diff options
author | Dima Zavin <dima@android.com> | 2011-04-19 22:06:30 -0700 |
---|---|---|
committer | Dima Zavin <dima@android.com> | 2011-04-27 10:48:38 -0700 |
commit | 5e20a3dd44ec8a5e00b90f17ce412784068f1f14 (patch) | |
tree | 9ab7321ffcb99b20653ffb711a926eab261f4384 | |
parent | f076aa5594840baf70fd78a00d1152bd13dfb80c (diff) | |
download | frameworks_base-5e20a3dd44ec8a5e00b90f17ce412784068f1f14.zip frameworks_base-5e20a3dd44ec8a5e00b90f17ce412784068f1f14.tar.gz frameworks_base-5e20a3dd44ec8a5e00b90f17ce412784068f1f14.tar.bz2 |
audioflinger: move legacy audio hw/policy out to libhardware_legacy
Change-Id: I4adcec73d3c08bcbe15bb19e1ba2ff18b195af45
Signed-off-by: Dima Zavin <dima@android.com>
-rw-r--r-- | services/audioflinger/A2dpAudioInterface.cpp | 498 | ||||
-rw-r--r-- | services/audioflinger/A2dpAudioInterface.h | 138 | ||||
-rw-r--r-- | services/audioflinger/Android.mk | 87 | ||||
-rw-r--r-- | services/audioflinger/AudioDumpInterface.cpp | 573 | ||||
-rw-r--r-- | services/audioflinger/AudioDumpInterface.h | 170 | ||||
-rw-r--r-- | services/audioflinger/AudioHardwareGeneric.cpp | 411 | ||||
-rw-r--r-- | services/audioflinger/AudioHardwareGeneric.h | 151 | ||||
-rw-r--r-- | services/audioflinger/AudioHardwareInterface.cpp | 183 | ||||
-rw-r--r-- | services/audioflinger/AudioHardwareStub.cpp | 209 | ||||
-rw-r--r-- | services/audioflinger/AudioHardwareStub.h | 106 | ||||
-rw-r--r-- | services/audioflinger/AudioPolicyManagerBase.cpp | 2287 |
11 files changed, 0 insertions, 4813 deletions
diff --git a/services/audioflinger/A2dpAudioInterface.cpp b/services/audioflinger/A2dpAudioInterface.cpp deleted file mode 100644 index d926cb1..0000000 --- a/services/audioflinger/A2dpAudioInterface.cpp +++ /dev/null @@ -1,498 +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" -#include <hardware_legacy/power.h> - -namespace android { - -static const char *sA2dpWakeLock = "A2dpOutputStream"; -#define MAX_WRITE_RETRIES 5 - -// ---------------------------------------------------------------------------- - -//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; - mBufferDurationUs = ((bufferSize() * 1000 )/ frameSize() / sampleRate()) * 1000; - return NO_ERROR; -} - -A2dpAudioInterface::A2dpAudioStreamOut::~A2dpAudioStreamOut() -{ - LOGV("A2dpAudioStreamOut destructor"); - close(); - LOGV("A2dpAudioStreamOut destructor returning from close()"); -} - -ssize_t A2dpAudioInterface::A2dpAudioStreamOut::write(const void* buffer, size_t bytes) -{ - status_t status = -1; - { - Mutex::Autolock lock(mLock); - - size_t remaining = bytes; - - if (!mBluetoothEnabled || mClosing || mSuspended) { - LOGV("A2dpAudioStreamOut::write(), but bluetooth disabled \ - mBluetoothEnabled %d, mClosing %d, mSuspended %d", - mBluetoothEnabled, mClosing, mSuspended); - goto Error; - } - - if (mStandby) { - acquire_wake_lock (PARTIAL_WAKE_LOCK, sA2dpWakeLock); - mStandby = false; - mLastWriteTime = systemTime(); - } - - status = init(); - if (status < 0) - goto Error; - - int retries = MAX_WRITE_RETRIES; - while (remaining > 0 && retries) { - status = a2dp_write(mData, buffer, remaining); - if (status < 0) { - LOGE("a2dp_write failed err: %d\n", status); - goto Error; - } - if (status == 0) { - retries--; - } - remaining -= status; - buffer = (char *)buffer + status; - } - - // if A2DP sink runs abnormally fast, sleep a little so that audioflinger mixer thread - // does no spin and starve other threads. - // NOTE: It is likely that the A2DP headset is being disconnected - nsecs_t now = systemTime(); - if ((uint32_t)ns2us(now - mLastWriteTime) < (mBufferDurationUs >> 2)) { - LOGV("A2DP sink runs too fast"); - usleep(mBufferDurationUs - (uint32_t)ns2us(now - mLastWriteTime)); - } - mLastWriteTime = now; - return bytes; - - } -Error: - - standby(); - - // Simulate audio output timing in case of error - usleep(mBufferDurationUs); - - 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() -{ - Mutex::Autolock lock(mLock); - return standby_l(); -} - -status_t A2dpAudioInterface::A2dpAudioStreamOut::standby_l() -{ - int result = NO_ERROR; - - if (!mStandby) { - LOGV_IF(mClosing || !mBluetoothEnabled, "Standby skip stop: closing %d enabled %d", - mClosing, mBluetoothEnabled); - if (!mClosing && mBluetoothEnabled) { - result = a2dp_stop(mData); - } - release_wake_lock(sA2dpWakeLock); - 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"); - if (mClosing) { - standby(); - } - 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() -{ - standby_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/services/audioflinger/A2dpAudioInterface.h b/services/audioflinger/A2dpAudioInterface.h deleted file mode 100644 index dbe2c6a..0000000 --- a/services/audioflinger/A2dpAudioInterface.h +++ /dev/null @@ -1,138 +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); - status_t standby_l(); - - 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; - nsecs_t mLastWriteTime; - uint32_t mBufferDurationUs; - }; - - friend class A2dpAudioStreamOut; - - A2dpAudioStreamOut* mOutput; - AudioHardwareInterface *mHardwareInterface; - char mA2dpAddress[20]; - bool mBluetoothEnabled; - bool mSuspended; -}; - - -// ---------------------------------------------------------------------------- - -}; // namespace android - -#endif // A2DP_AUDIO_HARDWARE_H diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk index 69a4adc..6d78614 100644 --- a/services/audioflinger/Android.mk +++ b/services/audioflinger/Android.mk @@ -1,77 +1,5 @@ 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 $(CLEAR_VARS) LOCAL_SRC_FILES:= \ @@ -90,12 +18,6 @@ LOCAL_SHARED_LIBRARIES := \ libhardware_legacy \ libeffects -ifeq ($(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 @@ -105,15 +27,6 @@ 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 diff --git a/services/audioflinger/AudioDumpInterface.cpp b/services/audioflinger/AudioDumpInterface.cpp deleted file mode 100644 index 6c11114..0000000 --- a/services/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/services/audioflinger/AudioDumpInterface.h b/services/audioflinger/AudioDumpInterface.h deleted file mode 100644 index 814ce5f..0000000 --- a/services/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/services/audioflinger/AudioHardwareGeneric.cpp b/services/audioflinger/AudioHardwareGeneric.cpp deleted file mode 100644 index d63c031..0000000 --- a/services/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/services/audioflinger/AudioHardwareGeneric.h b/services/audioflinger/AudioHardwareGeneric.h deleted file mode 100644 index aa4e78d..0000000 --- a/services/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/services/audioflinger/AudioHardwareInterface.cpp b/services/audioflinger/AudioHardwareInterface.cpp deleted file mode 100644 index f58e4c0..0000000 --- a/services/audioflinger/AudioHardwareInterface.cpp +++ /dev/null @@ -1,183 +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", - "IN_COMMUNICATION" -}; - -static const char* routeNone = "NONE"; - -static const char* displayMode(int mode) -{ - if ((mode < AudioSystem::MODE_INVALID) || (mode >= AudioSystem::NUM_MODES)) - 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/services/audioflinger/AudioHardwareStub.cpp b/services/audioflinger/AudioHardwareStub.cpp deleted file mode 100644 index d481150..0000000 --- a/services/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/services/audioflinger/AudioHardwareStub.h b/services/audioflinger/AudioHardwareStub.h deleted file mode 100644 index 06a29de..0000000 --- a/services/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/services/audioflinger/AudioPolicyManagerBase.cpp b/services/audioflinger/AudioPolicyManagerBase.cpp deleted file mode 100644 index 32d92dc..0000000 --- a/services/audioflinger/AudioPolicyManagerBase.cpp +++ /dev/null @@ -1,2287 +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> -#include <math.