diff options
64 files changed, 7473 insertions, 1082 deletions
diff --git a/CleanSpec.mk b/CleanSpec.mk index e6d9ebf..94aac5c 100644 --- a/CleanSpec.mk +++ b/CleanSpec.mk @@ -47,6 +47,8 @@ $(call add-clean-step, rm -rf $(PRODUCT_OUT)/obj/SHARED_LIBRARIES/libmedia_nativ $(call add-clean-step, rm -rf $(PRODUCT_OUT)/obj/lib/libmedia_native.so) $(call add-clean-step, rm -rf $(PRODUCT_OUT)/symbols/system/lib/libmedia_native.so) $(call add-clean-step, rm -rf $(PRODUCT_OUT)/system/lib/libmedia_native.so) +$(call add-clean-step, rm -rf $(PRODUCT_OUT)/obj/SHARED_LIBRARIES/libaudioflinger_intermediates) +$(call add-clean-step, rm -rf $(PRODUCT_OUT)/system/lib/libaudioflinger.so) # ************************************************ # NEWER CLEAN STEPS MUST BE AT THE END OF THE LIST # ************************************************ diff --git a/camera/Android.mk b/camera/Android.mk index e633450..369d0c5 100644 --- a/camera/Android.mk +++ b/camera/Android.mk @@ -1,3 +1,17 @@ +# Copyright 2010 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. + CAMERA_CLIENT_LOCAL_PATH:= $(call my-dir) include $(call all-subdir-makefiles) include $(CLEAR_VARS) @@ -21,6 +35,7 @@ LOCAL_SRC_FILES:= \ camera2/CaptureRequest.cpp \ ProCamera.cpp \ CameraBase.cpp \ + VendorTagDescriptor.cpp LOCAL_SHARED_LIBRARIES := \ libcutils \ @@ -34,6 +49,7 @@ LOCAL_SHARED_LIBRARIES := \ LOCAL_C_INCLUDES += \ system/media/camera/include \ + system/media/private/camera/include LOCAL_MODULE:= libcamera_client diff --git a/camera/ICameraService.cpp b/camera/ICameraService.cpp index 5fc89fb..b86651f 100644 --- a/camera/ICameraService.cpp +++ b/camera/ICameraService.cpp @@ -17,6 +17,7 @@ #define LOG_TAG "BpCameraService" #include <utils/Log.h> +#include <utils/Errors.h> #include <stdint.h> #include <sys/types.h> @@ -34,6 +35,7 @@ #include <camera/camera2/ICameraDeviceUser.h> #include <camera/camera2/ICameraDeviceCallbacks.h> #include <camera/CameraMetadata.h> +#include <camera/VendorTagDescriptor.h> namespace android { @@ -143,6 +145,24 @@ public: return result; } + // Get enumeration and description of vendor tags for camera + virtual status_t getCameraVendorTagDescriptor(/*out*/sp<VendorTagDescriptor>& desc) { + Parcel data, reply; + data.writeInterfaceToken(ICameraService::getInterfaceDescriptor()); + remote()->transact(BnCameraService::GET_CAMERA_VENDOR_TAG_DESCRIPTOR, data, &reply); + + if (readExceptionCode(reply)) return -EPROTO; + status_t result = reply.readInt32(); + + if (reply.readInt32() != 0) { + sp<VendorTagDescriptor> d; + if (VendorTagDescriptor::createFromParcel(&reply, /*out*/d) == OK) { + desc = d; + } + } + return result; + } + // connect to camera service (android.hardware.Camera) virtual status_t connect(const sp<ICameraClient>& cameraClient, int cameraId, const String16 &clientPackageName, int clientUid, @@ -275,6 +295,22 @@ status_t BnCameraService::onTransact( info.writeToParcel(reply); return NO_ERROR; } break; + case GET_CAMERA_VENDOR_TAG_DESCRIPTOR: { + CHECK_INTERFACE(ICameraService, data, reply); + sp<VendorTagDescriptor> d; + status_t result = getCameraVendorTagDescriptor(d); + reply->writeNoException(); + reply->writeInt32(result); + + // out-variables are after exception and return value + if (d == NULL) { + reply->writeInt32(0); + } else { + reply->writeInt32(1); // means the parcelable is included + d->writeToParcel(reply); + } + return NO_ERROR; + } break; case CONNECT: { CHECK_INTERFACE(ICameraService, data, reply); sp<ICameraClient> cameraClient = @@ -284,7 +320,7 @@ status_t BnCameraService::onTransact( int32_t clientUid = data.readInt32(); sp<ICamera> camera; status_t status = connect(cameraClient, cameraId, - clientName, clientUid, /*out*/ camera); + clientName, clientUid, /*out*/camera); reply->writeNoException(); reply->writeInt32(status); if (camera != NULL) { @@ -304,7 +340,7 @@ status_t BnCameraService::onTransact( int32_t clientUid = data.readInt32(); sp<IProCameraUser> camera; status_t status = connectPro(cameraClient, cameraId, - clientName, clientUid, /*out*/ camera); + clientName, clientUid, /*out*/camera); reply->writeNoException(); reply->writeInt32(status); if (camera != NULL) { @@ -324,7 +360,7 @@ status_t BnCameraService::onTransact( int32_t clientUid = data.readInt32(); sp<ICameraDeviceUser> camera; status_t status = connectDevice(cameraClient, cameraId, - clientName, clientUid, /*out*/ camera); + clientName, clientUid, /*out*/camera); reply->writeNoException(); reply->writeInt32(status); if (camera != NULL) { diff --git a/camera/VendorTagDescriptor.cpp b/camera/VendorTagDescriptor.cpp new file mode 100644 index 0000000..a0a6a51 --- /dev/null +++ b/camera/VendorTagDescriptor.cpp @@ -0,0 +1,319 @@ +/* + * Copyright (C) 2014 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 "VenderTagDescriptor" + +#include <binder/Parcel.h> +#include <utils/Errors.h> +#include <utils/Log.h> +#include <utils/Mutex.h> +#include <utils/Vector.h> +#include <system/camera_metadata.h> +#include <camera_metadata_hidden.h> + +#include "camera/VendorTagDescriptor.h" + +#include <string.h> + +namespace android { + +extern "C" { + +static int vendor_tag_descriptor_get_tag_count(const vendor_tag_ops_t* v); +static void vendor_tag_descriptor_get_all_tags(const vendor_tag_ops_t* v, uint32_t* tagArray); +static const char* vendor_tag_descriptor_get_section_name(const vendor_tag_ops_t* v, uint32_t tag); +static const char* vendor_tag_descriptor_get_tag_name(const vendor_tag_ops_t* v, uint32_t tag); +static int vendor_tag_descriptor_get_tag_type(const vendor_tag_ops_t* v, uint32_t tag); + +} /* extern "C" */ + + +static Mutex sLock; +static sp<VendorTagDescriptor> sGlobalVendorTagDescriptor; + +VendorTagDescriptor::VendorTagDescriptor() {} +VendorTagDescriptor::~VendorTagDescriptor() {} + +status_t VendorTagDescriptor::createDescriptorFromOps(const vendor_tag_ops_t* vOps, + /*out*/ + sp<VendorTagDescriptor>& descriptor) { + if (vOps == NULL) { + ALOGE("%s: vendor_tag_ops argument was NULL.", __FUNCTION__); + return BAD_VALUE; + } + + int tagCount = vOps->get_tag_count(vOps); + if (tagCount < 0 || tagCount > INT32_MAX) { + ALOGE("%s: tag count %d from vendor ops is invalid.", __FUNCTION__, tagCount); + return