_io_handle_t output = 0; - uint32_t latency = 0; - routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream); - uint32_t device = getDeviceForStrategy(strategy); - LOGV("getOutput() stream %d, samplingRate %d, format %d, channels %x, flags %x", stream, samplingRate, format, channels, flags); - -#ifdef AUDIO_POLICY_TEST - if (mCurOutput != 0) { - LOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channels %x, mDirectOutput %d", - mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput); - - if (mTestOutputs[mCurOutput] == 0) { - LOGV("getOutput() opening test output"); - AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(); - outputDesc->mDevice = mTestDevice; - outputDesc->mSamplingRate = mTestSamplingRate; - outputDesc->mFormat = mTestFormat; - outputDesc->mChannels = mTestChannels; - outputDesc->mLatency = mTestLatencyMs; - outputDesc->mFlags = (AudioSystem::output_flags)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0); - outputDesc->mRefCount[stream] = 0; - mTestOutputs[mCurOutput] = mpClientInterface->openOutput(&outputDesc->mDevice, - &outputDesc->mSamplingRate, - &outputDesc->mFormat, - &outputDesc->mChannels, - &outputDesc->mLatency, - outputDesc->mFlags); - if (mTestOutputs[mCurOutput]) { - AudioParameter outputCmd = AudioParameter(); - outputCmd.addInt(String8("set_id"),mCurOutput); - mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString()); - addOutput(mTestOutputs[mCurOutput], outputDesc); - } - } - return mTestOutputs[mCurOutput]; - } -#endif //AUDIO_POLICY_TEST - - // open a direct output if required by specified parameters - if (needsDirectOuput(stream, samplingRate, format, channels, flags, device)) { - - LOGV("getOutput() opening direct output device %x", device); - AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(); - outputDesc->mDevice = device; - outputDesc->mSamplingRate = samplingRate; - outputDesc->mFormat = format; - outputDesc->mChannels = channels; - outputDesc->mLatency = 0; - outputDesc->mFlags = (AudioSystem::output_flags)(flags | AudioSystem::OUTPUT_FLAG_DIRECT); - outputDesc->mRefCount[stream] = 0; - output = mpClientInterface->openOutput(&outputDesc->mDevice, - &outputDesc->mSamplingRate, - &outputDesc->mFormat, - &outputDesc->mChannels, - &outputDesc->mLatency, - outputDesc->mFlags); - - // only accept an output with the requeted parameters - if (output == 0 || - (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) || - (format != 0 && format != outputDesc->mFormat) || - (channels != 0 && channels != outputDesc->mChannels)) { - LOGV("getOutput() failed opening direct output: samplingRate %d, format %d, channels %d", - samplingRate, format, channels); - if (output != 0) { - mpClientInterface->closeOutput(output); - } - delete outputDesc; - return 0; - } - addOutput(output, outputDesc); - return output; - } - - if (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO && - channels != AudioSystem::CHANNEL_OUT_STEREO) { - return 0; - } - // open a non direct output - - // get which output is suitable for the specified stream. The actual routing change will happen - // when startOutput() will be called - uint32_t a2dpDevice = device & AudioSystem::DEVICE_OUT_ALL_A2DP; - if (AudioSystem::popCount((AudioSystem::audio_devices)device) == 2) { -#ifdef WITH_A2DP - if (a2dpUsedForSonification() && a2dpDevice != 0) { - // if playing on 2 devices among which one is A2DP, use duplicated output - LOGV("getOutput() using duplicated output"); - LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device in multiple %x selected but A2DP output not opened", device); - output = mDuplicatedOutput; - } else -#endif - { - // if playing on 2 devices among which none is A2DP, use hardware output - output = mHardwareOutput; - } - LOGV("getOutput() using output %d for 2 devices %x", output, device); - } else { -#ifdef WITH_A2DP - if (a2dpDevice != 0) { - // if playing on A2DP device, use a2dp output - LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device %x selected but A2DP output not opened", device); - output = mA2dpOutput; - } else -#endif - { - // if playing on not A2DP device, use hardware output - output = mHardwareOutput; - } - } - - - LOGW_IF((output ==0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x", - stream, samplingRate, format, channels, flags); - - return output; -} - -status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream) -{ - LOGV("startOutput() output %d, stream %d", output, stream); - ssize_t index = mOutputs.indexOfKey(output); - if (index < 0) { - LOGW("startOutput() unknow output %d", output); - return BAD_VALUE; - } - - AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); - routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream); - -#ifdef WITH_A2DP - if (mA2dpOutput != 0 && !a2dpUsedForSonification() && strategy == STRATEGY_SONIFICATION) { - setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput); - } -#endif - - // incremenent usage count for this stream on the requested output: - // NOTE that the usage count is the same for duplicated output and hardware output which is - // necassary for a correct control of hardware output routing by startOutput() and stopOutput() - outputDesc->changeRefCount(stream, 1); - - setOutputDevice(output, getNewDevice(output)); - - // handle special case for sonification while in call - if (mPhoneState == AudioSystem::MODE_IN_CALL) { - handleIncallSonification(stream, true, false); - } - - // apply volume rules for current stream and device if necessary - checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, outputDesc->device()); - - return NO_ERROR; -} - -status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream) -{ - LOGV("stopOutput() output %d, stream %d", output, stream); - ssize_t index = mOutputs.indexOfKey(output); - if (index < 0) { - LOGW("stopOutput() unknow output %d", output); - return BAD_VALUE; - } - - AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); - routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream); - - // handle special case for sonification while in call - if (mPhoneState == AudioSystem::MODE_IN_CALL) { - handleIncallSonification(stream, false, false); - } - - if (outputDesc->mRefCount[stream] > 0) { - // decrement usage count of this stream on the output - outputDesc->changeRefCount(stream, -1); - // store time at which the last music track was stopped - see computeVolume() - if (stream == AudioSystem::MUSIC) { - mMusicStopTime = systemTime(); - } - - setOutputDevice(output, getNewDevice(output)); - -#ifdef WITH_A2DP - if (mA2dpOutput != 0 && !a2dpUsedForSonification() && strategy == STRATEGY_SONIFICATION) { - setStrategyMute(STRATEGY_MEDIA, false, mA2dpOutput, mOutputs.valueFor(mHardwareOutput)->mLatency*2); - } -#endif - if (output != mHardwareOutput) { - setOutputDevice(mHardwareOutput, getNewDevice(mHardwareOutput), true); - } - return NO_ERROR; - } else { - LOGW("stopOutput() refcount is already 0 for output %d", output); - return INVALID_OPERATION; - } -} - -void AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output) -{ - LOGV("releaseOutput() %d", output); - ssize_t index = mOutputs.indexOfKey(output); - if (index < 0) { - LOGW("releaseOutput() releasing unknown output %d", output); - return; - } - -#ifdef AUDIO_POLICY_TEST - int testIndex = testOutputIndex(output); - if (testIndex != 0) { - AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); - if (outputDesc->refCount() == 0) { - mpClientInterface->closeOutput(output); - delete mOutputs.valueAt(index); - mOutputs.removeItem(output); - mTestOutputs[testIndex] = 0; - } - return; - } -#endif //AUDIO_POLICY_TEST - - if (mOutputs.valueAt(index)->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) { - mpClientInterface->closeOutput(output); - delete mOutputs.valueAt(index); - mOutputs.removeItem(output); - } -} - -audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource, - uint32_t samplingRate, - uint32_t format, - uint32_t channels, - AudioSystem::audio_in_acoustics acoustics) -{ - audio_io_handle_t input = 0; - uint32_t device = getDeviceForInputSource(inputSource); - - LOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics); - - if (device == 0) { - return 0; - } - - // adapt channel selection to input source - switch(inputSource) { - case AUDIO_SOURCE_VOICE_UPLINK: - channels = AudioSystem::CHANNEL_IN_VOICE_UPLINK; - break; - case AUDIO_SOURCE_VOICE_DOWNLINK: - channels = AudioSystem::CHANNEL_IN_VOICE_DNLINK; - break; - case AUDIO_SOURCE_VOICE_CALL: - channels = (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK); - break; - default: - break; - } - - AudioInputDescriptor *inputDesc = new AudioInputDescriptor(); - - inputDesc->mInputSource = inputSource; - inputDesc->mDevice = device; - inputDesc->mSamplingRate = samplingRate; - inputDesc->mFormat = format; - inputDesc->mChannels = channels; - inputDesc->mAcoustics = acoustics; - inputDesc->mRefCount = 0; - input = mpClientInterface->openInput(&inputDesc->mDevice, - &inputDesc->mSamplingRate, - &inputDesc->mFormat, - &inputDesc->mChannels, - inputDesc->mAcoustics); - - // only accept input with the exact requested set of parameters - if (input == 0 || - (samplingRate != inputDesc->mSamplingRate) || - (format != inputDesc->mFormat) || - (channels != inputDesc->mChannels)) { - LOGV("getInput() failed opening input: samplingRate %d, format %d, channels %d", - samplingRate, format, channels); - if (input != 0) { - mpClientInterface->closeInput(input); - } - delete inputDesc; - return 0; - } - mInputs.add(input, inputDesc); - return input; -} - -status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input) -{ - LOGV("startInput() input %d", input); - ssize_t index = mInputs.indexOfKey(input); - if (index < 0) { - LOGW("startInput() unknow input %d", input); - return BAD_VALUE; - } - AudioInputDescriptor *inputDesc = mInputs.valueAt(index); - -#ifdef AUDIO_POLICY_TEST - if (mTestInput == 0) -#endif //AUDIO_POLICY_TEST - { - // refuse 2 active AudioRecord clients at the same time - if (getActiveInput() != 0) { - LOGW("startInput() input %d failed: other input already started", input); - return INVALID_OPERATION; - } - } - - AudioParameter param = AudioParameter(); - param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice); - - // use Voice Recognition mode or not for this input based on input source - int vr_enabled = inputDesc->mInputSource == AUDIO_SOURCE_VOICE_RECOGNITION ? 1 : 0; - param.addInt(String8("vr_mode"), vr_enabled); - LOGV("AudioPolicyManager::startInput(%d), setting vr_mode to %d", inputDesc->mInputSource, vr_enabled); - - mpClientInterface->setParameters(input, param.toString()); - - inputDesc->mRefCount = 1; - return NO_ERROR; -} - -status_t AudioPolicyManagerBase::stopInput(audio_io_handle_t input) -{ - LOGV("stopInput() input %d", input); - ssize_t index = mInputs.indexOfKey(input); - if (index < 0) { - LOGW("stopInput() unknow input %d", input); - return BAD_VALUE; - } - AudioInputDescriptor *inputDesc = mInputs.valueAt(index); - - if (inputDesc->mRefCount == 0) { - LOGW("stopInput() input %d already stopped", input); - return INVALID_OPERATION; - } else { - AudioParameter param = AudioParameter(); - param.addInt(String8(AudioParameter::keyRouting), 0); - mpClientInterface->setParameters(input, param.toString()); - inputDesc->mRefCount = 0; - return NO_ERROR; - } -} - -void AudioPolicyManagerBase::releaseInput(audio_io_handle_t input) -{ - LOGV("releaseInput() %d", input); - ssize_t index = mInputs.indexOfKey(input); - if (index < 0) { - LOGW("releaseInput() releasing unknown input %d", input); - return; - } - mpClientInterface->closeInput(input); - delete mInputs.valueAt(index); - mInputs.removeItem(input); - LOGV("releaseInput() exit"); -} - -void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type stream, - int indexMin, - int indexMax) -{ - LOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax); - if (indexMin < 0 || indexMin >= indexMax) { - LOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax); - return; - } - mStreams[stream].mIndexMin = indexMin; - mStreams[stream].mIndexMax = indexMax; -} - -status_t AudioPolicyManagerBase::setStreamVolumeIndex(AudioSystem::stream_type stream, int index) -{ - - if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) { - return BAD_VALUE; - } - - // Force max volume if stream cannot be muted - if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax; - - LOGV("setStreamVolumeIndex() stream %d, index %d", stream, index); - mStreams[stream].mIndexCur = index; - - // compute and apply stream volume on all outputs according to connected device - status_t status = NO_ERROR; - for (size_t i = 0; i < mOutputs.size(); i++) { - status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), mOutputs.valueAt(i)->device()); - if (volStatus != NO_ERROR) { - status = volStatus; - } - } - return status; -} - -status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index) -{ - if (index == 0) { - return BAD_VALUE; - } - LOGV("getStreamVolumeIndex() stream %d", stream); - *index = mStreams[stream].mIndexCur; - return NO_ERROR; -} - -status_t AudioPolicyManagerBase::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this); - result.append(buffer); - snprintf(buffer, SIZE, " Hardware Output: %d\n", mHardwareOutput); - result.append(buffer); -#ifdef WITH_A2DP - snprintf(buffer, SIZE, " A2DP Output: %d\n", mA2dpOutput); - result.append(buffer); - snprintf(buffer, SIZE, " Duplicated Output: %d\n", mDuplicatedOutput); - result.append(buffer); - snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string()); - result.append(buffer); -#endif - snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string()); - result.append(buffer); - snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices); - result.append(buffer); - snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices); - result.append(buffer); - snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState); - result.append(buffer); - snprintf(buffer, SIZE, " Ringer mode: %d\n", mRingerMode); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[AudioSystem::FOR_COMMUNICATION]); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AudioSystem::FOR_MEDIA]); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AudioSystem::FOR_RECORD]); - result.append(buffer); - snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AudioSystem::FOR_DOCK]); - result.append(buffer); - write(fd, result.string(), result.size()); - - snprintf(buffer, SIZE, "\nOutputs dump:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mOutputs.size(); i++) { - snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i)); - write(fd, buffer, strlen(buffer)); - mOutputs.valueAt(i)->dump(fd); - } - - snprintf(buffer, SIZE, "\nInputs dump:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mInputs.size(); i++) { - snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i)); - write(fd, buffer, strlen(buffer)); - mInputs.valueAt(i)->dump(fd); - } - - snprintf(buffer, SIZE, "\nStreams dump:\n"); - write(fd, buffer, strlen(buffer)); - snprintf(buffer, SIZE, " Stream Index Min Index Max Index Cur Can be muted\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { - snprintf(buffer, SIZE, " %02d", i); - mStreams[i].dump(buffer + 3, SIZE); - write(fd, buffer, strlen(buffer)); - } - - return NO_ERROR; -} - -// ---------------------------------------------------------------------------- -// AudioPolicyManagerBase -// ---------------------------------------------------------------------------- - -AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface) - : -#ifdef AUDIO_POLICY_TEST - Thread(false), -#endif //AUDIO_POLICY_TEST - mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0), mMusicStopTime(0), mLimitRingtoneVolume(false) -{ - mpClientInterface = clientInterface; - - for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) { - mForceUse[i] = AudioSystem::FORCE_NONE; - } - - // devices available by default are speaker, ear piece and microphone - mAvailableOutputDevices = AudioSystem::DEVICE_OUT_EARPIECE | - AudioSystem::DEVICE_OUT_SPEAKER; - mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC; - -#ifdef WITH_A2DP - mA2dpOutput = 0; - mDuplicatedOutput = 0; - mA2dpDeviceAddress = String8(""); -#endif - mScoDeviceAddress = String8(""); - - // open hardware output - AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(); - outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER; - mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice, - &outputDesc->mSamplingRate, - &outputDesc->mFormat, - &outputDesc->mChannels, - &outputDesc->mLatency, - outputDesc->mFlags); - - if (mHardwareOutput == 0) { - LOGE("Failed to initialize hardware output stream, samplingRate: %d, format %d, channels %d", - outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels); - } else { - addOutput(mHardwareOutput, outputDesc); - setOutputDevice(mHardwareOutput, (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER, true); - } - - updateDeviceForStrategy(); -#ifdef AUDIO_POLICY_TEST - AudioParameter outputCmd = AudioParameter(); - outputCmd.addInt(String8("set_id"), 0); - mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString()); - - mTestDevice = AudioSystem::DEVICE_OUT_SPEAKER; - mTestSamplingRate = 44100; - mTestFormat = AudioSystem::PCM_16_BIT; - mTestChannels = AudioSystem::CHANNEL_OUT_STEREO; - mTestLatencyMs = 0; - mCurOutput = 0; - mDirectOutput = false; - for (int i = 0; i < NUM_TEST_OUTPUTS; i++) { - mTestOutputs[i] = 0; - } - - const size_t SIZE = 256; - char buffer[SIZE]; - snprintf(buffer, SIZE, "AudioPolicyManagerTest"); - run(buffer, ANDROID_PRIORITY_AUDIO); -#endif //AUDIO_POLICY_TEST -} - -AudioPolicyManagerBase::~AudioPolicyManagerBase() -{ -#ifdef AUDIO_POLICY_TEST - exit(); -#endif //AUDIO_POLICY_TEST - for (size_t i = 0; i < mOutputs.size(); i++) { - mpClientInterface->closeOutput(mOutputs.keyAt(i)); - delete mOutputs.valueAt(i); - } - mOutputs.clear(); - for (size_t i = 0; i < mInputs.size(); i++) { - mpClientInterface->closeInput(mInputs.keyAt(i)); - delete mInputs.valueAt(i); - } - mInputs.clear(); -} - -#ifdef AUDIO_POLICY_TEST -bool AudioPolicyManagerBase::threadLoop() -{ - LOGV("entering threadLoop()"); - while (!exitPending()) - { - String8 command; - int valueInt; - String8 value; - - Mutex::Autolock _l(mLock); - mWaitWorkCV.waitRelative(mLock, milliseconds(50)); - - command = mpClientInterface->getParameters(0, String8("test_cmd_policy")); - AudioParameter param = AudioParameter(command); - - if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR && - valueInt != 0) { - LOGV("Test command %s received", command.string()); - String8 target; - if (param.get(String8("target"), target) != NO_ERROR) { - target = "Manager"; - } - if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) { - param.remove(String8("test_cmd_policy_output")); - mCurOutput = valueInt; - } - if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) { - param.remove(String8("test_cmd_policy_direct")); - if (value == "false") { - mDirectOutput = false; - } else if (value == "true") { - mDirectOutput = true; - } - } - if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) { - param.remove(String8("test_cmd_policy_input")); - mTestInput = valueInt; - } - - if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) { - param.remove(String8("test_cmd_policy_format")); - int format = AudioSystem::INVALID_FORMAT; - if (value == "PCM 16 bits") { - format = AudioSystem::PCM_16_BIT; - } else if (value == "PCM 8 bits") { - format = AudioSystem::PCM_8_BIT; - } else if (value == "Compressed MP3") { - format = AudioSystem::MP3; - } - if (format != AudioSystem::INVALID_FORMAT) { - if (target == "Manager") { - mTestFormat = format; - } else if (mTestOutputs[mCurOutput] != 0) { - AudioParameter outputParam = AudioParameter(); - outputParam.addInt(String8("format"), format); - mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); - } - } - } - if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) { - param.remove(String8("test_cmd_policy_channels")); - int channels = 0; - - if (value == "Channels Stereo") { - channels = AudioSystem::CHANNEL_OUT_STEREO; - } else if (value == "Channels Mono") { - channels = AudioSystem::CHANNEL_OUT_MONO; - } - if (channels != 0) { - if (target == "Manager") { - mTestChannels = channels; - } else if (mTestOutputs[mCurOutput] != 0) { - AudioParameter outputParam = AudioParameter(); - outputParam.addInt(String8("channels"), channels); - mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); - } - } - } - if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) { - param.remove(String8("test_cmd_policy_sampleRate")); - if (valueInt >= 0 && valueInt <= 96000) { - int samplingRate = valueInt; - if (target == "Manager") { - mTestSamplingRate = samplingRate; - } else if (mTestOutputs[mCurOutput] != 0) { - AudioParameter outputParam = AudioParameter(); - outputParam.addInt(String8("sampling_rate"), samplingRate); - mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); - } - } - } - - if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) { - param.remove(String8("test_cmd_policy_reopen")); - - mpClientInterface->closeOutput(mHardwareOutput); - delete mOutputs.valueFor(mHardwareOutput); - mOutputs.removeItem(mHardwareOutput); - - AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(); - outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER; - mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice, - &outputDesc->mSamplingRate, - &outputDesc->mFormat, - &outputDesc->mChannels, - &outputDesc->mLatency, - outputDesc->mFlags); - if (mHardwareOutput == 0) { - LOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d", - outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels); - } else { - AudioParameter outputCmd = AudioParameter(); - outputCmd.addInt(String8("set_id"), 0); - mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString()); - addOutput(mHardwareOutput, outputDesc); - } - } - - - mpClientInterface->setParameters(0, String8("test_cmd_policy=")); - } - } - return false; -} - -void AudioPolicyManagerBase::exit() -{ - { - AutoMutex _l(mLock); - requestExit(); - mWaitWorkCV.signal(); - } - requestExitAndWait(); -} - -int AudioPolicyManagerBase::testOutputIndex(audio_io_handle_t output) -{ - for (int i = 0; i < NUM_TEST_OUTPUTS; i++) { - if (output == mTestOutputs[i]) return i; - } - return 0; -} -#endif //AUDIO_POLICY_TEST - -// --- - -void AudioPolicyManagerBase::addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc) -{ - outputDesc->mId = id; - mOutputs.add(id, outputDesc); -} - - -#ifdef WITH_A2DP -status_t AudioPolicyManagerBase::handleA2dpConnection(AudioSystem::audio_devices device, - const char *device_address) -{ - // when an A2DP device is connected, open an A2DP and a duplicated output - LOGV("opening A2DP output for device %s", device_address); - AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(); - outputDesc->mDevice = device; - mA2dpOutput = mpClientInterface->openOutput(&outputDesc->mDevice, - &outputDesc->mSamplingRate, - &outputDesc->mFormat, - &outputDesc->mChannels, - &outputDesc->mLatency, - outputDesc->mFlags); - if (mA2dpOutput) { - // add A2DP output descriptor - addOutput(mA2dpOutput, outputDesc); - // set initial stream volume for A2DP device - applyStreamVolumes(mA2dpOutput, device); - if (a2dpUsedForSonification()) { - mDuplicatedOutput = mpClientInterface->openDuplicateOutput(mA2dpOutput, mHardwareOutput); - } - if (mDuplicatedOutput != 0 || - !a2dpUsedForSonification()) { - // If both A2DP and duplicated outputs are open, send device address to A2DP hardware - // interface - AudioParameter param; - param.add(String8("a2dp_sink_address"), String8(device_address)); - mpClientInterface->setParameters(mA2dpOutput, param.toString()); - mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN); - - if (a2dpUsedForSonification()) { - // add duplicated output descriptor - AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(); - dupOutputDesc->mOutput1 = mOutputs.valueFor(mHardwareOutput); - dupOutputDesc->mOutput2 = mOutputs.valueFor(mA2dpOutput); - dupOutputDesc->mSamplingRate = outputDesc->mSamplingRate; - dupOutputDesc->mFormat = outputDesc->mFormat; - dupOutputDesc->mChannels = outputDesc->mChannels; - dupOutputDesc->mLatency = outputDesc->mLatency; - addOutput(mDuplicatedOutput, dupOutputDesc); - applyStreamVolumes(mDuplicatedOutput, device); - } - } else { - LOGW("getOutput() could not open duplicated output for %d and %d", - mHardwareOutput, mA2dpOutput); - mpClientInterface->closeOutput(mA2dpOutput); - mOutputs.removeItem(mA2dpOutput); - mA2dpOutput = 0; - delete outputDesc; - return NO_INIT; - } - } else { - LOGW("setDeviceConnectionState() could not open A2DP output for device %x", device); - delete outputDesc; - return NO_INIT; - } - AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput); - - if (mScoDeviceAddress != "") { - // It is normal to suspend twice if we are both in call, - // and have the hardware audio output routed to BT SCO - if (mPhoneState != AudioSystem::MODE_NORMAL) { - mpClientInterface->suspendOutput(mA2dpOutput); - } - if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)hwOutputDesc->device())) { - mpClientInterface->suspendOutput(mA2dpOutput); - } - } - - if (!a2dpUsedForSonification()) { - // mute music on A2DP output if a notification or ringtone is playing - uint32_t refCount = hwOutputDesc->strategyRefCount(STRATEGY_SONIFICATION); - for (uint32_t i = 0; i < refCount; i++) { - setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput); - } - } - return NO_ERROR; -} - -status_t AudioPolicyManagerBase::handleA2dpDisconnection(AudioSystem::audio_devices device, - const char *device_address) -{ - if (mA2dpOutput == 0) { - LOGW("setDeviceConnectionState() disconnecting A2DP and no A2DP output!"); - return INVALID_OPERATION; - } - - if (mA2dpDeviceAddress != device_address) { - LOGW("setDeviceConnectionState() disconnecting unknow A2DP sink address %s", device_address); - return INVALID_OPERATION; - } - - // mute media strategy to avoid outputting sound on hardware output while music stream - // is switched from A2DP output and before music is paused by music application - setStrategyMute(STRATEGY_MEDIA, true, mHardwareOutput); - setStrategyMute(STRATEGY_MEDIA, false, mHardwareOutput, MUTE_TIME_MS); - - if (!a2dpUsedForSonification()) { - // unmute music on A2DP output if a notification or ringtone is playing - uint32_t refCount = mOutputs.valueFor(mHardwareOutput)->strategyRefCount(STRATEGY_SONIFICATION); - for (uint32_t i = 0; i < refCount; i++) { - setStrategyMute(STRATEGY_MEDIA, false, mA2dpOutput); - } - } - mA2dpDeviceAddress = ""; - return NO_ERROR; -} - -void AudioPolicyManagerBase::closeA2dpOutputs() -{ - LOGV("setDeviceConnectionState() closing A2DP and duplicated output!"); - - if (mDuplicatedOutput != 0) { - AudioOutputDescriptor *dupOutputDesc = mOutputs.valueFor(mDuplicatedOutput); - AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput); - // As all active tracks on duplicated output will be deleted, - // and as they were also referenced on hardware output, the reference - // count for their stream type must be adjusted accordingly on - // hardware output. - for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) { - int refCount = dupOutputDesc->mRefCount[i]; - hwOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount); - } - - mpClientInterface->closeOutput(mDuplicatedOutput); - delete mOutputs.valueFor(mDuplicatedOutput); - mOutputs.removeItem(mDuplicatedOutput); - mDuplicatedOutput = 0; - } - if (mA2dpOutput != 0) { - AudioParameter param; - param.add(String8("closing"), String8("true")); - mpClientInterface->setParameters(mA2dpOutput, param.toString()); - mpClientInterface->closeOutput(mA2dpOutput); - delete mOutputs.valueFor(mA2dpOutput); - mOutputs.removeItem(mA2dpOutput); - mA2dpOutput = 0; - } -} - -void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy, uint32_t &newDevice) -{ - uint32_t prevDevice = getDeviceForStrategy(strategy); - uint32_t curDevice = getDeviceForStrategy(strategy, false); - bool a2dpWasUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(prevDevice & ~AudioSystem::DEVICE_OUT_SPEAKER)); - bool a2dpIsUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(curDevice & ~AudioSystem::DEVICE_OUT_SPEAKER)); - AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput); - AudioOutputDescriptor *a2dpOutputDesc; - - if (a2dpWasUsed && !a2dpIsUsed) { - bool dupUsed = a2dpUsedForSonification() && a2dpWasUsed && (AudioSystem::popCount(prevDevice) == 2); - - if (dupUsed) { - LOGV("checkOutputForStrategy() moving strategy %d to duplicated", strategy); - a2dpOutputDesc = mOutputs.valueFor(mDuplicatedOutput); - } else { - LOGV("checkOutputForStrategy() moving strategy %d to a2dp", strategy); - a2dpOutputDesc = mOutputs.valueFor(mA2dpOutput); - } - - for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) { - if (getStrategy((AudioSystem::stream_type)i) == strategy) { - mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mHardwareOutput); - } - } - // do not change newDevice if it was already set before this call by a previous call to - // getNewDevice() or checkOutputForStrategy() for a strategy with higher priority - if (newDevice == 0 && hwOutputDesc->isUsedByStrategy(strategy)) { - newDevice = getDeviceForStrategy(strategy, false); - } - } - if (a2dpIsUsed && !a2dpWasUsed) { - bool dupUsed = a2dpUsedForSonification() && a2dpIsUsed && (AudioSystem::popCount(curDevice) == 2); - audio_io_handle_t a2dpOutput; - - if (dupUsed) { - LOGV("checkOutputForStrategy() moving strategy %d from duplicated", strategy); - a2dpOutputDesc = mOutputs.valueFor(mDuplicatedOutput); - a2dpOutput = mDuplicatedOutput; - } else { - LOGV("checkOutputForStrategy() moving strategy %d from a2dp", strategy); - a2dpOutputDesc = mOutputs.valueFor(mA2dpOutput); - a2dpOutput = mA2dpOutput; - } - - for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) { - if (getStrategy((AudioSystem::stream_type)i) == strategy) { - mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, a2dpOutput); - } - } - } -} - -void AudioPolicyManagerBase::checkOutputForAllStrategies(uint32_t &newDevice) -{ - // Check strategies in order of priority so that once newDevice is set - // for a given strategy it is not modified by subsequent calls to - // checkOutputForStrategy() - checkOutputForStrategy(STRATEGY_PHONE, newDevice); - checkOutputForStrategy(STRATEGY_SONIFICATION, newDevice); - checkOutputForStrategy(STRATEGY_MEDIA, newDevice); - checkOutputForStrategy(STRATEGY_DTMF, newDevice); -} - -#endif - -uint32_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache) -{ - uint32_t device = 0; - - AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); - // check the following by order of priority to request a routing change if necessary: - // 1: we are in call or the strategy phone is active on the hardware output: - // use device for strategy phone - // 2: the strategy sonification is active on the hardware output: - // use device for strategy sonification - // 3: the strategy media is active on the hardware output: - // use device for strategy media - // 4: the strategy DTMF is active on the hardware output: - // use device for strategy DTMF - if (mPhoneState == AudioSystem::MODE_IN_CALL || - outputDesc->isUsedByStrategy(STRATEGY_PHONE)) { - device = getDeviceForStrategy(STRATEGY_PHONE, fromCache); - } else if (outputDesc->isUsedByStrategy(STRATEGY_SONIFICATION)) { - device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); - } else if (outputDesc->isUsedByStrategy(STRATEGY_MEDIA)) { - device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache); - } else if (outputDesc->isUsedByStrategy(STRATEGY_DTMF)) { - device = getDeviceForStrategy(STRATEGY_DTMF, fromCache); - } - - LOGV("getNewDevice() selected device %x", device); - return device; -} - -AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy(AudioSystem::stream_type stream) -{ - // stream to strategy mapping - switch (stream) { - case AudioSystem::VOICE_CALL: - case AudioSystem::BLUETOOTH_SCO: - return STRATEGY_PHONE; - case AudioSystem::RING: - case AudioSystem::NOTIFICATION: - case AudioSystem::ALARM: - case AudioSystem::ENFORCED_AUDIBLE: - return STRATEGY_SONIFICATION; - case AudioSystem::DTMF: - return STRATEGY_DTMF; - default: - LOGE("unknown stream type"); - case AudioSystem::SYSTEM: - // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs - // while key clicks are played produces a poor result - case AudioSystem::TTS: - case AudioSystem::MUSIC: - return STRATEGY_MEDIA; - } -} - -uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy, bool fromCache) -{ - uint32_t device = 0; - - if (fromCache) { - LOGV("getDeviceForStrategy() from cache strategy %d, device %x", strategy, mDeviceForStrategy[strategy]); - return mDeviceForStrategy[strategy]; - } - - switch (strategy) { - case STRATEGY_DTMF: - if (mPhoneState != AudioSystem::MODE_IN_CALL) { - // when off call, DTMF strategy follows the same rules as MEDIA strategy - device = getDeviceForStrategy(STRATEGY_MEDIA, false); - break; - } - // when in call, DTMF and PHONE strategies follow the same rules - // FALL THROUGH - - case STRATEGY_PHONE: - // for phone strategy, we first consider the forced use and then the available devices by order - // of priority - switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) { - case