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); - } - } - 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 = ""; - } - } - } 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 - checkA2dpSuspend(); - checkOutputForAllStrategies(); - // 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 (isInCall()) { - 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 (!isStateInCall(oldState) && isStateInCall(state)) { - 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 (isStateInCall(oldState) && !isStateInCall(state)) { - LOGV(" Exiting call in setPhoneState()"); - // force routing command to audio hardware when exiting a call - // even if no device change is needed - force = true; - } else if (isStateInCall(state) && (state != oldState)) { - LOGV(" Switching between telephony and VoIP in setPhoneState()"); - // force routing command to audio hardware when switching between telephony and VoIP - // 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 - checkA2dpSuspend(); - checkOutputForAllStrategies(); -#endif - updateDeviceForStrategy(); - - AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput); - - // force routing command to audio hardware when ending call - // even if no device change is needed - if (isStateInCall(oldState) && 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 (isStateInCall(state) && 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 (isStateInCall(state)) { - 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 && - isStreamActive(AudioSystem::MUSIC, 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; - } - forceVolumeReeval = true; - 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_ANALOG_DOCK && - config != AudioSystem::FORCE_DIGITAL_DOCK && 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 && - config != AudioSystem::FORCE_ANALOG_DOCK && - config != AudioSystem::FORCE_DIGITAL_DOCK) { - 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 - checkA2dpSuspend(); - checkOutputForAllStrategies(); -#endif - updateDeviceForStrategy(); - setOutputDevice(mHardwareOutput, newDevice); - if (forceVolumeReeval) { - applyStreamVolumes(mHardwareOutput, newDevice, 0, true); - } - - audio_io_handle_t activeInput = getActiveInput(); - if (activeInput != 0) { - AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput); - newDevice = getDeviceForInputSource(inputDesc->mInputSource); - if (newDevice != inputDesc->mDevice) { - LOGV("setForceUse() 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()); - } - } - -} - -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; - outputDesc->mStopTime[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, - int session) -{ - LOGV("startOutput() output %d, stream %d, session %d", output, stream, session); - 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 (isInCall()) { - 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, - int session) -{ - LOGV("stopOutput() output %d, stream %d, session %d", output, stream, session); - 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 (isInCall()) { - 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 stream was stopped - see isStreamActive() - outputDesc->mStopTime[stream] = systemTime(); - - setOutputDevice(output, getNewDevice(output), false, outputDesc->mLatency*2); - -#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); - - param.addInt(String8(AudioParameter::keyInputSource), (int)inputDesc->mInputSource); - LOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource); - - 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; -} - -audio_io_handle_t AudioPolicyManagerBase::getOutputForEffect(effect_descriptor_t *desc) -{ - LOGV("getOutputForEffect()"); - // apply simple rule where global effects are attached to the same output as MUSIC streams - return getOutput(AudioSystem::MUSIC); -} - -status_t AudioPolicyManagerBase::registerEffect(effect_descriptor_t *desc, - audio_io_handle_t output, - uint32_t strategy, - int session, - int id) -{ - ssize_t index = mOutputs.indexOfKey(output); - if (index < 0) { - LOGW("registerEffect() unknown output %d", output); - return INVALID_OPERATION; - } - - if (mTotalEffectsCpuLoad + desc->cpuLoad > getMaxEffectsCpuLoad()) { - LOGW("registerEffect() CPU Load limit exceeded for Fx %s, CPU %f MIPS", - desc->name, (float)desc->cpuLoad/10); - return