AudioSystem::FORCE_BT_SCO: - if (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) { - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT; - if (device) break; - } - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO; - if (device) break; - // if SCO device is requested but no SCO device is available, fall back to default case - // FALL THROUGH - - default: // FORCE_NONE - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET; - if (device) break; -#ifdef WITH_A2DP - // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP - if (mPhoneState != AudioSystem::MODE_IN_CALL) { - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; - if (device) break; - } -#endif - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_EARPIECE; - if (device == 0) { - LOGE("getDeviceForStrategy() earpiece device not found"); - } - break; - - case AudioSystem::FORCE_SPEAKER: - if (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) { - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT; - if (device) break; - } -#ifdef WITH_A2DP - // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to - // A2DP speaker when forcing to speaker output - if (mPhoneState != AudioSystem::MODE_IN_CALL) { - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; - if (device) break; - } -#endif - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER; - if (device == 0) { - LOGE("getDeviceForStrategy() speaker device not found"); - } - break; - } - break; - - case STRATEGY_SONIFICATION: - - // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by - // handleIncallSonification(). - if (mPhoneState == AudioSystem::MODE_IN_CALL) { - device = getDeviceForStrategy(STRATEGY_PHONE, false); - break; - } - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER; - if (device == 0) { - LOGE("getDeviceForStrategy() speaker device not found"); - } - // The second device used for sonification is the same as the device used by media strategy - // FALL THROUGH - - case STRATEGY_MEDIA: { - uint32_t device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL; - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE; - } - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET; - } -#ifdef WITH_A2DP - if (mA2dpOutput != 0) { - if (strategy == STRATEGY_SONIFICATION && !a2dpUsedForSonification()) { - break; - } - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP; - } - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; - } - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; - } - } -#endif - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER; - } - - // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION, 0 otherwise - device |= device2; - if (device == 0) { - LOGE("getDeviceForStrategy() speaker device not found"); - } - } break; - - default: - LOGW("getDeviceForStrategy() unknown strategy: %d", strategy); - break; - } - - LOGV("getDeviceForStrategy() strategy %d, device %x", strategy, device); - return device; -} - -void AudioPolicyManagerBase::updateDeviceForStrategy() -{ - for (int i = 0; i < NUM_STRATEGIES; i++) { - mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false); - } -} - -void AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output, uint32_t device, bool force, int delayMs) -{ - LOGV("setOutputDevice() output %d device %x delayMs %d", output, device, delayMs); - AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); - - - if (outputDesc->isDuplicated()) { - setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs); - setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs); - return; - } -#ifdef WITH_A2DP - // filter devices according to output selected - if (output == mA2dpOutput) { - device &= AudioSystem::DEVICE_OUT_ALL_A2DP; - } else { - device &= ~AudioSystem::DEVICE_OUT_ALL_A2DP; - } -#endif - - uint32_t prevDevice = (uint32_t)outputDesc->device(); - // Do not change the routing if: - // - the requestede device is 0 - // - the requested device is the same as current device and force is not specified. - // Doing this check here allows the caller to call setOutputDevice() without conditions - if ((device == 0 || device == prevDevice) && !force) { - LOGV("setOutputDevice() setting same device %x or null device for output %d", device, output); - return; - } - - outputDesc->mDevice = device; - // mute media streams if both speaker and headset are selected - if (output == mHardwareOutput && AudioSystem::popCount(device) == 2) { - setStrategyMute(STRATEGY_MEDIA, true, output); - // wait for the PCM output buffers to empty before proceeding with the rest of the command - usleep(outputDesc->mLatency*2*1000); - } -#ifdef WITH_A2DP - // suspend A2DP output if SCO device is selected - if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)device)) { - if (mA2dpOutput != 0) { - mpClientInterface->suspendOutput(mA2dpOutput); - } - } -#endif - // do the routing - AudioParameter param = AudioParameter(); - param.addInt(String8(AudioParameter::keyRouting), (int)device); - mpClientInterface->setParameters(mHardwareOutput, param.toString(), delayMs); - // update stream volumes according to new device - applyStreamVolumes(output, device, delayMs); - -#ifdef WITH_A2DP - // if disconnecting SCO device, restore A2DP output - if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)prevDevice)) { - if (mA2dpOutput != 0) { - LOGV("restore A2DP output"); - mpClientInterface->restoreOutput(mA2dpOutput); - } - } -#endif - // if changing from a combined headset + speaker route, unmute media streams - if (output == mHardwareOutput && AudioSystem::popCount(prevDevice) == 2) { - setStrategyMute(STRATEGY_MEDIA, false, output, delayMs); - } -} - -uint32_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource) -{ - uint32_t device; - - switch(inputSource) { - case AUDIO_SOURCE_DEFAULT: - case AUDIO_SOURCE_MIC: - case AUDIO_SOURCE_VOICE_RECOGNITION: - if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO && - mAvailableInputDevices & AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET) { - device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET; - } else if (mAvailableInputDevices & AudioSystem::DEVICE_IN_WIRED_HEADSET) { - device = AudioSystem::DEVICE_IN_WIRED_HEADSET; - } else { - device = AudioSystem::DEVICE_IN_BUILTIN_MIC; - } - break; - case AUDIO_SOURCE_CAMCORDER: - if (hasBackMicrophone()) { - device = AudioSystem::DEVICE_IN_BACK_MIC; - } else { - device = AudioSystem::DEVICE_IN_BUILTIN_MIC; - } - break; - case AUDIO_SOURCE_VOICE_UPLINK: - case AUDIO_SOURCE_VOICE_DOWNLINK: - case AUDIO_SOURCE_VOICE_CALL: - device = AudioSystem::DEVICE_IN_VOICE_CALL; - break; - default: - LOGW("getInput() invalid input source %d", inputSource); - device = 0; - break; - } - LOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device); - return device; -} - -audio_io_handle_t AudioPolicyManagerBase::getActiveInput() -{ - for (size_t i = 0; i < mInputs.size(); i++) { - if (mInputs.valueAt(i)->mRefCount > 0) { - return mInputs.keyAt(i); - } - } - return 0; -} - -float AudioPolicyManagerBase::computeVolume(int stream, int index, audio_io_handle_t output, uint32_t device) -{ - float volume = 1.0; - AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); - StreamDescriptor &streamDesc = mStreams[stream]; - - if (device == 0) { - device = outputDesc->device(); - } - - int volInt = (100 * (index - streamDesc.mIndexMin)) / (streamDesc.mIndexMax - streamDesc.mIndexMin); - volume = AudioSystem::linearToLog(volInt); - - // if a headset is connected, apply the following rules to ring tones and notifications - // to avoid sound level bursts in user's ears: - // - always attenuate ring tones and notifications volume by 6dB - // - if music is playing, always limit the volume to current music volume, - // with a minimum threshold at -36dB so that notification is always perceived. - if ((device & - (AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP | - AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | - AudioSystem::DEVICE_OUT_WIRED_HEADSET | - AudioSystem::DEVICE_OUT_WIRED_HEADPHONE)) && - (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) && - streamDesc.mCanBeMuted) { - volume *= SONIFICATION_HEADSET_VOLUME_FACTOR; - // when the phone is ringing we must consider that music could have been paused just before - // by the music application and behave as if music was active if the last music track was - // just stopped - if (outputDesc->mRefCount[AudioSystem::MUSIC] || mLimitRingtoneVolume) { - float musicVol = computeVolume(AudioSystem::MUSIC, mStreams[AudioSystem::MUSIC].mIndexCur, output, device); - float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? musicVol : SONIFICATION_HEADSET_VOLUME_MIN; - if (volume > minVol) { - volume = minVol; - LOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol); - } - } - } - - return volume; -} - -status_t AudioPolicyManagerBase::checkAndSetVolume(int stream, int index, audio_io_handle_t output, uint32_t device, int delayMs, bool force) -{ - - // do not change actual stream volume if the stream is muted - if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) { - LOGV("checkAndSetVolume() stream %d muted count %d", stream, mOutputs.valueFor(output)->mMuteCount[stream]); - return NO_ERROR; - } - - // do not change in call volume if bluetooth is connected and vice versa - if ((stream == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) || - (stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) { - LOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm", - stream, mForceUse[AudioSystem::FOR_COMMUNICATION]); - return INVALID_OPERATION; - } - - float volume = computeVolume(stream, index, output, device); - // do not set volume if the float value did not change - if (volume != mOutputs.valueFor(output)->mCurVolume[stream] || force) { - mOutputs.valueFor(output)->mCurVolume[stream] = volume; - LOGV("setStreamVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs); - if (stream == AudioSystem::VOICE_CALL || - stream == AudioSystem::DTMF || - stream == AudioSystem::BLUETOOTH_SCO) { - float voiceVolume = -1.0; - // offset value to reflect actual hardware volume that never reaches 0 - // 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java) - volume = 0.01 + 0.99 * volume; - if (stream == AudioSystem::VOICE_CALL) { - voiceVolume = (float)index/(float)mStreams[stream].mIndexMax; - } else if (stream == AudioSystem::BLUETOOTH_SCO) { - voiceVolume = 1.0; - } - if (voiceVolume >= 0 && output == mHardwareOutput) { - mpClientInterface->setVoiceVolume(voiceVolume, delayMs); - } - } - mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs); - } - - return NO_ERROR; -} - -void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output, uint32_t device, int delayMs) -{ - LOGV("applyStreamVolumes() for output %d and device %x", output, device); - - for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { - checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, device, delayMs); - } -} - -void AudioPolicyManagerBase::setStrategyMute(routing_strategy strategy, bool on, audio_io_handle_t output, int delayMs) -{ - LOGV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output); - for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { - if (getStrategy((AudioSystem::stream_type)stream) == strategy) { - setStreamMute(stream, on, output, delayMs); - } - } -} - -void AudioPolicyManagerBase::setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs) -{ - StreamDescriptor &streamDesc = mStreams[stream]; - AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); - - LOGV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d", stream, on, output, outputDesc->mMuteCount[stream]); - - if (on) { - if (outputDesc->mMuteCount[stream] == 0) { - if (streamDesc.mCanBeMuted) { - checkAndSetVolume(stream, 0, output, outputDesc->device(), delayMs); - } - } - // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored - outputDesc->mMuteCount[stream]++; - } else { - if (outputDesc->mMuteCount[stream] == 0) { - LOGW("setStreamMute() unmuting non muted stream!"); - return; - } - if (--outputDesc->mMuteCount[stream] == 0) { - checkAndSetVolume(stream, streamDesc.mIndexCur, output, outputDesc->device(), delayMs); - } - } -} - -void AudioPolicyManagerBase::handleIncallSonification(int stream, bool starting, bool stateChange) -{ - // if the stream pertains to sonification strategy and we are in call we must - // mute the stream if it is low visibility. If it is high visibility, we must play a tone - // in the device used for phone strategy and play the tone if the selected device does not - // interfere with the device used for phone strategy - // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as - // many times as there are active tracks on the output - - if (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) { - AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mHardwareOutput); - LOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d", - stream, starting, outputDesc->mDevice, stateChange); - if (outputDesc->mRefCount[stream]) { - int muteCount = 1; - if (stateChange) { - muteCount = outputDesc->mRefCount[stream]; - } - if (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) { - LOGV("handleIncallSonification() low visibility, muteCount %d", muteCount); - for (int i = 0; i < muteCount; i++) { - setStreamMute(stream, starting, mHardwareOutput); - } - } else { - LOGV("handleIncallSonification() high visibility"); - if (outputDesc->device() & getDeviceForStrategy(STRATEGY_PHONE)) { - LOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount); - for (int i = 0; i < muteCount; i++) { - setStreamMute(stream, starting, mHardwareOutput); - } - } - if (starting) { - mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL); - } else { - mpClientInterface->stopTone(); - } - } - } - } -} - -bool AudioPolicyManagerBase::needsDirectOuput(AudioSystem::stream_type stream, - uint32_t samplingRate, - uint32_t format, - uint32_t channels, - AudioSystem::output_flags flags, - uint32_t device) -{ - return ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || - (format !=0 && !AudioSystem::isLinearPCM(format))); -} - -// --- AudioOutputDescriptor class implementation - -AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor() - : mId(0), mSamplingRate(0), mFormat(0), mChannels(0), mLatency(0), - mFlags((AudioSystem::output_flags)0), mDevice(0), mOutput1(0), mOutput2(0) -{ - // clear usage count for all stream types - for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { - mRefCount[i] = 0; - mCurVolume[i] = -1.0; - mMuteCount[i] = 0; - } -} - -uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::device() -{ - uint32_t device = 0; - if (isDuplicated()) { - device = mOutput1->mDevice | mOutput2->mDevice; - } else { - device = mDevice; - } - return device; -} - -void AudioPolicyManagerBase::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type stream, int delta) -{ - // forward usage count change to attached outputs - if (isDuplicated()) { - mOutput1->changeRefCount(stream, delta); - mOutput2->changeRefCount(stream, delta); - } - if ((delta + (int)mRefCount[stream]) < 0) { - LOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]); - mRefCount[stream] = 0; - return; - } - mRefCount[stream] += delta; - LOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]); -} - -uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::refCount() -{ - uint32_t refcount = 0; - for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) { - refcount += mRefCount[i]; - } - return refcount; -} - -uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::strategyRefCount(routing_strategy strategy) -{ - uint32_t refCount = 0; - for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) { - if (getStrategy((AudioSystem::stream_type)i) == strategy) { - refCount += mRefCount[i]; - } - } - return refCount; -} - - -status_t AudioPolicyManagerBase::AudioOutputDescriptor::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate); - result.append(buffer); - snprintf(buffer, SIZE, " Format: %d\n", mFormat); - result.append(buffer); - snprintf(buffer, SIZE, " Channels: %08x\n", mChannels); - result.append(buffer); - snprintf(buffer, SIZE, " Latency: %d\n", mLatency); - result.append(buffer); - snprintf(buffer, SIZE, " Flags %08x\n", mFlags); - result.append(buffer); - snprintf(buffer, SIZE, " Devices %08x\n", device()); - result.append(buffer); - snprintf(buffer, SIZE, " Stream volume refCount muteCount\n"); - result.append(buffer); - for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { - snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n", i, mCurVolume[i], mRefCount[i], mMuteCount[i]); - result.append(buffer); - } - write(fd, result.string(), result.size()); - - return NO_ERROR; -} - -// --- AudioInputDescriptor class implementation - -AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor() - : mSamplingRate(0), mFormat(0), mChannels(0), - mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0) -{ -} - -status_t AudioPolicyManagerBase::AudioInputDescriptor::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate); - result.append(buffer); - snprintf(buffer, SIZE, " Format: %d\n", mFormat); - result.append(buffer); - snprintf(buffer, SIZE, " Channels: %08x\n", mChannels); - result.append(buffer); - snprintf(buffer, SIZE, " Acoustics %08x\n", mAcoustics); - result.append(buffer); - snprintf(buffer, SIZE, " Devices %08x\n", mDevice); - result.append(buffer); - snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount); - result.append(buffer); - write(fd, result.string(), result.size()); - - return NO_ERROR; -} - -// --- StreamDescriptor class implementation - -void AudioPolicyManagerBase::StreamDescriptor::dump(char* buffer, size_t size) -{ - snprintf(buffer, size, " %02d %02d %02d %d\n", - mIndexMin, - mIndexMax, - mIndexCur, - mCanBeMuted); -} - - -}; // namespace android diff --git a/libs/audioflinger/AudioPolicyService.cpp b/libs/audioflinger/AudioPolicyService.cpp deleted file mode 100644 index bb3905c..0000000 --- a/libs/audioflinger/AudioPolicyService.cpp +++ /dev/null @@ -1,924 +0,0 @@ -/* - * Copyright (C) 2009 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#define LOG_TAG "AudioPolicyService" -//#define LOG_NDEBUG 0 - -#undef __STRICT_ANSI__ -#define __STDINT_LIMITS -#define __STDC_LIMIT_MACROS -#include <stdint.h> - -#include <sys/time.h> -#include <binder/IServiceManager.h> -#include <utils/Log.h> -#include <cutils/properties.h> -#include <binder/IPCThreadState.h> -#include <utils/String16.h> -#include <utils/threads.h> -#include "AudioPolicyService.h" -#include <hardware_legacy/AudioPolicyManagerBase.h> -#include <cutils/properties.h> -#include <dlfcn.h> -#include <hardware_legacy/power.h> - -// ---------------------------------------------------------------------------- -// the sim build doesn't have gettid - -#ifndef HAVE_GETTID -# define gettid getpid -#endif - -namespace android { - - -static const char *kDeadlockedString = "AudioPolicyService may be deadlocked\n"; -static const char *kCmdDeadlockedString = "AudioPolicyService command thread may be deadlocked\n"; - -static const int kDumpLockRetries = 50; -static const int kDumpLockSleep = 20000; - -static bool checkPermission() { -#ifndef HAVE_ANDROID_OS - return true; -#endif - if (getpid() == IPCThreadState::self()->getCallingPid()) return true; - bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); - if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); - return ok; -} - -// ---------------------------------------------------------------------------- - -AudioPolicyService::AudioPolicyService() - : BnAudioPolicyService() , mpPolicyManager(NULL) -{ - char value[PROPERTY_VALUE_MAX]; - - // start tone playback thread - mTonePlaybackThread = new AudioCommandThread(String8("")); - // start audio commands thread - mAudioCommandThread = new AudioCommandThread(String8("ApmCommandThread")); - -#if (defined GENERIC_AUDIO) || (defined AUDIO_POLICY_TEST) - mpPolicyManager = new AudioPolicyManagerBase(this); - LOGV("build for GENERIC_AUDIO - using generic audio policy"); -#else - // if running in emulation - use the emulator driver - if (property_get("ro.kernel.qemu", value, 0)) { - LOGV("Running in emulation - using generic audio policy"); - mpPolicyManager = new AudioPolicyManagerBase(this); - } - else { - LOGV("Using hardware specific audio policy"); - mpPolicyManager = createAudioPolicyManager(this); - } -#endif - - // load properties - property_get("ro.camera.sound.forced", value, "0"); - mpPolicyManager->setSystemProperty("ro.camera.sound.forced", value); -} - -AudioPolicyService::~AudioPolicyService() -{ - mTonePlaybackThread->exit(); - mTonePlaybackThread.clear(); - mAudioCommandThread->exit(); - mAudioCommandThread.clear(); - - if (mpPolicyManager) { - delete mpPolicyManager; - } -} - - -status_t AudioPolicyService::setDeviceConnectionState(AudioSystem::audio_devices device, - AudioSystem::device_connection_state state, - const char *device_address) -{ - if (mpPolicyManager == NULL) { - return NO_INIT; - } - if (!checkPermission()) { - return PERMISSION_DENIED; - } - if (!AudioSystem::isOutputDevice(device) && !AudioSystem::isInputDevice(device)) { - return BAD_VALUE; - } - if (state != AudioSystem::DEVICE_STATE_AVAILABLE && state != AudioSystem::DEVICE_STATE_UNAVAILABLE) { - return BAD_VALUE; - } - - LOGV("setDeviceConnectionState() tid %d", gettid()); - Mutex::Autolock _l(mLock); - return mpPolicyManager->setDeviceConnectionState(device, state, device_address); -} - -AudioSystem::device_connection_state AudioPolicyService::getDeviceConnectionState(AudioSystem::audio_devices device, - const char *device_address) -{ - if (mpPolicyManager == NULL) { - return AudioSystem::DEVICE_STATE_UNAVAILABLE; - } - if (!checkPermission()) { - return AudioSystem::DEVICE_STATE_UNAVAILABLE; - } - return mpPolicyManager->getDeviceConnectionState(device, device_address); -} - -status_t AudioPolicyService::setPhoneState(int state) -{ - if (mpPolicyManager == NULL) { - return NO_INIT; - } - if (!checkPermission()) { - return PERMISSION_DENIED; - } - if (state < 0 || state >= AudioSystem::NUM_MODES) { - return BAD_VALUE; - } - - LOGV("setPhoneState() tid %d", gettid()); - - // TODO: check if it is more appropriate to do it in platform specific policy manager - AudioSystem::setMode(state); - - Mutex::Autolock _l(mLock); - mpPolicyManager->setPhoneState(state); - return NO_ERROR; -} - -status_t AudioPolicyService::setRingerMode(uint32_t mode, uint32_t mask) -{ - if (mpPolicyManager == NULL) { - return NO_INIT; - } - if (!checkPermission()) { - return PERMISSION_DENIED; - } - - mpPolicyManager->setRingerMode(mode, mask); - return NO_ERROR; -} - -status_t AudioPolicyService::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) -{ - if (mpPolicyManager == NULL) { - return NO_INIT; - } - if (!checkPermission()) { - return PERMISSION_DENIED; - } - if (usage < 0 || usage >= AudioSystem::NUM_FORCE_USE) { - return BAD_VALUE; - } - if (config < 0 || config >= AudioSystem::NUM_FORCE_CONFIG) { - return BAD_VALUE; - } - LOGV("setForceUse() tid %d", gettid()); - Mutex::Autolock _l(mLock); - mpPolicyManager->setForceUse(usage, config); - return NO_ERROR; -} - -AudioSystem::forced_config AudioPolicyService::getForceUse(AudioSystem::force_use usage) -{ - if (mpPolicyManager == NULL) { - return AudioSystem::FORCE_NONE; - } - if (!checkPermission()) { - return AudioSystem::FORCE_NONE; - } - if (usage < 0 || usage >= AudioSystem::NUM_FORCE_USE) { - return AudioSystem::FORCE_NONE; - } - return mpPolicyManager->getForceUse(usage); -} - -audio_io_handle_t AudioPolicyService::getOutput(AudioSystem::stream_type stream, - uint32_t samplingRate, - uint32_t format, - uint32_t channels, - AudioSystem::output_flags flags) -{ - if (mpPolicyManager == NULL) { - return 0; - } - LOGV("getOutput() tid %d", gettid()); - Mutex::Autolock _l(mLock); - return mpPolicyManager->getOutput(stream, samplingRate, format, channels, flags); -} - -status_t AudioPolicyService::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream) -{ - if (mpPolicyManager == NULL) { - return NO_INIT; - } - LOGV("startOutput() tid %d", gettid()); - Mutex::Autolock _l(mLock); - return mpPolicyManager->startOutput(output, stream); -} - -status_t AudioPolicyService::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream) -{ - if (mpPolicyManager == NULL) { - return NO_INIT; - } - LOGV("stopOutput() tid %d", gettid()); - Mutex::Autolock _l(mLock); - return mpPolicyManager->stopOutput(output, stream); -} - -void AudioPolicyService::releaseOutput(audio_io_handle_t output) -{ - if (mpPolicyManager == NULL) { - return; - } - LOGV("releaseOutput() tid %d", gettid()); - Mutex::Autolock _l(mLock); - mpPolicyManager->releaseOutput(output); -} - -audio_io_handle_t AudioPolicyService::getInput(int inputSource, - uint32_t samplingRate, - uint32_t format, - uint32_t channels, - AudioSystem::audio_in_acoustics acoustics) -{ - if (mpPolicyManager == NULL) { - return 0; - } - Mutex::Autolock _l(mLock); - return mpPolicyManager->getInput(inputSource, samplingRate, format, channels, acoustics); -} - -status_t AudioPolicyService::startInput(audio_io_handle_t input) -{ - if (mpPolicyManager == NULL) { - return NO_INIT; - } - Mutex::Autolock _l(mLock); - return mpPolicyManager->startInput(input); -} - -status_t AudioPolicyService::stopInput(audio_io_handle_t input) -{ - if (mpPolicyManager == NULL) { - return NO_INIT; - } - Mutex::Autolock _l(mLock); - return mpPolicyManager->stopInput(input); -} - -void AudioPolicyService::releaseInput(audio_io_handle_t input) -{ - if (mpPolicyManager == NULL) { - return; - } - Mutex::Autolock _l(mLock); - mpPolicyManager->releaseInput(input); -} - -status_t AudioPolicyService::initStreamVolume(AudioSystem::stream_type stream, - int indexMin, - int indexMax) -{ - if (mpPolicyManager == NULL) { - return NO_INIT; - } - if (!checkPermission()) { - return PERMISSION_DENIED; - } - if (stream < 0 || stream >= AudioSystem::NUM_STREAM_TYPES) { - return BAD_VALUE; - } - mpPolicyManager->initStreamVolume(stream, indexMin, indexMax); - return NO_ERROR; -} - -status_t AudioPolicyService::setStreamVolumeIndex(AudioSystem::stream_type stream, int index) -{ - if (mpPolicyManager == NULL) { - return NO_INIT; - } - if (!checkPermission()) { - return PERMISSION_DENIED; - } - if (stream < 0 || stream >= AudioSystem::NUM_STREAM_TYPES) { - return BAD_VALUE; - } - - return mpPolicyManager->setStreamVolumeIndex(stream, index); -} - -status_t AudioPolicyService::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index) -{ - if (mpPolicyManager == NULL) { - return NO_INIT; - } - if (!checkPermission()) { - return PERMISSION_DENIED; - } - if (stream < 0 || stream >= AudioSystem::NUM_STREAM_TYPES) { - return BAD_VALUE; - } - return mpPolicyManager->getStreamVolumeIndex(stream, index); -} - -void AudioPolicyService::binderDied(const wp<IBinder>& who) { - LOGW("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid()); -} - -static bool tryLock(Mutex& mutex) -{ - bool locked = false; - for (int i = 0; i < kDumpLockRetries; ++i) { - if (mutex.tryLock() == NO_ERROR) { - locked = true; - break; - } - usleep(kDumpLockSleep); - } - return locked; -} - -status_t AudioPolicyService::dumpInternals(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "PolicyManager Interface: %p\n", mpPolicyManager); - result.append(buffer); - snprintf(buffer, SIZE, "Command Thread: %p\n", mAudioCommandThread.get()); - result.append(buffer); - snprintf(buffer, SIZE, "Tones Thread: %p\n", mTonePlaybackThread.get()); - result.append(buffer); - - write(fd, result.string(), result.size()); - return NO_ERROR; -} - -status_t AudioPolicyService::dump(int fd, const Vector<String16>& args) -{ - if (checkCallingPermission(String16("android.permission.DUMP")) == false) { - dumpPermissionDenial(fd); - } else { - bool locked = tryLock(mLock); - if (!locked) { - String8 result(kDeadlockedString); - write(fd, result.string(), result.size()); - } - - dumpInternals(fd); - if (mAudioCommandThread != NULL) { - mAudioCommandThread->dump(fd); - } - if (mTonePlaybackThread != NULL) { - mTonePlaybackThread->dump(fd); - } - - if (mpPolicyManager) { - mpPolicyManager->dump(fd); - } - - if (locked) mLock.unlock(); - } - return NO_ERROR; -} - -status_t AudioPolicyService::dumpPermissionDenial(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - snprintf(buffer, SIZE, "Permission Denial: " - "can't dump AudioPolicyService from pid=%d, uid=%d\n", - IPCThreadState::self()->getCallingPid(), - IPCThreadState::self()->getCallingUid()); - result.append(buffer); - write(fd, result.string(), result.size()); - return NO_ERROR; -} - -status_t AudioPolicyService::onTransact( - uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) -{ - return BnAudioPolicyService::onTransact(code, data, reply, flags); -} - - -// ---------------------------------------------------------------------------- -void AudioPolicyService::instantiate() { - defaultServiceManager()->addService( - String16("media.audio_policy"), new AudioPolicyService()); -} - - -// ---------------------------------------------------------------------------- -// AudioPolicyClientInterface implementation -// ---------------------------------------------------------------------------- - - -audio_io_handle_t AudioPolicyService::openOutput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t *pLatencyMs, - AudioSystem::output_flags flags) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - LOGW("openOutput() could not get AudioFlinger"); - return 0; - } - - return af->openOutput(pDevices, pSamplingRate, (uint32_t *)pFormat, pChannels, pLatencyMs, flags); -} - -audio_io_handle_t AudioPolicyService::openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - LOGW("openDuplicateOutput() could not get AudioFlinger"); - return 0; - } - return af->openDuplicateOutput(output1, output2); -} - -status_t AudioPolicyService::closeOutput(audio_io_handle_t output) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) return PERMISSION_DENIED; - - return af->closeOutput(output); -} - - -status_t AudioPolicyService::suspendOutput(audio_io_handle_t output) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - LOGW("suspendOutput() could not get AudioFlinger"); - return PERMISSION_DENIED; - } - - return af->suspendOutput(output); -} - -status_t AudioPolicyService::restoreOutput(audio_io_handle_t output) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - LOGW("restoreOutput() could not get AudioFlinger"); - return PERMISSION_DENIED; - } - - return af->restoreOutput(output); -} - -audio_io_handle_t AudioPolicyService::openInput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t acoustics) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - LOGW("openInput() could not get AudioFlinger"); - return 0; - } - - return af->openInput(pDevices, pSamplingRate, (uint32_t *)pFormat, pChannels, acoustics); -} - -status_t AudioPolicyService::closeInput(audio_io_handle_t input) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) return PERMISSION_DENIED; - - return af->closeInput(input); -} - -status_t AudioPolicyService::setStreamVolume(AudioSystem::stream_type stream, float volume, audio_io_handle_t output, int delayMs) -{ - return mAudioCommandThread->volumeCommand((int)stream, volume, (int)output, delayMs); -} - -status_t AudioPolicyService::setStreamOutput(AudioSystem::stream_type stream, audio_io_handle_t output) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) return PERMISSION_DENIED; - - return af->setStreamOutput(stream, output); -} - - -void AudioPolicyService::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs, int delayMs) -{ - mAudioCommandThread->parametersCommand((int)ioHandle, keyValuePairs, delayMs); -} - -String8 AudioPolicyService::getParameters(audio_io_handle_t ioHandle, const String8& keys) -{ - String8 result = AudioSystem::getParameters(ioHandle, keys); - return result; -} - -status_t AudioPolicyService::startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream) -{ - mTonePlaybackThread->startToneCommand(tone, stream); - return NO_ERROR; -} - -status_t AudioPolicyService::stopTone() -{ - mTonePlaybackThread->stopToneCommand(); - return NO_ERROR; -} - -status_t AudioPolicyService::setVoiceVolume(float volume, int delayMs) -{ - return mAudioCommandThread->voiceVolumeCommand(volume, delayMs); -} - -// ----------- AudioPolicyService::AudioCommandThread implementation ---------- - -AudioPolicyService::AudioCommandThread::AudioCommandThread(String8 name) - : Thread(false), mName(name) -{ - mpToneGenerator = NULL; -} - - -AudioPolicyService::AudioCommandThread::~AudioCommandThread() -{ - if (mName != "" && !mAudioCommands.isEmpty()) { - release_wake_lock(mName.string()); - } - mAudioCommands.clear(); - if (mpToneGenerator != NULL) delete mpToneGenerator; -} - -void AudioPolicyService::AudioCommandThread::onFirstRef() -{ - if (mName != "") { - run(mName.string(), ANDROID_PRIORITY_AUDIO); - } else { - run("AudioCommandThread", ANDROID_PRIORITY_AUDIO); - } -} - -bool AudioPolicyService::AudioCommandThread::threadLoop() -{ - nsecs_t waitTime = INT64_MAX; - - mLock.lock(); - while (!exitPending()) - { - while(!mAudioCommands.isEmpty()) { - nsecs_t curTime = systemTime(); - // commands are sorted by increasing time stamp: execute them from index 0 and up - if (mAudioCommands[0]->mTime <= curTime) { - AudioCommand *command = mAudioCommands[0]; - mAudioCommands.removeAt(0); - mLastCommand = *command; - - switch (command->mCommand) { - case START_TONE: { - mLock.unlock(); - ToneData *data = (ToneData *)command->mParam; - LOGV("AudioCommandThread() processing start tone %d on stream %d", - data->mType, data->mStream); - if (mpToneGenerator != NULL) - delete mpToneGenerator; - mpToneGenerator = new ToneGenerator(data->mStream, 1.0); - mpToneGenerator->startTone(data->mType); - delete data; - mLock.lock(); - }break; - case STOP_TONE: { - mLock.unlock(); - LOGV("AudioCommandThread() processing stop tone"); - if (mpToneGenerator != NULL) { - mpToneGenerator->stopTone(); - delete mpToneGenerator; - mpToneGenerator = NULL; - } - mLock.lock(); - }break; - case SET_VOLUME: { - VolumeData *data = (VolumeData *)command->mParam; - LOGV("AudioCommandThread() processing set volume stream %d, volume %f, output %d", data->mStream, data->mVolume, data->mIO); - command->mStatus = AudioSystem::setStreamVolume(data->mStream, data->mVolume, data->mIO); - if (command->mWaitStatus) { - command->mCond.signal(); - mWaitWorkCV.wait(mLock); - } - delete data; - }break; - case SET_PARAMETERS: { - ParametersData *data = (ParametersData *)command->mParam; - LOGV("AudioCommandThread() processing set parameters string %s, io %d", data->mKeyValuePairs.string(), data->mIO); - command->mStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs); - if (command->mWaitStatus) { - command->mCond.signal(); - mWaitWorkCV.wait(mLock); - } - delete data; - }break; - case SET_VOICE_VOLUME: { - VoiceVolumeData *data = (VoiceVolumeData *)command->mParam; - LOGV("AudioCommandThread() processing set voice volume volume %f", data->mVolume); - command->mStatus = AudioSystem::setVoiceVolume(data->mVolume); - if (command->mWaitStatus) { - command->mCond.signal(); - mWaitWorkCV.wait(mLock); - } - delete data; - }break; - default: - LOGW("AudioCommandThread() unknown command %d", command->mCommand); - } - delete command; - waitTime = INT64_MAX; - } else { - waitTime = mAudioCommands[0]->mTime - curTime; - break; - } - } - // release delayed commands wake lock - if (mName != "" && mAudioCommands.isEmpty()) { - release_wake_lock(mName.string()); - } - LOGV("AudioCommandThread() going to sleep"); - mWaitWorkCV.waitRelative(mLock, waitTime); - LOGV("AudioCommandThread() waking up"); - } - mLock.unlock(); - return false; -} - -status_t AudioPolicyService::AudioCommandThread::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "AudioCommandThread %p Dump\n", this); - result.append(buffer); - write(fd, result.string(), result.size()); - - bool locked = tryLock(mLock); - if (!locked) { - String8 result2(kCmdDeadlockedString); - write(fd, result2.string(), result2.size()); - } - - snprintf(buffer, SIZE, "- Commands:\n"); - result = String8(buffer); - result.append(" Command Time Wait pParam\n"); - for (int i = 0; i < (int)mAudioCommands.size(); i++) { - mAudioCommands[i]->dump(buffer, SIZE); - result.append(buffer); - } - result.append(" Last Command\n"); - mLastCommand.dump(buffer, SIZE); - result.append(buffer); - - write(fd, result.string(), result.size()); - - if (locked) mLock.unlock(); - - return NO_ERROR; -} - -void AudioPolicyService::AudioCommandThread::startToneCommand(int type, int stream) -{ - AudioCommand *command = new AudioCommand(); - command->mCommand = START_TONE; - ToneData *data = new ToneData(); - data->mType = type; - data->mStream = stream; - command->mParam = (void *)data; - command->mWaitStatus = false; - Mutex::Autolock _l(mLock); - insertCommand_l(command); - LOGV("AudioCommandThread() adding tone start type %d, stream %d", type, stream); - mWaitWorkCV.signal(); -} - -void AudioPolicyService::AudioCommandThread::stopToneCommand() -{ - AudioCommand *command = new AudioCommand(); - command->mCommand = STOP_TONE; - command->mParam = NULL; - command->mWaitStatus = false; - Mutex::Autolock _l(mLock); - insertCommand_l(command); - LOGV("AudioCommandThread() adding tone stop"); - mWaitWorkCV.signal(); -} - -status_t AudioPolicyService::AudioCommandThread::volumeCommand(int stream, float volume, int output, int delayMs) -{ - status_t status = NO_ERROR; - - AudioCommand *command = new AudioCommand(); - command->mCommand = SET_VOLUME; - VolumeData *data = new VolumeData(); - data->mStream = stream; - data->mVolume = volume; - data->mIO = output; - command->mParam = data; - if (delayMs == 0) { - command->mWaitStatus = true; - } else { - command->mWaitStatus = false; - } - Mutex::Autolock _l(mLock); - insertCommand_l(command, delayMs); - LOGV("AudioCommandThread() adding set volume stream %d, volume %f, output %d", stream, volume, output); - mWaitWorkCV.signal(); - if (command->mWaitStatus) { - command->mCond.wait(mLock); - status = command->mStatus; - mWaitWorkCV.signal(); - } - return status; -} - -status_t AudioPolicyService::AudioCommandThread::parametersCommand(int ioHandle, const String8& keyValuePairs, int delayMs) -{ - status_t status = NO_ERROR; - - AudioCommand *command = new AudioCommand(); - command->mCommand = SET_PARAMETERS; - ParametersData *data = new ParametersData(); - data->mIO = ioHandle; - data->mKeyValuePairs = keyValuePairs; - command->mParam = data; - if (delayMs == 0) { - command->mWaitStatus = true; - } else { - command->mWaitStatus = false; - } - Mutex::Autolock _l(mLock); - insertCommand_l(command, delayMs); - LOGV("AudioCommandThread() adding set parameter string %s, io %d ,delay %d", keyValuePairs.string(), ioHandle, delayMs); - mWaitWorkCV.signal(); - if (command->mWaitStatus) { - command->mCond.wait(mLock); - status = command->mStatus; - mWaitWorkCV.signal(); - } - return status; -} - -status_t AudioPolicyService::AudioCommandThread::voiceVolumeCommand(float volume, int delayMs) -{ - status_t status = NO_ERROR; - - AudioCommand *command = new AudioCommand(); - command->mCommand = SET_VOICE_VOLUME; - VoiceVolumeData *data = new VoiceVolumeData(); - data->mVolume = volume; - command->mParam = data; - if (delayMs == 0) { - command->mWaitStatus = true; - } else { - command->mWaitStatus = false; - } - Mutex::Autolock _l(mLock); - insertCommand_l(command, delayMs); - LOGV("AudioCommandThread() adding set voice volume volume %f", volume); - mWaitWorkCV.signal(); - if (command->mWaitStatus) { - command->mCond.wait(mLock); - status = command->mStatus; - mWaitWorkCV.signal(); - } - return status; -} - -// insertCommand_l() must be called with mLock held -void AudioPolicyService::AudioCommandThread::insertCommand_l(AudioCommand *command, int delayMs) -{ - ssize_t i; - Vector <AudioCommand *> removedCommands; - - command->mTime = systemTime() + milliseconds(delayMs); - - // acquire wake lock to make sure delayed commands are processed - if (mName != "" && mAudioCommands.isEmpty()) { - acquire_wake_lock(PARTIAL_WAKE_LOCK, mName.string()); - } - - // check same pending commands with later time stamps and eliminate them - for (i = mAudioCommands.size()-1; i >= 0; i--) { - AudioCommand *command2 = mAudioCommands[i]; - // commands are sorted by increasing time stamp: no need to scan the rest of mAudioCommands - if (command2->mTime <= command->mTime) break; - if (command2->mCommand != command->mCommand) continue; - - switch (command->mCommand) { - case SET_PARAMETERS: { - ParametersData *data = (ParametersData *)command->mParam; - ParametersData *data2 = (ParametersData *)command2->mParam; - if (data->mIO != data2->mIO) break; - LOGV("Comparing parameter command %s to new command %s", data2->mKeyValuePairs.string(), data->mKeyValuePairs.string()); - AudioParameter param = AudioParameter(data->mKeyValuePairs); - AudioParameter param2 = AudioParameter(data2->mKeyValuePairs); - for (size_t j = 0; j < param.size(); j++) { - String8 key; - String8 value; - param.getAt(j, key, value); - for (size_t k = 0; k < param2.size(); k++) { - String8 key2; - String8 value2; - param2.getAt(k, key2, value2); - if (key2 == key) { - param2.remove(key2); - LOGV("Filtering out parameter %s", key2.string()); - break; - } - } - } - // if all keys have been filtered out, remove the command. - // otherwise, update the key value pairs - if (param2.size() == 0) { - removedCommands.add(command2); - } else { - data2->mKeyValuePairs = param2.toString(); - } - } break; - - case SET_VOLUME: { - VolumeData *data = (VolumeData *)command->mParam; - VolumeData *data2 = (VolumeData *)command2->mParam; - if (data->mIO != data2->mIO) break; - if (data->mStream != data2->mStream) break; - LOGV("Filtering out volume command on output %d for stream %d", data->mIO, data->mStream); - removedCommands.add(command2); - } break; - case START_TONE: - case STOP_TONE: - default: - break; - } - } - - // remove filtered commands - for (size_t j = 0; j < removedCommands.size(); j++) { - // removed commands always have time stamps greater than current command - for (size_t k = i + 1; k < mAudioCommands.size(); k++) { - if (mAudioCommands[k] == removedCommands[j]) { - LOGV("suppressing command: %d", mAudioCommands[k]->mCommand); - mAudioCommands.removeAt(k); - break; - } - } - } - removedCommands.clear(); - - // insert command at the right place according to its time stamp - LOGV("inserting command: %d at index %d, num commands %d", command->mCommand, (int)i+1, mAudioCommands.size()); - mAudioCommands.insertAt(command, i + 1); -} - -void AudioPolicyService::AudioCommandThread::exit() -{ - LOGV("AudioCommandThread::exit"); - { - AutoMutex _l(mLock); - requestExit(); - mWaitWorkCV.signal(); - } - requestExitAndWait(); -} - -void AudioPolicyService::AudioCommandThread::AudioCommand::dump(char* buffer, size_t size) -{ - snprintf(buffer, size, " %02d %06d.%03d %01u %p\n", - mCommand, - (int)ns2s(mTime), - (int)ns2ms(mTime)%1000, - mWaitStatus, - mParam); -} - -}; // namespace android diff --git a/libs/audioflinger/AudioPolicyService.h b/libs/audioflinger/AudioPolicyService.h deleted file mode 100644 index a13d0bd..0000000 --- a/libs/audioflinger/AudioPolicyService.h +++ /dev/null @@ -1,223 +0,0 @@ -/* - * Copyright (C) 2009 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#ifndef ANDROID_AUDIOPOLICYSERVICE_H -#define ANDROID_AUDIOPOLICYSERVICE_H - -#include <media/IAudioPolicyService.h> -#include <hardware_legacy/AudioPolicyInterface.h> -#include <media/ToneGenerator.h> -#include <utils/Vector.h> - -namespace android { - -class String8; - -// ---------------------------------------------------------------------------- - -class AudioPolicyService: public BnAudioPolicyService, public AudioPolicyClientInterface, public IBinder::DeathRecipient -{ - -public: - static void instantiate(); - - virtual status_t dump(int fd, const Vector<String16>& args); - - // - // BnAudioPolicyService (see AudioPolicyInterface for method descriptions) - // - - virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device, - AudioSystem::device_connection_state state, - const char *device_address); - virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device, - const char *device_address); - virtual status_t setPhoneState(int state); - virtual status_t setRingerMode(uint32_t mode, uint32_t mask); - virtual status_t setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config); - virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage); - virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream, - uint32_t samplingRate = 0, - uint32_t format = AudioSystem::FORMAT_DEFAULT, - uint32_t channels = 0, - AudioSystem::output_flags flags = AudioSystem::OUTPUT_FLAG_INDIRECT); - virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream); - virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream); - virtual void releaseOutput(audio_io_handle_t output); - virtual audio_io_handle_t getInput(int inputSource, - uint32_t samplingRate = 0, - uint32_t format = AudioSystem::FORMAT_DEFAULT, - uint32_t channels = 0, - AudioSystem::audio_in_acoustics acoustics = (AudioSystem::audio_in_acoustics)0); - virtual status_t startInput(audio_io_handle_t input); - virtual status_t stopInput(audio_io_handle_t input); - virtual void releaseInput(audio_io_handle_t input); - virtual status_t initStreamVolume(AudioSystem::stream_type stream, - int indexMin, - int indexMax); - virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index); - virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index); - - virtual status_t onTransact( - uint32_t code, - const Parcel& data, - Parcel* reply, - uint32_t flags); - - // IBinder::DeathRecipient - virtual void binderDied(const wp<IBinder>& who); - - // - // AudioPolicyClientInterface - // - virtual audio_io_handle_t openOutput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t *pLatencyMs, - AudioSystem::output_flags flags); - virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2); - virtual status_t closeOutput(audio_io_handle_t output); - virtual status_t suspendOutput(audio_io_handle_t output); - virtual status_t restoreOutput(audio_io_handle_t output); - virtual audio_io_handle_t openInput(uint32_t *pDevices, - uint32_t *pSamplingRate, - uint32_t *pFormat, - uint32_t *pChannels, - uint32_t acoustics); - virtual status_t closeInput(audio_io_handle_t input); - virtual status_t setStreamVolume(AudioSystem::stream_type stream, float volume, audio_io_handle_t output, int delayMs = 0); - virtual status_t setStreamOutput(AudioSystem::stream_type stream, audio_io_handle_t output); - virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs, int delayMs = 0); - virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); - virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream); - virtual status_t stopTone(); - virtual status_t setVoiceVolume(float volume, int delayMs = 0); - -private: - AudioPolicyService(); - virtual ~AudioPolicyService(); - - status_t dumpInternals(int fd); - - // Thread used for tone playback and to send audio config commands to audio flinger - // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because startTone() - // and stopTone() are normally called with mLock locked and requesting a tone start or stop will cause - // calls to AudioPolicyService and an attempt to lock mLock. - // For audio config commands, it is necessary because audio flinger requires that the calling process (user) - // has permission to modify audio settings. - class AudioCommandThread : public Thread { - class AudioCommand; - public: - - // commands for tone AudioCommand - enum { - START_TONE, - STOP_TONE, - SET_VOLUME, - SET_PARAMETERS, - SET_VOICE_VOLUME - }; - - AudioCommandThread (String8 name); - virtual ~AudioCommandThread(); - - status_t dump(int fd); - - // Thread virtuals - virtual void onFirstRef(); - virtual bool threadLoop(); - - void exit(); - void startToneCommand(int type = 0, int stream = 0); - void stopToneCommand(); - status_t volumeCommand(int stream, float volume, int output, int delayMs = 0); - status_t parametersCommand(int ioHandle, const String8& keyValuePairs, int delayMs = 0); - status_t voiceVolumeCommand(float volume, int delayMs = 0); - void insertCommand_l(AudioCommand *command, int delayMs = 0); - - private: - // descriptor for requested tone playback event - class AudioCommand { - - public: - AudioCommand() - : mCommand(-1) {} - - void dump(char* buffer, size_t size); - - int mCommand; // START_TONE, STOP_TONE ... - nsecs_t mTime; // time stamp - Condition mCond; // condition for status return - status_t mStatus; // command status - bool mWaitStatus; // true if caller is waiting for status - void *mParam; // command parameter (ToneData, VolumeData, ParametersData) - }; - - class ToneData { - public: - int mType; // tone type (START_TONE only) - int mStream; // stream type (START_TONE only) - }; - - class VolumeData { - public: - int mStream; - float mVolume; - int mIO; - }; - - class ParametersData { - public: - int mIO; - String8 mKeyValuePairs; - }; - - class VoiceVolumeData { - public: - float mVolume; - }; - - Mutex mLock; - Condition mWaitWorkCV; - Vector <AudioCommand *> mAudioCommands; // list of pending commands - ToneGenerator *mpToneGenerator; // the tone generator - AudioCommand mLastCommand; // last processed command (used by dump) - String8 mName; // string used by wake lock fo delayed commands - }; - - // Internal dump utilities. - status_t dumpPermissionDenial(int fd); - - - Mutex mLock; // prevents concurrent access to AudioPolicy manager functions changing device - // connection stated our routing - AudioPolicyInterface* mpPolicyManager; // the platform specific policy manager - sp <AudioCommandThread> mAudioCommandThread; // audio commands thread - sp <AudioCommandThread> mTonePlaybackThread; // tone playback thread -}; - -}; // namespace android - -#endif // ANDROID_AUDIOPOLICYSERVICE_H - - - - - - - - diff --git a/libs/audioflinger/AudioResampler.cpp b/libs/audioflinger/AudioResampler.cpp deleted file mode 