INVALID_OPERATION; - } - if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) { - LOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB", - desc->name, desc->memoryUsage); - return INVALID_OPERATION; - } - mTotalEffectsCpuLoad += desc->cpuLoad; - mTotalEffectsMemory += desc->memoryUsage; - LOGV("registerEffect() effect %s, output %d, strategy %d session %d id %d", - desc->name, output, strategy, session, id); - - LOGV("registerEffect() CPU %d, memory %d", desc->cpuLoad, desc->memoryUsage); - LOGV(" total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory); - - EffectDescriptor *pDesc = new EffectDescriptor(); - memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t)); - pDesc->mOutput = output; - pDesc->mStrategy = (routing_strategy)strategy; - pDesc->mSession = session; - mEffects.add(id, pDesc); - - return NO_ERROR; -} - -status_t AudioPolicyManagerBase::unregisterEffect(int id) -{ - ssize_t index = mEffects.indexOfKey(id); - if (index < 0) { - LOGW("unregisterEffect() unknown effect ID %d", id); - return INVALID_OPERATION; - } - - EffectDescriptor *pDesc = mEffects.valueAt(index); - - if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) { - LOGW("unregisterEffect() CPU load %d too high for total %d", - pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad); - pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad; - } - mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad; - if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) { - LOGW("unregisterEffect() memory %d too big for total %d", - pDesc->mDesc.memoryUsage, mTotalEffectsMemory); - pDesc->mDesc.memoryUsage = mTotalEffectsMemory; - } - mTotalEffectsMemory -= pDesc->mDesc.memoryUsage; - LOGV("unregisterEffect() effect %s, ID %d, CPU %d, memory %d", - pDesc->mDesc.name, id, pDesc->mDesc.cpuLoad, pDesc->mDesc.memoryUsage); - LOGV(" total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory); - - mEffects.removeItem(id); - delete pDesc; - - return NO_ERROR; -} - -bool AudioPolicyManagerBase::isStreamActive(int stream, uint32_t inPastMs) const -{ - nsecs_t sysTime = systemTime(); - for (size_t i = 0; i < mOutputs.size(); i++) { - if (mOutputs.valueAt(i)->mRefCount[stream] != 0 || - ns2ms(sysTime - mOutputs.valueAt(i)->mStopTime[stream]) < inPastMs) { - return true; - } - } - return false; -} - - -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)); - } - - snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n", - (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory); - write(fd, buffer, strlen(buffer)); - - snprintf(buffer, SIZE, "Registered effects:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mEffects.size(); i++) { - snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i)); - write(fd, buffer, strlen(buffer)); - mEffects.valueAt(i)->dump(fd); - } - - - return NO_ERROR; -} - -// ---------------------------------------------------------------------------- -// AudioPolicyManagerBase -// ---------------------------------------------------------------------------- - -AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface) - : -#ifdef AUDIO_POLICY_TEST - Thread(false), -#endif //AUDIO_POLICY_TEST - mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0), - mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), - mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0), - mA2dpSuspended(false) -{ - mpClientInterface = clientInterface; - - for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) { - mForceUse[i] = AudioSystem::FORCE_NONE; - } - - initializeVolumeCurves(); - - // 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); - //TODO: configure audio effect output stage here - } - - updateDeviceForStrategy(); -#ifdef AUDIO_POLICY_TEST - if (mHardwareOutput != 0) { - 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(); -} - -status_t AudioPolicyManagerBase::initCheck() -{ - return (mHardwareOutput == 0) ? NO_INIT : NO_ERROR; -} - -#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); - - //TODO: configure audio effect output stage here - - // 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 (!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); - } - } - - mA2dpSuspended = false; - - 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 = ""; - mA2dpSuspended = false; - 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 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)); - audio_io_handle_t srcOutput = 0; - audio_io_handle_t dstOutput = 0; - - if (a2dpWasUsed && !a2dpIsUsed) { - bool dupUsed = a2dpUsedForSonification() && a2dpWasUsed && (AudioSystem::popCount(prevDevice) == 2); - dstOutput = mHardwareOutput; - if (dupUsed) { - LOGV("checkOutputForStrategy() moving strategy %d from duplicated", strategy); - srcOutput = mDuplicatedOutput; - } else { - LOGV("checkOutputForStrategy() moving strategy %d from a2dp", strategy); - srcOutput = mA2dpOutput; - } - } - if (a2dpIsUsed && !a2dpWasUsed) { - bool dupUsed = a2dpUsedForSonification() && a2dpIsUsed && (AudioSystem::popCount(curDevice) == 2); - srcOutput = mHardwareOutput; - if (dupUsed) { - LOGV("checkOutputForStrategy() moving strategy %d to duplicated", strategy); - dstOutput = mDuplicatedOutput; - } else { - LOGV("checkOutputForStrategy() moving strategy %d to a2dp", strategy); - dstOutput = mA2dpOutput; - } - } - - if (srcOutput != 0 && dstOutput != 0) { - // Move effects associated to this strategy from previous output to new output - for (size_t i = 0; i < mEffects.size(); i++) { - EffectDescriptor *desc = mEffects.valueAt(i); - if (desc->mSession != AudioSystem::SESSION_OUTPUT_STAGE && - desc->mStrategy == strategy && - desc->mOutput == srcOutput) { - LOGV("checkOutputForStrategy() moving effect %d to output %d", mEffects.keyAt(i), dstOutput); - mpClientInterface->moveEffects(desc->mSession, srcOutput, dstOutput); - desc->mOutput = dstOutput; - } - } - // Move tracks associated to this strategy from previous output to new output - for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) { - if (getStrategy((AudioSystem::stream_type)i) == strategy) { - mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, dstOutput); - } - } - } -} - -void AudioPolicyManagerBase::checkOutputForAllStrategies() -{ - checkOutputForStrategy(STRATEGY_PHONE); - checkOutputForStrategy(STRATEGY_SONIFICATION); - checkOutputForStrategy(STRATEGY_MEDIA); - checkOutputForStrategy(STRATEGY_DTMF); -} - -void AudioPolicyManagerBase::checkA2dpSuspend() -{ - // suspend A2DP output if: - // (NOT already suspended) && - // ((SCO device is connected && - // (forced usage for communication || for record is SCO))) || - // (phone state is ringing || in call) - // - // restore A2DP output if: - // (Already suspended) && - // ((SCO device is NOT connected || - // (forced usage NOT for communication && NOT for record is SCO))) && - // (phone state is NOT ringing && NOT in call) - // - if (mA2dpOutput == 0) { - return; - } - - if (mA2dpSuspended) { - if (((mScoDeviceAddress == "") || - ((mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO) && - (mForceUse[AudioSystem::FOR_RECORD] != AudioSystem::FORCE_BT_SCO))) && - ((mPhoneState != AudioSystem::MODE_IN_CALL) && - (mPhoneState != AudioSystem::MODE_RINGTONE))) { - - mpClientInterface->restoreOutput(mA2dpOutput); - mA2dpSuspended = false; - } - } else { - if (((mScoDeviceAddress != "") && - ((mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) || - (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO))) || - ((mPhoneState == AudioSystem::MODE_IN_CALL) || - (mPhoneState == AudioSystem::MODE_RINGTONE))) { - - mpClientInterface->suspendOutput(mA2dpOutput); - mA2dpSuspended = true; - } - } -} - - -#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 (isInCall() || - 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; -} - -uint32_t AudioPolicyManagerBase::getStrategyForStream(AudioSystem::stream_type stream) { - return (uint32_t)getStrategy(stream); -} - -uint32_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) { - uint32_t devices; - // By checking the range of stream before calling getStrategy, we avoid - // getStrategy's behavior for invalid streams. getStrategy would do a LOGE - // and then return STRATEGY_MEDIA, but we want to return the empty set. - if (stream < (AudioSystem::stream_type) 0 || stream >= AudioSystem::NUM_STREAM_TYPES) { - devices = 0; - } else { - AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream); - devices = getDeviceForStrategy(strategy, true); - } - return devices; -} - -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 (!isInCall()) { - // 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 (!isInCall() || 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 (!isInCall() && !mA2dpSuspended) { - 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_AUX_DIGITAL; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_EARPIECE; - if (device == 0) { - LOGE("getDeviceForStrategy() earpiece device not found"); - } - break; - - case AudioSystem::FORCE_SPEAKER: -#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 (!isInCall() && !mA2dpSuspended) { - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; - if (device) break; - } -#endif - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET; - if (device) break; - device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET; - if (device) break; - 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 (isInCall()) { - 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_WIRED_HEADPHONE; - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET; - } -#ifdef WITH_A2DP - if ((mA2dpOutput != 0) && !mA2dpSuspended && - (strategy != STRATEGY_SONIFICATION || a2dpUsedForSonification())) { - 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_AUX_DIGITAL; - } - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET; - } - if (device2 == 0) { - device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET; - } - 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); - } - - // 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); - - // 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: - case AUDIO_SOURCE_VOICE_COMMUNICATION: - 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::volIndexToAmpl(uint32_t device, const StreamDescriptor& streamDesc, - int indexInUi) { - // the volume index in the UI is relative to the min and max volume indices for this stream type - int nbSteps = 1 + streamDesc.mVolIndex[StreamDescriptor::VOLMAX] - - streamDesc.mVolIndex[StreamDescriptor::VOLMIN]; - int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) / - (streamDesc.mIndexMax - streamDesc.mIndexMin); - - // find what part of the curve this index volume belongs to, or if it's out of bounds - int segment = 0; - if (volIdx < streamDesc.mVolIndex[StreamDescriptor::VOLMIN]) { // out of bounds - return 0.0f; - } else if (volIdx < streamDesc.mVolIndex[StreamDescriptor::VOLKNEE1]) { - segment = 0; - } else if (volIdx < streamDesc.mVolIndex[StreamDescriptor::VOLKNEE2]) { - segment = 1; - } else if (volIdx <= streamDesc.mVolIndex[StreamDescriptor::VOLMAX]) { - segment = 2; - } else { // out of bounds - return 1.0f; - } - - // linear interpolation in the attenuation table in dB - float decibels = streamDesc.mVolDbAtt[segment] + - ((float)(volIdx - streamDesc.mVolIndex[segment])) * - ( (streamDesc.mVolDbAtt[segment+1] - streamDesc.mVolDbAtt[segment]) / - ((float)(streamDesc.mVolIndex[segment+1] - streamDesc.mVolIndex[segment])) ); - - float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 ) - - LOGV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f", - streamDesc.mVolIndex[segment], volIdx, streamDesc.mVolIndex[segment+1], - streamDesc.mVolDbAtt[segment], decibels, streamDesc.mVolDbAtt[segment+1], - amplification); - - return amplification; -} - -void AudioPolicyManagerBase::initializeVolumeCurves() { - // initialize the volume curves to a (-49.5 - 0 dB) attenuation in 0.5dB steps - for (int i=0 ; i< AudioSystem::NUM_STREAM_TYPES ; i++) { - mStreams[i].mVolIndex[StreamDescriptor::VOLMIN] = 1; - mStreams[i].mVolDbAtt[StreamDescriptor::VOLMIN] = -49.5f; - mStreams[i].mVolIndex[StreamDescriptor::VOLKNEE1] = 33; - mStreams[i].mVolDbAtt[StreamDescriptor::VOLKNEE1] = -33.5f; - mStreams[i].mVolIndex[StreamDescriptor::VOLKNEE2] = 66; - mStreams[i].mVolDbAtt[StreamDescriptor::VOLKNEE2] = -17.0f; - // here