100644 index 5dabacb..0000000 --- a/libs/audioflinger/AudioResampler.cpp +++ /dev/null @@ -1,595 +0,0 @@ -/* - * Copyright (C) 2007 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#define LOG_TAG "AudioResampler" -//#define LOG_NDEBUG 0 - -#include <stdint.h> -#include <stdlib.h> -#include <sys/types.h> -#include <cutils/log.h> -#include <cutils/properties.h> -#include "AudioResampler.h" -#include "AudioResamplerSinc.h" -#include "AudioResamplerCubic.h" - -namespace android { - -#ifdef __ARM_ARCH_5E__ // optimized asm option - #define ASM_ARM_RESAMP1 // enable asm optimisation for ResamplerOrder1 -#endif // __ARM_ARCH_5E__ -// ---------------------------------------------------------------------------- - -class AudioResamplerOrder1 : public AudioResampler { -public: - AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) : - AudioResampler(bitDepth, inChannelCount, sampleRate), mX0L(0), mX0R(0) { - } - virtual void resample(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider); -private: - // number of bits used in interpolation multiply - 15 bits avoids overflow - static const int kNumInterpBits = 15; - - // bits to shift the phase fraction down to avoid overflow - static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits; - - void init() {} - void resampleMono16(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider); - void resampleStereo16(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider); -#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1 - void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx, - size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr, - uint32_t &phaseFraction, uint32_t phaseIncrement); - void AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx, - size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr, - uint32_t &phaseFraction, uint32_t phaseIncrement); -#endif // ASM_ARM_RESAMP1 - - static inline int32_t Interp(int32_t x0, int32_t x1, uint32_t f) { - return x0 + (((x1 - x0) * (int32_t)(f >> kPreInterpShift)) >> kNumInterpBits); - } - static inline void Advance(size_t* index, uint32_t* frac, uint32_t inc) { - *frac += inc; - *index += (size_t)(*frac >> kNumPhaseBits); - *frac &= kPhaseMask; - } - int mX0L; - int mX0R; -}; - -// ---------------------------------------------------------------------------- -AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount, - int32_t sampleRate, int quality) { - - // can only create low quality resample now - AudioResampler* resampler; - - char value[PROPERTY_VALUE_MAX]; - if (property_get("af.resampler.quality", value, 0)) { - quality = atoi(value); - LOGD("forcing AudioResampler quality to %d", quality); - } - - if (quality == DEFAULT) - quality = LOW_QUALITY; - - switch (quality) { - default: - case LOW_QUALITY: - LOGV("Create linear Resampler"); - resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate); - break; - case MED_QUALITY: - LOGV("Create cubic Resampler"); - resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate); - break; - case HIGH_QUALITY: - LOGV("Create sinc Resampler"); - resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate); - break; - } - - // initialize resampler - resampler->init(); - return resampler; -} - -AudioResampler::AudioResampler(int bitDepth, int inChannelCount, - int32_t sampleRate) : - mBitDepth(bitDepth), mChannelCount(inChannelCount), - mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0), - mPhaseFraction(0) { - // sanity check on format - if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) { - LOGE("Unsupported sample format, %d bits, %d channels", bitDepth, - inChannelCount); - // LOG_ASSERT(0); - } - - // initialize common members - mVolume[0] = mVolume[1] = 0; - mBuffer.frameCount = 0; - - // save format for quick lookup - if (inChannelCount == 1) { - mFormat = MONO_16_BIT; - } else { - mFormat = STEREO_16_BIT; - } -} - -AudioResampler::~AudioResampler() { -} - -void AudioResampler::setSampleRate(int32_t inSampleRate) { - mInSampleRate = inSampleRate; - mPhaseIncrement = (uint32_t)((kPhaseMultiplier * inSampleRate) / mSampleRate); -} - -void AudioResampler::setVolume(int16_t left, int16_t right) { - // TODO: Implement anti-zipper filter - mVolume[0] = left; - mVolume[1] = right; -} - -// ---------------------------------------------------------------------------- - -void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider) { - - // should never happen, but we overflow if it does - // LOG_ASSERT(outFrameCount < 32767); - - // select the appropriate resampler - switch (mChannelCount) { - case 1: - resampleMono16(out, outFrameCount, provider); - break; - case 2: - resampleStereo16(out, outFrameCount, provider); - break; - } -} - -void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider) { - - int32_t vl = mVolume[0]; - int32_t vr = mVolume[1]; - - size_t inputIndex = mInputIndex; - uint32_t phaseFraction = mPhaseFraction; - uint32_t phaseIncrement = mPhaseIncrement; - size_t outputIndex = 0; - size_t outputSampleCount = outFrameCount * 2; - size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate; - - // LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n", - // outFrameCount, inputIndex, phaseFraction, phaseIncrement); - - while (outputIndex < outputSampleCount) { - - // buffer is empty, fetch a new one - while (mBuffer.frameCount == 0) { - mBuffer.frameCount = inFrameCount; - provider->getNextBuffer(&mBuffer); - if (mBuffer.raw == NULL) { - goto resampleStereo16_exit; - } - - // LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount); - if (mBuffer.frameCount > inputIndex) break; - - inputIndex -= mBuffer.frameCount; - mX0L = mBuffer.i16[mBuffer.frameCount*2-2]; - mX0R = mBuffer.i16[mBuffer.frameCount*2-1]; - provider->releaseBuffer(&mBuffer); - // mBuffer.frameCount == 0 now so we reload a new buffer - } - - int16_t *in = mBuffer.i16; - - // handle boundary case - while (inputIndex == 0) { - // LOGE("boundary case\n"); - out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction); - out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction); - Advance(&inputIndex, &phaseFraction, phaseIncrement); - if (outputIndex == outputSampleCount) - break; - } - - // process input samples - // LOGE("general case\n"); - -#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1 - if (inputIndex + 2 < mBuffer.frameCount) { - int32_t* maxOutPt; - int32_t maxInIdx; - - maxOutPt = out + (outputSampleCount - 2); // 2 because 2 frames per loop - maxInIdx = mBuffer.frameCount - 2; - AsmStereo16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr, - phaseFraction, phaseIncrement); - } -#endif // ASM_ARM_RESAMP1 - - while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) { - out[outputIndex++] += vl * Interp(in[inputIndex*2-2], - in[inputIndex*2], phaseFraction); - out[outputIndex++] += vr * Interp(in[inputIndex*2-1], - in[inputIndex*2+1], phaseFraction); - Advance(&inputIndex, &phaseFraction, phaseIncrement); - } - - // LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); - - // if done with buffer, save samples - if (inputIndex >= mBuffer.frameCount) { - inputIndex -= mBuffer.frameCount; - - // LOGE("buffer done, new input index %d", inputIndex); - - mX0L = mBuffer.i16[mBuffer.frameCount*2-2]; - mX0R = mBuffer.i16[mBuffer.frameCount*2-1]; - provider->releaseBuffer(&mBuffer); - - // verify that the releaseBuffer resets the buffer frameCount - // LOG_ASSERT(mBuffer.frameCount == 0); - } - } - - // LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); - -resampleStereo16_exit: - // save state - mInputIndex = inputIndex; - mPhaseFraction = phaseFraction; -} - -void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider) { - - int32_t vl = mVolume[0]; - int32_t vr = mVolume[1]; - - size_t inputIndex = mInputIndex; - uint32_t phaseFraction = mPhaseFraction; - uint32_t phaseIncrement = mPhaseIncrement; - size_t outputIndex = 0; - size_t outputSampleCount = outFrameCount * 2; - size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate; - - // LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n", - // outFrameCount, inputIndex, phaseFraction, phaseIncrement); - while (outputIndex < outputSampleCount) { - // buffer is empty, fetch a new one - while (mBuffer.frameCount == 0) { - mBuffer.frameCount = inFrameCount; - provider->getNextBuffer(&mBuffer); - if (mBuffer.raw == NULL) { - mInputIndex = inputIndex; - mPhaseFraction = phaseFraction; - goto resampleMono16_exit; - } - // LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount); - if (mBuffer.frameCount > inputIndex) break; - - inputIndex -= mBuffer.frameCount; - mX0L = mBuffer.i16[mBuffer.frameCount-1]; - provider->releaseBuffer(&mBuffer); - // mBuffer.frameCount == 0 now so we reload a new buffer - } - int16_t *in = mBuffer.i16; - - // handle boundary case - while (inputIndex == 0) { - // LOGE("boundary case\n"); - int32_t sample = Interp(mX0L, in[0], phaseFraction); - out[outputIndex++] += vl * sample; - out[outputIndex++] += vr * sample; - Advance(&inputIndex, &phaseFraction, phaseIncrement); - if (outputIndex == outputSampleCount) - break; - } - - // process input samples - // LOGE("general case\n"); - -#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1 - if (inputIndex + 2 < mBuffer.frameCount) { - int32_t* maxOutPt; - int32_t maxInIdx; - - maxOutPt = out + (outputSampleCount - 2); - maxInIdx = (int32_t)mBuffer.frameCount - 2; - AsmMono16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr, - phaseFraction, phaseIncrement); - } -#endif // ASM_ARM_RESAMP1 - - while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) { - int32_t sample = Interp(in[inputIndex-1], in[inputIndex], - phaseFraction); - out[outputIndex++] += vl * sample; - out[outputIndex++] += vr * sample; - Advance(&inputIndex, &phaseFraction, phaseIncrement); - } - - - // LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); - - // if done with buffer, save samples - if (inputIndex >= mBuffer.frameCount) { - inputIndex -= mBuffer.frameCount; - - // LOGE("buffer done, new input index %d", inputIndex); - - mX0L = mBuffer.i16[mBuffer.frameCount-1]; - provider->releaseBuffer(&mBuffer); - - // verify that the releaseBuffer resets the buffer frameCount - // LOG_ASSERT(mBuffer.frameCount == 0); - } - } - - // LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); - -resampleMono16_exit: - // save state - mInputIndex = inputIndex; - mPhaseFraction = phaseFraction; -} - -#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1 - -/******************************************************************* -* -* AsmMono16Loop -* asm optimized monotonic loop version; one loop is 2 frames -* Input: -* in : pointer on input samples -* maxOutPt : pointer on first not filled -* maxInIdx : index on first not used -* outputIndex : pointer on current output index -* out : pointer on output buffer -* inputIndex : pointer on current input index -* vl, vr : left and right gain -* phaseFraction : pointer on current phase fraction -* phaseIncrement -* Ouput: -* outputIndex : -* out : updated buffer -* inputIndex : index of next to use -* phaseFraction : phase fraction for next interpolation -* -*******************************************************************/ -void AudioResamplerOrder1::AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx, - size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr, - uint32_t &phaseFraction, uint32_t phaseIncrement) -{ -#define MO_PARAM5 "36" // offset of parameter 5 (outputIndex) - - asm( - "stmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, lr}\n" - // get parameters - " ldr r6, [sp, #" MO_PARAM5 " + 20]\n" // &phaseFraction - " ldr r6, [r6]\n" // phaseFraction - " ldr r7, [sp, #" MO_PARAM5 " + 8]\n" // &inputIndex - " ldr r7, [r7]\n" // inputIndex - " ldr r8, [sp, #" MO_PARAM5 " + 4]\n" // out - " ldr r0, [sp, #" MO_PARAM5 " + 0]\n" // &outputIndex - " ldr r0, [r0]\n" // outputIndex - " add r8, r0, asl #2\n" // curOut - " ldr r9, [sp, #" MO_PARAM5 " + 24]\n" // phaseIncrement - " ldr r10, [sp, #" MO_PARAM5 " + 12]\n" // vl - " ldr r11, [sp, #" MO_PARAM5 " + 16]\n" // vr - - // r0 pin, x0, Samp - - // r1 in - // r2 maxOutPt - // r3 maxInIdx - - // r4 x1, i1, i3, Out1 - // r5 out0 - - // r6 frac - // r7 inputIndex - // r8 curOut - - // r9 inc - // r10 vl - // r11 vr - - // r12 - // r13 sp - // r14 - - // the following loop works on 2 frames - - ".Y4L01:\n" - " cmp r8, r2\n" // curOut - maxCurOut - " bcs .Y4L02\n" - -#define MO_ONE_FRAME \ - " add r0, r1, r7, asl #1\n" /* in + inputIndex */\ - " ldrsh r4, [r0]\n" /* in[inputIndex] */\ - " ldr r5, [r8]\n" /* out[outputIndex] */\ - " ldrsh r0, [r0, #-2]\n" /* in[inputIndex-1] */\ - " bic r6, r6, #0xC0000000\n" /* phaseFraction & ... */\ - " sub r4, r4, r0\n" /* in[inputIndex] - in[inputIndex-1] */\ - " mov r4, r4, lsl #2\n" /* <<2 */\ - " smulwt r4, r4, r6\n" /* (x1-x0)*.. */\ - " add r6, r6, r9\n" /* phaseFraction + phaseIncrement */\ - " add r0, r0, r4\n" /* x0 - (..) */\ - " mla r5, r0, r10, r5\n" /* vl*interp + out[] */\ - " ldr r4, [r8, #4]\n" /* out[outputIndex+1] */\ - " str r5, [r8], #4\n" /* out[outputIndex++] = ... */\ - " mla r4, r0, r11, r4\n" /* vr*interp + out[] */\ - " add r7, r7, r6, lsr #30\n" /* inputIndex + phaseFraction>>30 */\ - " str r4, [r8], #4\n" /* out[outputIndex++] = ... */ - - MO_ONE_FRAME // frame 1 - MO_ONE_FRAME // frame 2 - - " cmp r7, r3\n" // inputIndex - maxInIdx - " bcc .Y4L01\n" - ".Y4L02:\n" - - " bic r6, r6, #0xC0000000\n" // phaseFraction & ... - // save modified values - " ldr r0, [sp, #" MO_PARAM5 " + 20]\n" // &phaseFraction - " str r6, [r0]\n" // phaseFraction - " ldr r0, [sp, #" MO_PARAM5 " + 8]\n" // &inputIndex - " str r7, [r0]\n" // inputIndex - " ldr r0, [sp, #" MO_PARAM5 " + 4]\n" // out - " sub r8, r0\n" // curOut - out - " asr r8, #2\n" // new outputIndex - " ldr r0, [sp, #" MO_PARAM5 " + 0]\n" // &outputIndex - " str r8, [r0]\n" // save outputIndex - - " ldmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, pc}\n" - ); -} - -/******************************************************************* -* -* AsmStereo16Loop -* asm optimized stereo loop version; one loop is 2 frames -* Input: -* in : pointer on input samples -* maxOutPt : pointer on first not filled -* maxInIdx : index on first not used -* outputIndex : pointer on current output index -* out : pointer on output buffer -* inputIndex : pointer on current input index -* vl, vr : left and right gain -* phaseFraction : pointer on current phase fraction -* phaseIncrement -* Ouput: -* outputIndex : -* out : updated buffer -* inputIndex : index of next to use -* phaseFraction : phase fraction for next interpolation -* -*******************************************************************/ -void AudioResamplerOrder1::AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx, - size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr, - uint32_t &phaseFraction, uint32_t phaseIncrement) -{ -#define ST_PARAM5 "40" // offset of parameter 5 (outputIndex) - asm( - "stmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, lr}\n" - // get parameters - " ldr r6, [sp, #" ST_PARAM5 " + 20]\n" // &phaseFraction - " ldr r6, [r6]\n" // phaseFraction - " ldr r7, [sp, #" ST_PARAM5 " + 8]\n" // &inputIndex - " ldr r7, [r7]\n" // inputIndex - " ldr r8, [sp, #" ST_PARAM5 " + 4]\n" // out - " ldr r0, [sp, #" ST_PARAM5 " + 0]\n" // &outputIndex - " ldr r0, [r0]\n" // outputIndex - " add r8, r0, asl #2\n" // curOut - " ldr r9, [sp, #" ST_PARAM5 " + 24]\n" // phaseIncrement - " ldr r10, [sp, #" ST_PARAM5 " + 12]\n" // vl - " ldr r11, [sp, #" ST_PARAM5 " + 16]\n" // vr - - // r0 pin, x0, Samp - - // r1 in - // r2 maxOutPt - // r3 maxInIdx - - // r4 x1, i1, i3, out1 - // r5 out0 - - // r6 frac - // r7 inputIndex - // r8 curOut - - // r9 inc - // r10 vl - // r11 vr - - // r12 temporary - // r13 sp - // r14 - - ".Y5L01:\n" - " cmp r8, r2\n" // curOut - maxCurOut - " bcs .Y5L02\n" - -#define ST_ONE_FRAME \ - " bic r6, r6, #0xC0000000\n" /* phaseFraction & ... */\ -\ - " add r0, r1, r7, asl #2\n" /* in + 2*inputIndex */\ -\ - " ldrsh r4, [r0]\n" /* in[2*inputIndex] */\ - " ldr r5, [r8]\n" /* out[outputIndex] */\ - " ldrsh r12, [r0, #-4]\n" /* in[2*inputIndex-2] */\ - " sub r4, r4, r12\n" /* in[2*InputIndex] - in[2*InputIndex-2] */\ - " mov r4, r4, lsl #2\n" /* <<2 */\ - " smulwt r4, r4, r6\n" /* (x1-x0)*.. */\ - " add r12, r12, r4\n" /* x0 - (..) */\ - " mla r5, r12, r10, r5\n" /* vl*interp + out[] */\ - " ldr r4, [r8, #4]\n" /* out[outputIndex+1] */\ - " str r5, [r8], #4\n" /* out[outputIndex++] = ... */\ -\ - " ldrsh r12, [r0, #+2]\n" /* in[2*inputIndex+1] */\ - " ldrsh r0, [r0, #-2]\n" /* in[2*inputIndex-1] */\ - " sub r12, r12, r0\n" /* in[2*InputIndex] - in[2*InputIndex-2] */\ - " mov r12, r12, lsl #2\n" /* <<2 */\ - " smulwt r12, r12, r6\n" /* (x1-x0)*.. */\ - " add r12, r0, r12\n" /* x0 - (..) */\ - " mla r4, r12, r11, r4\n" /* vr*interp + out[] */\ - " str r4, [r8], #4\n" /* out[outputIndex++] = ... */\ -\ - " add r6, r6, r9\n" /* phaseFraction + phaseIncrement */\ - " add r7, r7, r6, lsr #30\n" /* inputIndex + phaseFraction>>30 */ - - ST_ONE_FRAME // frame 1 - ST_ONE_FRAME // frame 1 - - " cmp r7, r3\n" // inputIndex - maxInIdx - " bcc .Y5L01\n" - ".Y5L02:\n" - - " bic r6, r6, #0xC0000000\n" // phaseFraction & ... - // save modified values - " ldr r0, [sp, #" ST_PARAM5 " + 20]\n" // &phaseFraction - " str r6, [r0]\n" // phaseFraction - " ldr r0, [sp, #" ST_PARAM5 " + 8]\n" // &inputIndex - " str r7, [r0]\n" // inputIndex - " ldr r0, [sp, #" ST_PARAM5 " + 4]\n" // out - " sub r8, r0\n" // curOut - out - " asr r8, #2\n" // new outputIndex - " ldr r0, [sp, #" ST_PARAM5 " + 0]\n" // &outputIndex - " str r8, [r0]\n" // save outputIndex - - " ldmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, pc}\n" - ); -} - -#endif // ASM_ARM_RESAMP1 - - -// ---------------------------------------------------------------------------- -} -; // namespace android - diff --git a/libs/audioflinger/AudioResampler.h b/libs/audioflinger/AudioResampler.h deleted file mode 100644 index 2dfac76..0000000 --- a/libs/audioflinger/AudioResampler.h +++ /dev/null @@ -1,93 +0,0 @@ -/* - * Copyright (C) 2007 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#ifndef ANDROID_AUDIO_RESAMPLER_H -#define ANDROID_AUDIO_RESAMPLER_H - -#include <stdint.h> -#include <sys/types.h> - -#include "AudioBufferProvider.h" - -namespace android { -// ---------------------------------------------------------------------------- - -class AudioResampler { -public: - // Determines quality of SRC. - // LOW_QUALITY: linear interpolator (1st order) - // MED_QUALITY: cubic interpolator (3rd order) - // HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz) - // NOTE: high quality SRC will only be supported for - // certain fixed rate conversions. Sample rate cannot be - // changed dynamically. - enum src_quality { - DEFAULT=0, - LOW_QUALITY=1, - MED_QUALITY=2, - HIGH_QUALITY=3 - }; - - static AudioResampler* create(int bitDepth, int inChannelCount, - int32_t sampleRate, int quality=DEFAULT); - - virtual ~AudioResampler(); - - virtual void init() = 0; - virtual void setSampleRate(int32_t inSampleRate); - virtual void setVolume(int16_t left, int16_t right); - - virtual void resample(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider) = 0; - -protected: - // number of bits for phase fraction - 30 bits allows nearly 2x downsampling - static const int kNumPhaseBits = 30; - - // phase mask for fraction - static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1; - - // multiplier to calculate fixed point phase increment - static const double kPhaseMultiplier = 1L << kNumPhaseBits; - - enum format {MONO_16_BIT, STEREO_16_BIT}; - AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate); - - // prevent copying - AudioResampler(const AudioResampler&); - AudioResampler& operator=(const AudioResampler&); - - int32_t mBitDepth; - int32_t mChannelCount; - int32_t mSampleRate; - int32_t mInSampleRate; - AudioBufferProvider::Buffer mBuffer; - union { - int16_t mVolume[2]; - uint32_t mVolumeRL; - }; - int16_t mTargetVolume[2]; - format mFormat; - size_t mInputIndex; - int32_t mPhaseIncrement; - uint32_t mPhaseFraction; -}; - -// ---------------------------------------------------------------------------- -} -; // namespace android - -#endif // ANDROID_AUDIO_RESAMPLER_H diff --git a/libs/audioflinger/AudioResamplerCubic.cpp b/libs/audioflinger/AudioResamplerCubic.cpp deleted file mode 100644 index 1d247bd..0000000 --- a/libs/audioflinger/AudioResamplerCubic.cpp +++ /dev/null @@ -1,184 +0,0 @@ -/* - * Copyright (C) 2007 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#include <stdint.h> -#include <string.h> -#include <sys/types.h> -#include <cutils/log.h> - -#include "AudioResampler.h" -#include "AudioResamplerCubic.h" - -#define LOG_TAG "AudioSRC" - -namespace android { -// ---------------------------------------------------------------------------- - -void AudioResamplerCubic::init() { - memset(&left, 0, sizeof(state)); - memset(&right, 0, sizeof(state)); -} - -void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider) { - - // should never happen, but we overflow if it does - // LOG_ASSERT(outFrameCount < 32767); - - // select the appropriate resampler - switch (mChannelCount) { - case 1: - resampleMono16(out, outFrameCount, provider); - break; - case 2: - resampleStereo16(out, outFrameCount, provider); - break; - } -} - -void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider) { - - int32_t vl = mVolume[0]; - int32_t vr = mVolume[1]; - - size_t inputIndex = mInputIndex; - uint32_t phaseFraction = mPhaseFraction; - uint32_t phaseIncrement = mPhaseIncrement; - size_t outputIndex = 0; - size_t outputSampleCount = outFrameCount * 2; - size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate; - - // fetch first buffer - if (mBuffer.frameCount == 0) { - mBuffer.frameCount = inFrameCount; - provider->getNextBuffer(&mBuffer); - if (mBuffer.raw == NULL) - return; - // LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); - } - int16_t *in = mBuffer.i16; - - while (outputIndex < outputSampleCount) { - int32_t sample; - int32_t x; - - // calculate output sample - x = phaseFraction >> kPreInterpShift; - out[outputIndex++] += vl * interp(&left, x); - out[outputIndex++] += vr * interp(&right, x); - // out[outputIndex++] += vr * in[inputIndex*2]; - - // increment phase - phaseFraction += phaseIncrement; - uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits); - phaseFraction &= kPhaseMask; - - // time to fetch another sample - while (indexIncrement--) { - - inputIndex++; - if (inputIndex == mBuffer.frameCount) { - inputIndex = 0; - provider->releaseBuffer(&mBuffer); - mBuffer.frameCount = inFrameCount; - provider->getNextBuffer(&mBuffer); - if (mBuffer.raw == NULL) - goto save_state; // ugly, but efficient - in = mBuffer.i16; - // LOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount); - } - - // advance sample state - advance(&left, in[inputIndex*2]); - advance(&right, in[inputIndex*2+1]); - } - } - -save_state: - // LOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction); - mInputIndex = inputIndex; - mPhaseFraction = phaseFraction; -} - -void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider) { - - int32_t vl = mVolume[0]; - int32_t vr = mVolume[1]; - - size_t inputIndex = mInputIndex; - uint32_t phaseFraction = mPhaseFraction; - uint32_t phaseIncrement = mPhaseIncrement; - size_t outputIndex = 0; - size_t outputSampleCount = outFrameCount * 2; - size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate; - - // fetch first buffer - if (mBuffer.frameCount == 0) { - mBuffer.frameCount = inFrameCount; - provider->getNextBuffer(&mBuffer); - if (mBuffer.raw == NULL) - return; - // LOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount); - } - int16_t *in = mBuffer.i16; - - while (outputIndex < outputSampleCount) { - int32_t sample; - int32_t x; - - // calculate output sample - x = phaseFraction >> kPreInterpShift; - sample = interp(&left, x); - out[outputIndex++] += vl * sample; - out[outputIndex++] += vr * sample; - - // increment phase - phaseFraction += phaseIncrement; - uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits); - phaseFraction &= kPhaseMask; - - // time to fetch another sample - while (indexIncrement--) { - - inputIndex++; - if (inputIndex == mBuffer.frameCount) { - inputIndex = 0; - provider->releaseBuffer(&mBuffer); - mBuffer.frameCount = inFrameCount; - provider->getNextBuffer(&mBuffer); - if (mBuffer.raw == NULL) - goto save_state; // ugly, but efficient - // LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); - in = mBuffer.i16; - } - - // advance sample state - advance(&left, in[inputIndex]); - } - } - -save_state: - // LOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction); - mInputIndex = inputIndex; - mPhaseFraction = phaseFraction; -} - -// ---------------------------------------------------------------------------- -} -; // namespace android - diff --git a/libs/audioflinger/AudioResamplerCubic.h b/libs/audioflinger/AudioResamplerCubic.h deleted file mode 100644 index b72b62a..0000000 --- a/libs/audioflinger/AudioResamplerCubic.h +++ /dev/null @@ -1,68 +0,0 @@ -/* - * Copyright (C) 2007 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#ifndef ANDROID_AUDIO_RESAMPLER_CUBIC_H -#define ANDROID_AUDIO_RESAMPLER_CUBIC_H - -#include <stdint.h> -#include <sys/types.h> -#include <cutils/log.h> - -#include "AudioResampler.h" - -namespace android { -// ---------------------------------------------------------------------------- - -class AudioResamplerCubic : public AudioResampler { -public: - AudioResamplerCubic(int bitDepth, int inChannelCount, int32_t sampleRate) : - AudioResampler(bitDepth, inChannelCount, sampleRate) { - } - virtual void resample(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider); -private: - // number of bits used in interpolation multiply - 14 bits avoids overflow - static const int kNumInterpBits = 14; - - // bits to shift the phase fraction down to avoid overflow - static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits; - typedef struct { - int32_t a, b, c, y0, y1, y2, y3; - } state; - void init(); - void resampleMono16(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider); - void resampleStereo16(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider); - static inline int32_t interp(state* p, int32_t x) { - return (((((p->a * x >> 14) + p->b) * x >> 14) + p->c) * x >> 14) + p->y1; - } - static inline void advance(state* p, int16_t in) { - p->y0 = p->y1; - p->y1 = p->y2; - p->y2 = p->y3; - p->y3 = in; - p->a = (3 * (p->y1 - p->y2) - p->y0 + p->y3) >> 1; - p->b = (p->y2 << 1) + p->y0 - (((5 * p->y1 + p->y3)) >> 1); - p->c = (p->y2 - p->y0) >> 1; - } - state left, right; -}; - -// ---------------------------------------------------------------------------- -}; // namespace android - -#endif /*ANDROID_AUDIO_RESAMPLER_CUBIC_H*/ diff --git a/libs/audioflinger/AudioResamplerSinc.cpp b/libs/audioflinger/AudioResamplerSinc.cpp deleted file mode 100644 index 9e5e254..0000000 --- a/libs/audioflinger/AudioResamplerSinc.cpp +++ /dev/null @@ -1,358 +0,0 @@ -/* - * Copyright (C) 2007 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#include <string.h> -#include "AudioResamplerSinc.h" - -namespace android { -// ---------------------------------------------------------------------------- - - -/* - * These coeficients are computed with the "fir" utility found in - * tools/resampler_tools - * TODO: A good optimization would be to transpose this matrix, to take - * better advantage of the data-cache. - */ -const int32_t AudioResamplerSinc::mFirCoefsUp[] = { - 0x7fffffff, 0x7f15d078, 0x7c5e0da6, 0x77ecd867, 0x71e2e251, 0x6a6c304a, 0x61be7269, 0x58170412, 0x4db8ab05, 0x42e92ea6, 0x37eee214, 0x2d0e3bb1, 0x22879366, 0x18951e95, 0x0f693d0d, 0x072d2621, - 0x00000000, 0xf9f66655, 0xf51a5fd7, 0xf16bbd84, 0xeee0d9ac, 0xed67a922, 0xece70de6, 0xed405897, 0xee50e505, 0xeff3be30, 0xf203370f, 0xf45a6741, 0xf6d67d53, 0xf957db66, 0xfbc2f647, 0xfe00f2b9, - 0x00000000, 0x01b37218, 0x0313a0c6, 0x041d930d, 0x04d28057, 0x053731b0, 0x05534dff, 0x05309bfd, 0x04da440d, 0x045c1aee, 0x03c1fcdd, 0x03173ef5, 0x02663ae8, 0x01b7f736, 0x0113ec79, 0x007fe6a9, - 0x00000000, 0xff96b229, 0xff44f99f, 0xff0a86be, 0xfee5f803, 0xfed518fd, 0xfed521fd, 0xfee2f4fd, 0xfefb54f8, 0xff1b159b, 0xff3f4203, 0xff6539e0, 0xff8ac502, 0xffae1ddd, 0xffcdf3f9, 0xffe96798, - 0x00000000, 0x00119de6, 0x001e6b7e, 0x0026cb7a, 0x002b4830, 0x002c83d6, 0x002b2a82, 0x0027e67a, 0x002356f9, 0x001e098e, 0x001875e4, 0x0012fbbe, 0x000de2d1, 0x00095c10, 0x00058414, 0x00026636, - 0x00000000, 0xfffe44a9, 0xfffd206d, 0xfffc7b7f, 0xfffc3c8f, 0xfffc4ac2, 0xfffc8f2b, 0xfffcf5c4, 0xfffd6df3, 0xfffdeab2, 0xfffe6275, 0xfffececf, 0xffff2c07, 0xffff788c, 0xffffb471, 0xffffe0f2, - 0x00000000, 0x000013e6, 0x00001f03, 0x00002396, 0x00002399, 0x000020b6, 0x00001c3c, 0x00001722, 0x00001216, 0x00000d81, 0x0000099c, 0x0000067c, 0x00000419, 0x0000025f, 0x00000131, 0x00000070, - 0x00000000, 0xffffffc7, 0xffffffb3, 0xffffffb3, 0xffffffbe, 0xffffffcd, 0xffffffdb, 0xffffffe7, 0xfffffff0, 0xfffffff7, 0xfffffffb, 0xfffffffe, 0xffffffff, 0x00000000, 0x00000000, 0x00000000, - 0x00000000 // this one is needed for lerping the last coefficient -}; - -/* - * These coefficients are optimized for 48KHz -> 44.1KHz (stop-band at 22.050KHz) - * It's possible to use the above coefficient for any down-sampling - * at the expense of a slower processing loop (we can interpolate - * these coefficient from the above by "Stretching" them in time). - */ -const int32_t AudioResamplerSinc::mFirCoefsDown[] = { - 0x7fffffff, 0x7f55e46d, 0x7d5b4c60, 0x7a1b4b98, 0x75a7fb14, 0x7019f0bd, 0x698f875a, 0x622bfd59, 0x5a167256, 0x5178cc54, 0x487e8e6c, 0x3f53aae8, 0x36235ad4, 0x2d17047b, 0x245539ab, 0x1c00d540, - 0x14383e57, 0x0d14d5ca, 0x06aa910b, 0x0107c38b, 0xfc351654, 0xf835abae, 0xf5076b45, 0xf2a37202, 0xf0fe9faa, 0xf00a3bbd, 0xefb4aa81, 0xefea2b05, 0xf0959716, 0xf1a11e83, 0xf2f6f7a0, 0xf481fff4, - 0xf62e48ce, 0xf7e98ca5, 0xf9a38b4c, 0xfb4e4bfa, 0xfcde456f, 0xfe4a6d30, 0xff8c2fdf, 0x009f5555, 0x0181d393, 0x0233940f, 0x02b62f06, 0x030ca07d, 0x033afa62, 0x03461725, 0x03334f83, 0x030835fa, - 0x02ca59cc, 0x027f12d1, 0x022b570d, 0x01d39a49, 0x017bb78f, 0x0126e414, 0x00d7aaaf, 0x008feec7, 0x0050f584, 0x001b73e3, 0xffefa063, 0xffcd46ed, 0xffb3ddcd, 0xffa29aaa, 0xff988691, 0xff949066, - 0xff959d24, 0xff9a959e, 0xffa27195, 0xffac4011, 0xffb72d2b, 0xffc28569, 0xffcdb706, 0xffd85171, 0xffe20364, 0xffea97e9, 0xfff1f2b2, 0xfff80c06, 0xfffcec92, 0x0000a955, 0x00035fd8, 0x000532cf, - 0x00064735, 0x0006c1f9, 0x0006c62d, 0x000673ba, 0x0005e68f, 0x00053630, 0x000475a3, 0x0003b397, 0x0002fac1, 0x00025257, 0x0001be9e, 0x0001417a, 0x0000dafd, 0x000089eb, 0x00004c28, 0x00001f1d, - 0x00000000, 0xffffec10, 0xffffe0be, 0xffffdbc5, 0xffffdb39, 0xffffdd8b, 0xffffe182, 0xffffe638, 0xffffeb0a, 0xffffef8f, 0xfffff38b, 0xfffff6e3, 0xfffff993, 0xfffffba6, 0xfffffd30, 0xfffffe4a, - 0xffffff09, 0xffffff85, 0xffffffd1, 0xfffffffb, 0x0000000f, 0x00000016, 0x00000015, 0x00000012, 0x0000000d, 0x00000009, 0x00000006, 0x00000003, 0x00000002, 0x00000001, 0x00000000, 0x00000000, - 0x00000000 // this one is needed for lerping the last coefficient -}; - -// ---------------------------------------------------------------------------- - -static inline -int32_t mulRL(int left, int32_t in, uint32_t vRL) -{ -#if defined(__arm__) && !defined(__thumb__) - int32_t out; - if (left) { - asm( "smultb %[out], %[in], %[vRL] \n" - : [out]"=r"(out) - : [in]"%r"(in), [vRL]"r"(vRL) - : ); - } else { - asm( "smultt %[out], %[in], %[vRL] \n" - : [out]"=r"(out) - : [in]"%r"(in), [vRL]"r"(vRL) - : ); - } - return out; -#else - if (left) { - return int16_t(in>>16) * int16_t(vRL&0xFFFF); - } else { - return int16_t(in>>16) * int16_t(vRL>>16); - } -#endif -} - -static inline -int32_t mulAdd(int16_t in, int32_t v, int32_t a) -{ -#if defined(__arm__) && !defined(__thumb__) - int32_t out; - asm( "smlawb %[out], %[v], %[in], %[a] \n" - : [out]"=r"(out) - : [in]"%r"(in), [v]"r"(v), [a]"r"(a) - : ); - return out; -#else - return a + in * (v>>16); - // improved precision - // return a + in * (v>>16) + ((in * (v & 0xffff)) >> 16); -#endif -} - -static inline -int32_t mulAddRL(int left, uint32_t inRL, int32_t v, int32_t a) -{ -#if defined(__arm__) && !defined(__thumb__) - int32_t out; - if (left) { - asm( "smlawb %[out], %[v], %[inRL], %[a] \n" - : [out]"=r"(out) - : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a) - : ); - } else { - asm( "smlawt %[out], %[v], %[inRL], %[a] \n" - : [out]"=r"(out) - : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a) - : ); - } - return out; -#else - if (left) { - return a + (int16_t(inRL&0xFFFF) * (v>>16)); - //improved precision - // return a + (int16_t(inRL&0xFFFF) * (v>>16)) + ((int16_t(inRL&0xFFFF) * (v & 0xffff)) >> 16); - } else { - return a + (int16_t(inRL>>16) * (v>>16)); - } -#endif -} - -// ---------------------------------------------------------------------------- - -AudioResamplerSinc::AudioResamplerSinc(int bitDepth, - int inChannelCount, int32_t sampleRate) - : AudioResampler(bitDepth, inChannelCount, sampleRate), - mState(0) -{ - /* - * Layout of the state buffer for 32 tap: - * - * "present" sample beginning of 2nd buffer - * v v - * 0 01 2 23 3 - * 0 F0 0 F0 F - * [pppppppppppppppInnnnnnnnnnnnnnnnpppppppppppppppInnnnnnnnnnnnnnnn] - * ^ ^ head - * - * p = past