we use 100 steps to avoid rounding errors - // when computing the volume in volIndexToAmpl() - mStreams[i].mVolIndex[StreamDescriptor::VOLMAX] = 100; - mStreams[i].mVolDbAtt[StreamDescriptor::VOLMAX] = 0.0f; - } - - // Modification for music: more attenuation for lower volumes, finer steps at high volumes - mStreams[AudioSystem::MUSIC].mVolIndex[StreamDescriptor::VOLMIN] = 1; - mStreams[AudioSystem::MUSIC].mVolDbAtt[StreamDescriptor::VOLMIN] = -58.0f; - mStreams[AudioSystem::MUSIC].mVolIndex[StreamDescriptor::VOLKNEE1] = 20; - mStreams[AudioSystem::MUSIC].mVolDbAtt[StreamDescriptor::VOLKNEE1] = -40.0f; - mStreams[AudioSystem::MUSIC].mVolIndex[StreamDescriptor::VOLKNEE2] = 60; - mStreams[AudioSystem::MUSIC].mVolDbAtt[StreamDescriptor::VOLKNEE2] = -17.0f; - mStreams[AudioSystem::MUSIC].mVolIndex[StreamDescriptor::VOLMAX] = 100; - mStreams[AudioSystem::MUSIC].mVolDbAtt[StreamDescriptor::VOLMAX] = 0.0f; -} - -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(); - } - - // if volume is not 0 (not muted), force media volume to max on digital output - if (stream == AudioSystem::MUSIC && - index != mStreams[stream].mIndexMin && - device == AudioSystem::DEVICE_OUT_AUX_DIGITAL) { - return 1.0; - } - - volume = volIndexToAmpl(device, streamDesc, index); - - // 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) || - (stream == AudioSystem::SYSTEM)) && - 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); - // We actually change the volume if: - // - the float value returned by computeVolume() changed - // - the force flag is set - 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) { - // 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; - // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is - // enabled - if (stream == AudioSystem::BLUETOOTH_SCO) { - mpClientInterface->setStreamVolume(AudioSystem::VOICE_CALL, volume, output, delayMs); - } - } - - mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs); - } - - if (stream == AudioSystem::VOICE_CALL || - stream == AudioSystem::BLUETOOTH_SCO) { - float voiceVolume; - // Force voice volume to max for bluetooth SCO as volume is managed by the headset - if (stream == AudioSystem::VOICE_CALL) { - voiceVolume = (float)index/(float)mStreams[stream].mIndexMax; - } else { - voiceVolume = 1.0; - } - - if (voiceVolume != mLastVoiceVolume && output == mHardwareOutput) { - mpClientInterface->setVoiceVolume(voiceVolume, delayMs); - mLastVoiceVolume = voiceVolume; - } - } - - return NO_ERROR; -} - -void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output, uint32_t device, int delayMs, bool force) -{ - 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, force); - } -} - -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::isInCall() -{ - return isStateInCall(mPhoneState); -} - -bool AudioPolicyManagerBase::isStateInCall(int state) { - return ((state == AudioSystem::MODE_IN_CALL) || - (state == AudioSystem::MODE_IN_COMMUNICATION)); -} - -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))); -} - -uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad() -{ - return MAX_EFFECTS_CPU_LOAD; -} - -uint32_t AudioPolicyManagerBase::getMaxEffectsMemory() -{ - return MAX_EFFECTS_MEMORY; -} - -// --- 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; - mStopTime[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() delta %d, stream %d, refCount %d", delta, 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), - mInputSource(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); -} - -// --- EffectDescriptor class implementation - -status_t AudioPolicyManagerBase::EffectDescriptor::dump(int fd) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, " Output: %d\n", mOutput); - result.append(buffer); - snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy); - result.append(buffer); - snprintf(buffer, SIZE, " Session: %d\n", mSession); - result.append(buffer); - snprintf(buffer, SIZE, " Name: %s\n", mDesc.name); - result.append(buffer); - write(fd, result.string(), result.size()); - - return NO_ERROR; -} - - - -}; // namespace android |