samples, convoluted with the (p)ositive side of sinc() - * n = future samples, convoluted with the (n)egative side of sinc() - * r = extra space for implementing the ring buffer - * - */ - - const size_t numCoefs = 2*halfNumCoefs; - const size_t stateSize = numCoefs * inChannelCount * 2; - mState = new int16_t[stateSize]; - memset(mState, 0, sizeof(int16_t)*stateSize); - mImpulse = mState + (halfNumCoefs-1)*inChannelCount; - mRingFull = mImpulse + (numCoefs+1)*inChannelCount; -} - -AudioResamplerSinc::~AudioResamplerSinc() -{ - delete [] mState; -} - -void AudioResamplerSinc::init() { -} - -void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider) -{ - mFirCoefs = (mInSampleRate <= mSampleRate) ? mFirCoefsUp : mFirCoefsDown; - - // select the appropriate resampler - switch (mChannelCount) { - case 1: - resample<1>(out, outFrameCount, provider); - break; - case 2: - resample<2>(out, outFrameCount, provider); - break; - } -} - - -template<int CHANNELS> -void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider) -{ - int16_t* impulse = mImpulse; - uint32_t vRL = mVolumeRL; - size_t inputIndex = mInputIndex; - uint32_t phaseFraction = mPhaseFraction; - uint32_t phaseIncrement = mPhaseIncrement; - size_t outputIndex = 0; - size_t outputSampleCount = outFrameCount * 2; - size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate; - - AudioBufferProvider::Buffer& buffer(mBuffer); - while (outputIndex < outputSampleCount) { - // buffer is empty, fetch a new one - while (buffer.frameCount == 0) { - buffer.frameCount = inFrameCount; - provider->getNextBuffer(&buffer); - if (buffer.raw == NULL) { - goto resample_exit; - } - const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits; - if (phaseIndex == 1) { - // read one frame - read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex); - } else if (phaseIndex == 2) { - // read 2 frames - read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex); - inputIndex++; - if (inputIndex >= mBuffer.frameCount) { - inputIndex -= mBuffer.frameCount; - provider->releaseBuffer(&buffer); - } else { - read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex); - } - } - } - int16_t *in = buffer.i16; - const size_t frameCount = buffer.frameCount; - - // Always read-in the first samples from the input buffer - int16_t* head = impulse + halfNumCoefs*CHANNELS; - head[0] = in[inputIndex*CHANNELS + 0]; - if (CHANNELS == 2) - head[1] = in[inputIndex*CHANNELS + 1]; - - // handle boundary case - int32_t l, r; - while (outputIndex < outputSampleCount) { - filterCoefficient<CHANNELS>(l, r, phaseFraction, impulse); - out[outputIndex++] += 2 * mulRL(1, l, vRL); - out[outputIndex++] += 2 * mulRL(0, r, vRL); - - phaseFraction += phaseIncrement; - const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits; - if (phaseIndex == 1) { - inputIndex++; - if (inputIndex >= frameCount) - break; // need a new buffer - read<CHANNELS>(impulse, phaseFraction, in, inputIndex); - } else if(phaseIndex == 2) { // maximum value - inputIndex++; - if (inputIndex >= frameCount) - break; // 0 frame available, 2 frames needed - // read first frame - read<CHANNELS>(impulse, phaseFraction, in, inputIndex); - inputIndex++; - if (inputIndex >= frameCount) - break; // 0 frame available, 1 frame needed - // read second frame - read<CHANNELS>(impulse, phaseFraction, in, inputIndex); - } - } - - // if done with buffer, save samples - if (inputIndex >= frameCount) { - inputIndex -= frameCount; - provider->releaseBuffer(&buffer); - } - } - -resample_exit: - mImpulse = impulse; - mInputIndex = inputIndex; - mPhaseFraction = phaseFraction; -} - -template<int CHANNELS> -/*** -* read() -* -* This function reads only one frame from input buffer and writes it in -* state buffer -* -**/ -void AudioResamplerSinc::read( - int16_t*& impulse, uint32_t& phaseFraction, - int16_t const* in, size_t inputIndex) -{ - const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits; - impulse += CHANNELS; - phaseFraction -= 1LU<<kNumPhaseBits; - if (impulse >= mRingFull) { - const size_t stateSize = (halfNumCoefs*2)*CHANNELS; - memcpy(mState, mState+stateSize, sizeof(int16_t)*stateSize); - impulse -= stateSize; - } - int16_t* head = impulse + halfNumCoefs*CHANNELS; - head[0] = in[inputIndex*CHANNELS + 0]; - if (CHANNELS == 2) - head[1] = in[inputIndex*CHANNELS + 1]; -} - -template<int CHANNELS> -void AudioResamplerSinc::filterCoefficient( - int32_t& l, int32_t& r, uint32_t phase, int16_t const *samples) -{ - // compute the index of the coefficient on the positive side and - // negative side - uint32_t indexP = (phase & cMask) >> cShift; - uint16_t lerpP = (phase & pMask) >> pShift; - uint32_t indexN = (-phase & cMask) >> cShift; - uint16_t lerpN = (-phase & pMask) >> pShift; - if ((indexP == 0) && (lerpP == 0)) { - indexN = cMask >> cShift; - lerpN = pMask >> pShift; - } - - l = 0; - r = 0; - int32_t const* coefs = mFirCoefs; - int16_t const *sP = samples; - int16_t const *sN = samples+CHANNELS; - for (unsigned int i=0 ; i<halfNumCoefs/4 ; i++) { - interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); - interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN); - sP -= CHANNELS; sN += CHANNELS; coefs += 1<<coefsBits; - interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); - interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN); - sP -= CHANNELS; sN += CHANNELS; coefs += 1<<coefsBits; - interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); - interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN); - sP -= CHANNELS; sN += CHANNELS; coefs += 1<<coefsBits; - interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); - interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN); - sP -= CHANNELS; sN += CHANNELS; coefs += 1<<coefsBits; - } -} - -template<int CHANNELS> -void AudioResamplerSinc::interpolate( - int32_t& l, int32_t& r, - int32_t const* coefs, int16_t lerp, int16_t const* samples) -{ - int32_t c0 = coefs[0]; - int32_t c1 = coefs[1]; - int32_t sinc = mulAdd(lerp, (c1-c0)<<1, c0); - if (CHANNELS == 2) { - uint32_t rl = *reinterpret_cast<uint32_t const*>(samples); - l = mulAddRL(1, rl, sinc, l); - r = mulAddRL(0, rl, sinc, r); - } else { - r = l = mulAdd(samples[0], sinc, l); - } -} - -// ---------------------------------------------------------------------------- -}; // namespace android - diff --git a/libs/audioflinger/AudioResamplerSinc.h b/libs/audioflinger/AudioResamplerSinc.h deleted file mode 100644 index e6cb90b..0000000 --- a/libs/audioflinger/AudioResamplerSinc.h +++ /dev/null @@ -1,88 +0,0 @@ -/* - * Copyright (C) 2007 The Android Open Source Project - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#ifndef ANDROID_AUDIO_RESAMPLER_SINC_H -#define ANDROID_AUDIO_RESAMPLER_SINC_H - -#include <stdint.h> -#include <sys/types.h> -#include <cutils/log.h> - -#include "AudioResampler.h" - -namespace android { - -// ---------------------------------------------------------------------------- - -class AudioResamplerSinc : public AudioResampler { -public: - AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate); - - ~AudioResamplerSinc(); - - virtual void resample(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider); -private: - void init(); - - template<int CHANNELS> - void resample(int32_t* out, size_t outFrameCount, - AudioBufferProvider* provider); - - template<int CHANNELS> - inline void filterCoefficient( - int32_t& l, int32_t& r, uint32_t phase, int16_t const *samples); - - template<int CHANNELS> - inline void interpolate( - int32_t& l, int32_t& r, - int32_t const* coefs, int16_t lerp, int16_t const* samples); - - template<int CHANNELS> - inline void read(int16_t*& impulse, uint32_t& phaseFraction, - int16_t const* in, size_t inputIndex); - - int16_t *mState; - int16_t *mImpulse; - int16_t *mRingFull; - - int32_t const * mFirCoefs; - static const int32_t mFirCoefsDown[]; - static const int32_t mFirCoefsUp[]; - - // ---------------------------------------------------------------------------- - static const int32_t RESAMPLE_FIR_NUM_COEF = 8; - static const int32_t RESAMPLE_FIR_LERP_INT_BITS = 4; - - // we have 16 coefs samples per zero-crossing - static const int coefsBits = RESAMPLE_FIR_LERP_INT_BITS; // 4 - static const int cShift = kNumPhaseBits - coefsBits; // 26 - static const uint32_t cMask = ((1<<coefsBits)-1) << cShift; // 0xf<<26 = 3c00 0000 - - // and we use 15 bits to interpolate between these samples - // this cannot change because the mul below rely on it. - static const int pLerpBits = 15; - static const int pShift = kNumPhaseBits - coefsBits - pLerpBits; // 11 - static const uint32_t pMask = ((1<<pLerpBits)-1) << pShift; // 0x7fff << 11 - - // number of zero-crossing on each side - static const unsigned int halfNumCoefs = RESAMPLE_FIR_NUM_COEF; -}; - -// ---------------------------------------------------------------------------- -}; // namespace android - -#endif /*ANDROID_AUDIO_RESAMPLER_SINC_H*/ diff --git a/libs/surfaceflinger/Android.mk b/services/surfaceflinger/Android.mk index a14bfb5..a14bfb5 100644 --- a/libs/surfaceflinger/Android.mk +++ b/services/surfaceflinger/Android.mk diff --git a/libs/surfaceflinger/Barrier.h b/services/surfaceflinger/Barrier.h index 6f8507e..6f8507e 100644 --- a/libs/surfaceflinger/Barrier.h +++ b/services/surfaceflinger/Barrier.h diff --git a/libs/surfaceflinger/BlurFilter.cpp b/services/surfaceflinger/BlurFilter.cpp index 1ffbd5b..1ffbd5b 100644 --- a/libs/surfaceflinger/BlurFilter.cpp +++ b/services/surfaceflinger/BlurFilter.cpp diff --git a/libs/surfaceflinger/BlurFilter.h b/services/surfaceflinger/BlurFilter.h index 294db43..294db43 100644 --- a/libs/surfaceflinger/BlurFilter.h +++ b/services/surfaceflinger/BlurFilter.h diff --git a/libs/surfaceflinger/DisplayHardware/DisplayHardware.cpp b/services/surfaceflinger/DisplayHardware/DisplayHardware.cpp index 2eac0a8..2eac0a8 100644 --- a/libs/surfaceflinger/DisplayHardware/DisplayHardware.cpp +++ b/services/surfaceflinger/DisplayHardware/DisplayHardware.cpp diff --git a/libs/surfaceflinger/DisplayHardware/DisplayHardware.h b/services/surfaceflinger/DisplayHardware/DisplayHardware.h index 66bf521..66bf521 100644 --- a/libs/surfaceflinger/DisplayHardware/DisplayHardware.h +++ b/services/surfaceflinger/DisplayHardware/DisplayHardware.h diff --git a/libs/surfaceflinger/DisplayHardware/DisplayHardwareBase.cpp b/services/surfaceflinger/DisplayHardware/DisplayHardwareBase.cpp index 1d09f84..1d09f84 100644 --- a/libs/surfaceflinger/DisplayHardware/DisplayHardwareBase.cpp +++ b/services/surfaceflinger/DisplayHardware/DisplayHardwareBase.cpp diff --git a/libs/surfaceflinger/DisplayHardware/DisplayHardwareBase.h b/services/surfaceflinger/DisplayHardware/DisplayHardwareBase.h index 8369bb8..8369bb8 100644 --- a/libs/surfaceflinger/DisplayHardware/DisplayHardwareBase.h +++ b/services/surfaceflinger/DisplayHardware/DisplayHardwareBase.h diff --git a/libs/surfaceflinger/GLExtensions.cpp b/services/surfaceflinger/GLExtensions.cpp index 7f4f9fc..7f4f9fc 100644 --- a/libs/surfaceflinger/GLExtensions.cpp +++ b/services/surfaceflinger/GLExtensions.cpp diff --git a/libs/surfaceflinger/GLExtensions.h b/services/surfaceflinger/GLExtensions.h index bbb284e..bbb284e 100644 --- a/libs/surfaceflinger/GLExtensions.h +++ b/services/surfaceflinger/GLExtensions.h diff --git a/libs/surfaceflinger/Layer.cpp b/services/surfaceflinger/Layer.cpp index 758da4e..758da4e 100644 --- a/libs/surfaceflinger/Layer.cpp +++ b/services/surfaceflinger/Layer.cpp diff --git a/libs/surfaceflinger/Layer.h b/services/surfaceflinger/Layer.h index e1d283b..e1d283b 100644 --- a/libs/surfaceflinger/Layer.h +++ b/services/surfaceflinger/Layer.h diff --git a/libs/surfaceflinger/LayerBase.cpp b/services/surfaceflinger/LayerBase.cpp index d5aa53f..d5aa53f 100644 --- a/libs/surfaceflinger/LayerBase.cpp +++ b/services/surfaceflinger/LayerBase.cpp diff --git a/libs/surfaceflinger/LayerBase.h b/services/surfaceflinger/LayerBase.h index 4288cf7..4288cf7 100644 --- a/libs/surfaceflinger/LayerBase.h +++ b/services/surfaceflinger/LayerBase.h diff --git a/libs/surfaceflinger/LayerBlur.cpp b/services/surfaceflinger/LayerBlur.cpp index 64a43c7..64a43c7 100644 --- a/libs/surfaceflinger/LayerBlur.cpp +++ b/services/surfaceflinger/LayerBlur.cpp diff --git a/libs/surfaceflinger/LayerBlur.h b/services/surfaceflinger/LayerBlur.h index 4c9ec64..4c9ec64 100644 --- a/libs/surfaceflinger/LayerBlur.h +++ b/services/surfaceflinger/LayerBlur.h diff --git a/libs/surfaceflinger/LayerBuffer.cpp b/services/surfaceflinger/LayerBuffer.cpp index 5f83636..5f83636 100644 --- a/libs/surfaceflinger/LayerBuffer.cpp +++ b/services/surfaceflinger/LayerBuffer.cpp diff --git a/libs/surfaceflinger/LayerBuffer.h b/services/surfaceflinger/LayerBuffer.h index 1c0bf83..1c0bf83 100644 --- a/libs/surfaceflinger/LayerBuffer.h +++ b/services/surfaceflinger/LayerBuffer.h diff --git a/libs/surfaceflinger/LayerDim.cpp b/services/surfaceflinger/LayerDim.cpp index a1f339e..a1f339e 100644 --- a/libs/surfaceflinger/LayerDim.cpp +++ b/services/surfaceflinger/LayerDim.cpp diff --git a/libs/surfaceflinger/LayerDim.h b/services/surfaceflinger/LayerDim.h index f032314..f032314 100644 --- a/libs/surfaceflinger/LayerDim.h +++ b/services/surfaceflinger/LayerDim.h diff --git a/libs/surfaceflinger/MODULE_LICENSE_APACHE2 b/services/surfaceflinger/MODULE_LICENSE_APACHE2 index e69de29..e69de29 100644 --- a/libs/surfaceflinger/MODULE_LICENSE_APACHE2 +++ b/services/surfaceflinger/MODULE_LICENSE_APACHE2 diff --git a/libs/surfaceflinger/MessageQueue.cpp b/services/surfaceflinger/MessageQueue.cpp index d668e88..d668e88 100644 --- a/libs/surfaceflinger/MessageQueue.cpp +++ b/services/surfaceflinger/MessageQueue.cpp diff --git a/libs/surfaceflinger/MessageQueue.h b/services/surfaceflinger/MessageQueue.h index 890f809..890f809 100644 --- a/libs/surfaceflinger/MessageQueue.h +++ b/services/surfaceflinger/MessageQueue.h diff --git a/libs/surfaceflinger/SurfaceFlinger.cpp b/services/surfaceflinger/SurfaceFlinger.cpp index 68e8f19..68e8f19 100644 --- a/libs/surfaceflinger/SurfaceFlinger.cpp +++ b/services/surfaceflinger/SurfaceFlinger.cpp diff --git a/libs/surfaceflinger/SurfaceFlinger.h b/services/surfaceflinger/SurfaceFlinger.h index 0bfc170..0bfc170 100644 --- a/libs/surfaceflinger/SurfaceFlinger.h +++ b/services/surfaceflinger/SurfaceFlinger.h diff --git a/libs/surfaceflinger/TextureManager.cpp b/services/surfaceflinger/TextureManager.cpp index 3b326df..3b326df 100644 --- a/libs/surfaceflinger/TextureManager.cpp +++ b/services/surfaceflinger/TextureManager.cpp diff --git a/libs/surfaceflinger/TextureManager.h b/services/surfaceflinger/TextureManager.h index c7c14e7..c7c14e7 100644 --- a/libs/surfaceflinger/TextureManager.h +++ b/services/surfaceflinger/TextureManager.h diff --git a/libs/surfaceflinger/Transform.cpp b/services/surfaceflinger/Transform.cpp index 5e27cc9..5e27cc9 100644 --- a/libs/surfaceflinger/Transform.cpp +++ b/services/surfaceflinger/Transform.cpp diff --git a/libs/surfaceflinger/Transform.h b/services/surfaceflinger/Transform.h index 20fa11a..20fa11a 100644 --- a/libs/surfaceflinger/Transform.h +++ b/services/surfaceflinger/Transform.h diff --git a/libs/surfaceflinger/clz.cpp b/services/surfaceflinger/clz.cpp index 2456b86..2456b86 100644 --- a/libs/surfaceflinger/clz.cpp +++ b/services/surfaceflinger/clz.cpp diff --git a/libs/surfaceflinger/clz.h b/services/surfaceflinger/clz.h index 0ddf986..0ddf986 100644 --- a/libs/surfaceflinger/clz.h +++ b/services/surfaceflinger/clz.h diff --git a/libs/surfaceflinger/tests/Android.mk b/services/surfaceflinger/tests/Android.mk index 5053e7d..5053e7d 100644 --- a/libs/surfaceflinger/tests/Android.mk +++ b/services/surfaceflinger/tests/Android.mk diff --git a/libs/surfaceflinger/tests/overlays/Android.mk b/services/surfaceflinger/tests/overlays/Android.mk index 592b601..592b601 100644 --- a/libs/surfaceflinger/tests/overlays/Android.mk +++ b/services/surfaceflinger/tests/overlays/Android.mk diff --git a/libs/surfaceflinger/tests/overlays/overlays.cpp b/services/surfaceflinger/tests/overlays/overlays.cpp index c248a615..c248a615 100644 --- a/libs/surfaceflinger/tests/overlays/overlays.cpp +++ b/services/surfaceflinger/tests/overlays/overlays.cpp diff --git a/libs/surfaceflinger/tests/resize/Android.mk b/services/surfaceflinger/tests/resize/Android.mk index 24c2d01..24c2d01 100644 --- a/libs/surfaceflinger/tests/resize/Android.mk +++ b/services/surfaceflinger/tests/resize/Android.mk diff --git a/libs/surfaceflinger/tests/resize/resize.cpp b/services/surfaceflinger/tests/resize/resize.cpp index 127cca3..127cca3 100644 --- a/libs/surfaceflinger/tests/resize/resize.cpp +++ b/services/surfaceflinger/tests/resize/resize.cpp |