BAD_VALUE; + } + + Vector<uint32_t> tagArray; + LOG_ALWAYS_FATAL_IF(tagArray.resize(tagCount) != tagCount, + "%s: too many (%u) vendor tags defined.", __FUNCTION__, tagCount); + + vOps->get_all_tags(vOps, /*out*/tagArray.editArray()); + + sp<VendorTagDescriptor> desc = new VendorTagDescriptor(); + desc->mTagCount = tagCount; + + for (size_t i = 0; i < static_cast<size_t>(tagCount); ++i) { + uint32_t tag = tagArray[i]; + if (tag < CAMERA_METADATA_VENDOR_TAG_BOUNDARY) { + ALOGE("%s: vendor tag %d not in vendor tag section.", __FUNCTION__, tag); + return BAD_VALUE; + } + const char *tagName = vOps->get_tag_name(vOps, tag); + if (tagName == NULL) { + ALOGE("%s: no tag name defined for vendor tag %d.", __FUNCTION__, tag); + return BAD_VALUE; + } + desc->mTagToNameMap.add(tag, String8(tagName)); + const char *sectionName = vOps->get_section_name(vOps, tag); + if (sectionName == NULL) { + ALOGE("%s: no section name defined for vendor tag %d.", __FUNCTION__, tag); + return BAD_VALUE; + } + desc->mTagToSectionMap.add(tag, String8(sectionName)); + int tagType = vOps->get_tag_type(vOps, tag); + if (tagType < 0 || tagType >= NUM_TYPES) { + ALOGE("%s: tag type %d from vendor ops does not exist.", __FUNCTION__, tagType); + return BAD_VALUE; + } + desc->mTagToTypeMap.add(tag, tagType); + } + descriptor = desc; + return OK; +} + +status_t VendorTagDescriptor::createFromParcel(const Parcel* parcel, + /*out*/ + sp<VendorTagDescriptor>& descriptor) { + status_t res = OK; + if (parcel == NULL) { + ALOGE("%s: parcel argument was NULL.", __FUNCTION__); + return BAD_VALUE; + } + + int32_t tagCount = 0; + if ((res = parcel->readInt32(&tagCount)) != OK) { + ALOGE("%s: could not read tag count from parcel", __FUNCTION__); + return res; + } + + if (tagCount < 0 || tagCount > INT32_MAX) { + ALOGE("%s: tag count %d from vendor ops is invalid.", __FUNCTION__, tagCount); + return BAD_VALUE; + } + + sp<VendorTagDescriptor> desc = new VendorTagDescriptor(); + desc->mTagCount = tagCount; + + uint32_t tag; + int32_t tagType; + for (int32_t i = 0; i < tagCount; ++i) { + if ((res = parcel->readInt32(reinterpret_cast<int32_t*>(&tag))) != OK) { + ALOGE("%s: could not read tag id from parcel for index %d", __FUNCTION__, i); + break; + } + if (tag < CAMERA_METADATA_VENDOR_TAG_BOUNDARY) { + ALOGE("%s: vendor tag %d not in vendor tag section.", __FUNCTION__, tag); + res = BAD_VALUE; + break; + } + if ((res = parcel->readInt32(&tagType)) != OK) { + ALOGE("%s: could not read tag type from parcel for tag %d", __FUNCTION__, tag); + break; + } + if (tagType < 0 || tagType >= NUM_TYPES) { + ALOGE("%s: tag type %d from vendor ops does not exist.", __FUNCTION__, tagType); + res = BAD_VALUE; + break; + } + String8 tagName = parcel->readString8(); + if (tagName.isEmpty()) { + ALOGE("%s: parcel tag name was NULL for tag %d.", __FUNCTION__, tag); + res = NOT_ENOUGH_DATA; + break; + } + String8 sectionName = parcel->readString8(); + if (sectionName.isEmpty()) { + ALOGE("%s: parcel section name was NULL for tag %d.", __FUNCTION__, tag); + res = NOT_ENOUGH_DATA; + break; + } + + desc->mTagToNameMap.add(tag, tagName); + desc->mTagToSectionMap.add(tag, sectionName); + desc->mTagToTypeMap.add(tag, tagType); + } + + if (res != OK) { + return res; + } + + descriptor = desc; + return res; +} + +int VendorTagDescriptor::getTagCount() const { + size_t size = mTagToNameMap.size(); + if (size == 0) { + return VENDOR_TAG_COUNT_ERR; + } + return size; +} + +void VendorTagDescriptor::getTagArray(uint32_t* tagArray) const { + size_t size = mTagToNameMap.size(); + for (size_t i = 0; i < size; ++i) { + tagArray[i] = mTagToNameMap.keyAt(i); + } +} + +const char* VendorTagDescriptor::getSectionName(uint32_t tag) const { + ssize_t index = mTagToSectionMap.indexOfKey(tag); + if (index < 0) { + return VENDOR_SECTION_NAME_ERR; + } + return mTagToSectionMap.valueAt(index).string(); +} + +const char* VendorTagDescriptor::getTagName(uint32_t tag) const { + ssize_t index = mTagToNameMap.indexOfKey(tag); + if (index < 0) { + return VENDOR_TAG_NAME_ERR; + } + return mTagToNameMap.valueAt(index).string(); +} + +int VendorTagDescriptor::getTagType(uint32_t tag) const { + ssize_t index = mTagToNameMap.indexOfKey(tag); + if (index < 0) { + return VENDOR_TAG_TYPE_ERR; + } + return mTagToTypeMap.valueFor(tag); +} + +status_t VendorTagDescriptor::writeToParcel(Parcel* parcel) const { + status_t res = OK; + if (parcel == NULL) { + ALOGE("%s: parcel argument was NULL.", __FUNCTION__); + return BAD_VALUE; + } + + if ((res = parcel->writeInt32(mTagCount)) != OK) { + return res; + } + + size_t size = mTagToNameMap.size(); + uint32_t tag; + int32_t tagType; + for (size_t i = 0; i < size; ++i) { + tag = mTagToNameMap.keyAt(i); + String8 tagName = mTagToNameMap[i]; + String8 sectionName = mTagToSectionMap.valueFor(tag); + tagType = mTagToTypeMap.valueFor(tag); + if ((res = parcel->writeInt32(tag)) != OK) break; + if ((res = parcel->writeInt32(tagType)) != OK) break; + if ((res = parcel->writeString8(tagName)) != OK) break; + if ((res = parcel->writeString8(sectionName)) != OK) break; + } + + return res; +} + +status_t VendorTagDescriptor::setAsGlobalVendorTagDescriptor(const sp<VendorTagDescriptor>& desc) { + status_t res = OK; + Mutex::Autolock al(sLock); + sGlobalVendorTagDescriptor = desc; + + vendor_tag_ops_t* opsPtr = NULL; + if (desc != NULL) { + opsPtr = &(desc->mVendorOps); + opsPtr->get_tag_count = vendor_tag_descriptor_get_tag_count; + opsPtr->get_all_tags = vendor_tag_descriptor_get_all_tags; + opsPtr->get_section_name = vendor_tag_descriptor_get_section_name; + opsPtr->get_tag_name = vendor_tag_descriptor_get_tag_name; + opsPtr->get_tag_type = vendor_tag_descriptor_get_tag_type; + } + if((res = set_camera_metadata_vendor_ops(opsPtr)) != OK) { + ALOGE("%s: Could not set vendor tag descriptor, received error %s (%d)." + , __FUNCTION__, strerror(-res), res); + } + return res; +} + +void VendorTagDescriptor::clearGlobalVendorTagDescriptor() { + Mutex::Autolock al(sLock); + set_camera_metadata_vendor_ops(NULL); + sGlobalVendorTagDescriptor.clear(); +} + +sp<VendorTagDescriptor> VendorTagDescriptor::getGlobalVendorTagDescriptor() { + Mutex::Autolock al(sLock); + return sGlobalVendorTagDescriptor; +} + +extern "C" { + +int vendor_tag_descriptor_get_tag_count(const vendor_tag_ops_t* v) { + Mutex::Autolock al(sLock); + if (sGlobalVendorTagDescriptor == NULL) { + ALOGE("%s: Vendor tag descriptor not initialized.", __FUNCTION__); + return VENDOR_TAG_COUNT_ERR; + } + return sGlobalVendorTagDescriptor->getTagCount(); +} + +void vendor_tag_descriptor_get_all_tags(const vendor_tag_ops_t* v, uint32_t* tagArray) { + Mutex::Autolock al(sLock); + if (sGlobalVendorTagDescriptor == NULL) { + ALOGE("%s: Vendor tag descriptor not initialized.", __FUNCTION__); + return; + } + sGlobalVendorTagDescriptor->getTagArray(tagArray); +} + +const char* vendor_tag_descriptor_get_section_name(const vendor_tag_ops_t* v, uint32_t tag) { + Mutex::Autolock al(sLock); + if (sGlobalVendorTagDescriptor == NULL) { + ALOGE("%s: Vendor tag descriptor not initialized.", __FUNCTION__); + return VENDOR_SECTION_NAME_ERR; + } + return sGlobalVendorTagDescriptor->getSectionName(tag); +} + +const char* vendor_tag_descriptor_get_tag_name(const vendor_tag_ops_t* v, uint32_t tag) { + Mutex::Autolock al(sLock); + if (sGlobalVendorTagDescriptor == NULL) { + ALOGE("%s: Vendor tag descriptor not initialized.", __FUNCTION__); + return VENDOR_TAG_NAME_ERR; + } + return sGlobalVendorTagDescriptor->getTagName(tag); +} + +int vendor_tag_descriptor_get_tag_type(const vendor_tag_ops_t* v, uint32_t tag) { + Mutex::Autolock al(sLock); + if (sGlobalVendorTagDescriptor == NULL) { + ALOGE("%s: Vendor tag descriptor not initialized.", __FUNCTION__); + return VENDOR_TAG_TYPE_ERR; + } + return sGlobalVendorTagDescriptor->getTagType(tag); +} + +} /* extern "C" */ +} /* namespace android */ diff --git a/camera/tests/Android.mk b/camera/tests/Android.mk index ec13911..61385e5 100644 --- a/camera/tests/Android.mk +++ b/camera/tests/Android.mk @@ -1,9 +1,24 @@ +# Copyright 2013 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. + LOCAL_PATH:= $(call my-dir) include $(CLEAR_VARS) LOCAL_SRC_FILES:= \ main.cpp \ ProCameraTests.cpp \ + VendorTagDescriptorTests.cpp LOCAL_SHARED_LIBRARIES := \ libutils \ @@ -26,6 +41,8 @@ LOCAL_C_INCLUDES += \ external/gtest/include \ external/stlport/stlport \ system/media/camera/include \ + system/media/private/camera/include \ + system/media/camera/tests \ frameworks/av/services/camera/libcameraservice \ frameworks/av/include/camera \ frameworks/native/include \ diff --git a/camera/tests/VendorTagDescriptorTests.cpp b/camera/tests/VendorTagDescriptorTests.cpp new file mode 100644 index 0000000..6624e79 --- /dev/null +++ b/camera/tests/VendorTagDescriptorTests.cpp @@ -0,0 +1,204 @@ +/* + * Copyright (C) 2014 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_NDEBUG 0 +#define LOG_TAG "VendorTagDescriptorTests" + +#include <binder/Parcel.h> +#include <camera/VendorTagDescriptor.h> +#include <camera_metadata_tests_fake_vendor.h> +#include <camera_metadata_hidden.h> +#include <system/camera_vendor_tags.h> +#include <utils/Errors.h> +#include <utils/Log.h> +#include <utils/RefBase.h> + +#include <gtest/gtest.h> +#include <stdint.h> + +using namespace android; + +enum { + BAD_TAG_ARRAY = 0xDEADBEEFu, + BAD_TAG = 0x8DEADBADu, +}; + +#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) + +static bool ContainsTag(uint32_t* tagArray, size_t size, uint32_t tag) { + for (size_t i = 0; i < size; ++i) { + if (tag == tagArray[i]) return true; + } + return false; +} + +#define EXPECT_CONTAINS_TAG(t, a) \ + EXPECT_TRUE(ContainsTag(a, ARRAY_SIZE(a), t)) + +#define ASSERT_NOT_NULL(x) \ + ASSERT_TRUE((x) != NULL) + +extern "C" { + +static int default_get_tag_count(const vendor_tag_ops_t* vOps) { + return VENDOR_TAG_COUNT_ERR; +} + +static void default_get_all_tags(const vendor_tag_ops_t* vOps, uint32_t* tagArray) { + //Noop +} + +static const char* default_get_section_name(const vendor_tag_ops_t* vOps, uint32_t tag) { + return VENDOR_SECTION_NAME_ERR; +} + +static const char* default_get_tag_name(const vendor_tag_ops_t* vOps, uint32_t tag) { + return VENDOR_TAG_NAME_ERR; +} + +static int default_get_tag_type(const vendor_tag_ops_t* vOps, uint32_t tag) { + return VENDOR_TAG_TYPE_ERR; +} + +} /*extern "C"*/ + +// Set default vendor operations for a vendor_tag_ops struct +static void FillWithDefaults(vendor_tag_ops_t* vOps) { + ASSERT_NOT_NULL(vOps); + vOps->get_tag_count = default_get_tag_count; + vOps->get_all_tags = default_get_all_tags; + vOps->get_section_name = default_get_section_name; + vOps->get_tag_name = default_get_tag_name; + vOps->get_tag_type = default_get_tag_type; +} + +/** + * Test if values from VendorTagDescriptor methods match corresponding values + * from vendor_tag_ops functions. + */ +TEST(VendorTagDescriptorTest, ConsistentWithVendorTags) { + sp<VendorTagDescriptor> vDesc; + const vendor_tag_ops_t *vOps = &fakevendor_ops; + EXPECT_EQ(OK, VendorTagDescriptor::createDescriptorFromOps(vOps, /*out*/vDesc)); + + ASSERT_NOT_NULL(vDesc); + + // Ensure reasonable tag count + int tagCount = vDesc->getTagCount(); + EXPECT_EQ(tagCount, vOps->get_tag_count(vOps)); + + uint32_t descTagArray[tagCount]; + uint32_t opsTagArray[tagCount]; + + // Get all tag ids + vDesc->getTagArray(descTagArray); + vOps->get_all_tags(vOps, opsTagArray); + + ASSERT_NOT_NULL(descTagArray); + ASSERT_NOT_NULL(opsTagArray); + + uint32_t tag; + for (int i = 0; i < tagCount; ++i) { + // For each tag id, check whether type, section name, tag name match + tag = descTagArray[i]; + EXPECT_CONTAINS_TAG(tag, opsTagArray); + EXPECT_EQ(vDesc->getTagType(tag), vOps->get_tag_type(vOps, tag)); + EXPECT_STREQ(vDesc->getSectionName(tag), vOps->get_section_name(vOps, tag)); + EXPECT_STREQ(vDesc->getTagName(tag), vOps->get_tag_name(vOps, tag)); + } +} + +/** + * Test if values from VendorTagDescriptor methods stay consistent after being + * parcelled/unparcelled. + */ +TEST(VendorTagDescriptorTest, ConsistentAcrossParcel) { + sp<VendorTagDescriptor> vDescOriginal, vDescParceled; + const vendor_tag_ops_t *vOps = &fakevendor_ops; + EXPECT_EQ(OK, VendorTagDescriptor::createDescriptorFromOps(vOps, /*out*/vDescOriginal)); + + ASSERT_TRUE(vDescOriginal != NULL); + + Parcel p; + + // Check whether parcel read/write succeed + EXPECT_EQ(OK, vDescOriginal->writeToParcel(&p)); + p.setDataPosition(0); + ASSERT_EQ(OK, VendorTagDescriptor::createFromParcel(&p, vDescParceled)); + + // Ensure consistent tag count + int tagCount = vDescOriginal->getTagCount(); + ASSERT_EQ(tagCount, vDescParceled->getTagCount()); + + uint32_t descTagArray[tagCount]; + uint32_t desc2TagArray[tagCount]; + + // Get all tag ids + vDescOriginal->getTagArray(descTagArray); + vDescParceled->getTagArray(desc2TagArray); + + ASSERT_NOT_NULL(descTagArray); + ASSERT_NOT_NULL(desc2TagArray); + + uint32_t tag; + for (int i = 0; i < tagCount; ++i) { + // For each tag id, check consistency between the two vendor tag + // descriptors for each type, section name, tag name + tag = descTagArray[i]; + EXPECT_CONTAINS_TAG(tag, desc2TagArray); + EXPECT_EQ(vDescOriginal->getTagType(tag), vDescParceled->getTagType(tag)); + EXPECT_STREQ(vDescOriginal->getSectionName(tag), vDescParceled->getSectionName(tag)); + EXPECT_STREQ(vDescOriginal->getTagName(tag), vDescParceled->getTagName(tag)); + } +} + +/** + * Test defaults and error conditions. + */ +TEST(VendorTagDescriptorTest, ErrorConditions) { + sp<VendorTagDescriptor> vDesc; + vendor_tag_ops_t vOps; + FillWithDefaults(&vOps); + + // Ensure create fails when using null vOps + EXPECT_EQ(BAD_VALUE, VendorTagDescriptor::createDescriptorFromOps(/*vOps*/NULL, vDesc)); + + // Ensure create works when there are no vtags defined in a well-formed vOps + ASSERT_EQ(OK, VendorTagDescriptor::createDescriptorFromOps(&vOps, vDesc)); + + // Ensure defaults are returned when no vtags are defined, or tag is unknown + EXPECT_EQ(VENDOR_TAG_COUNT_ERR, vDesc->getTagCount()); + uint32_t* tagArray = reinterpret_cast<uint32_t*>(BAD_TAG_ARRAY); + uint32_t* testArray = tagArray; + vDesc->getTagArray(tagArray); + EXPECT_EQ(testArray, tagArray); + EXPECT_EQ(VENDOR_SECTION_NAME_ERR, vDesc->getSectionName(BAD_TAG)); + EXPECT_EQ(VENDOR_TAG_NAME_ERR, vDesc->getTagName(BAD_TAG)); + EXPECT_EQ(VENDOR_TAG_TYPE_ERR, vDesc->getTagType(BAD_TAG)); + + // Make sure global can be set/cleared + const vendor_tag_ops_t *fakeOps = &fakevendor_ops; + sp<VendorTagDescriptor> prevGlobal = VendorTagDescriptor::getGlobalVendorTagDescriptor(); + VendorTagDescriptor::clearGlobalVendorTagDescriptor(); + + EXPECT_TRUE(VendorTagDescriptor::getGlobalVendorTagDescriptor() == NULL); + EXPECT_EQ(OK, VendorTagDescriptor::setAsGlobalVendorTagDescriptor(vDesc)); + EXPECT_TRUE(VendorTagDescriptor::getGlobalVendorTagDescriptor() != NULL); + EXPECT_EQ(VENDOR_SECTION_NAME_ERR, vDesc->getSectionName(BAD_TAG)); + EXPECT_EQ(OK, VendorTagDescriptor::setAsGlobalVendorTagDescriptor(prevGlobal)); + EXPECT_EQ(prevGlobal, VendorTagDescriptor::getGlobalVendorTagDescriptor()); +} + diff --git a/cmds/screenrecord/Android.mk b/cmds/screenrecord/Android.mk index 6747e60..6ee2884 100644 --- a/cmds/screenrecord/Android.mk +++ b/cmds/screenrecord/Android.mk @@ -41,4 +41,6 @@ LOCAL_MODULE_TAGS := optional LOCAL_MODULE:= screenrecord +LOCAL_32_BIT_ONLY := true + include $(BUILD_EXECUTABLE) diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk index 561ce02..e2e389b 100644 --- a/cmds/stagefright/Android.mk +++ b/cmds/stagefright/Android.mk @@ -23,6 +23,8 @@ LOCAL_MODULE_TAGS := optional LOCAL_MODULE:= stagefright +LOCAL_32_BIT_ONLY := true + include $(BUILD_EXECUTABLE) ################################################################################ @@ -46,6 +48,8 @@ LOCAL_MODULE_TAGS := optional LOCAL_MODULE:= record +LOCAL_32_BIT_ONLY := true + include $(BUILD_EXECUTABLE) ################################################################################ @@ -69,6 +73,8 @@ LOCAL_MODULE_TAGS := optional LOCAL_MODULE:= recordvideo +LOCAL_32_BIT_ONLY := true + include $(BUILD_EXECUTABLE) @@ -93,6 +99,8 @@ LOCAL_MODULE_TAGS := optional LOCAL_MODULE:= audioloop +LOCAL_32_BIT_ONLY := true + include $(BUILD_EXECUTABLE) ################################################################################ @@ -116,6 +124,8 @@ LOCAL_MODULE_TAGS := optional LOCAL_MODULE:= stream +LOCAL_32_BIT_ONLY := true + include $(BUILD_EXECUTABLE) ################################################################################ @@ -139,6 +149,8 @@ LOCAL_MODULE_TAGS := optional LOCAL_MODULE:= sf2 +LOCAL_32_BIT_ONLY := true + include $(BUILD_EXECUTABLE) ################################################################################ @@ -163,6 +175,8 @@ LOCAL_MODULE_TAGS := optional LOCAL_MODULE:= codec +LOCAL_32_BIT_ONLY := true + include $(BUILD_EXECUTABLE) ################################################################################ @@ -186,4 +200,6 @@ LOCAL_MODULE_TAGS := optional LOCAL_MODULE:= muxer +LOCAL_32_BIT_ONLY := true + include $(BUILD_EXECUTABLE) diff --git a/drm/drmserver/Android.mk b/drm/drmserver/Android.mk index dc973da..aa0ab9b 100644 --- a/drm/drmserver/Android.mk +++ b/drm/drmserver/Android.mk @@ -39,4 +39,6 @@ LOCAL_MODULE:= drmserver LOCAL_MODULE_TAGS := optional +LOCAL_32_BIT_ONLY := true + include $(BUILD_EXECUTABLE) diff --git a/include/camera/ICameraService.h b/include/camera/ICameraService.h index f342122..6e48f22 100644 --- a/include/camera/ICameraService.h +++ b/include/camera/ICameraService.h @@ -31,6 +31,7 @@ class ICameraServiceListener; class ICameraDeviceUser; class ICameraDeviceCallbacks; class CameraMetadata; +class VendorTagDescriptor; class ICameraService : public IInterface { @@ -47,6 +48,7 @@ public: ADD_LISTENER, REMOVE_LISTENER, GET_CAMERA_CHARACTERISTICS, + GET_CAMERA_VENDOR_TAG_DESCRIPTOR, }; enum { @@ -58,10 +60,16 @@ public: virtual int32_t getNumberOfCameras() = 0; virtual status_t getCameraInfo(int cameraId, - struct CameraInfo* cameraInfo) = 0; + /*out*/ + struct CameraInfo* cameraInfo) = 0; virtual status_t getCameraCharacteristics(int cameraId, - CameraMetadata* cameraInfo) = 0; + /*out*/ + CameraMetadata* cameraInfo) = 0; + + virtual status_t getCameraVendorTagDescriptor( + /*out*/ + sp<VendorTagDescriptor>& desc) = 0; // Returns 'OK' if operation succeeded // - Errors: ALREADY_EXISTS if the listener was already added diff --git a/include/camera/VendorTagDescriptor.h b/include/camera/VendorTagDescriptor.h new file mode 100644 index 0000000..ea21d31 --- /dev/null +++ b/include/camera/VendorTagDescriptor.h @@ -0,0 +1,124 @@ +/* + * Copyright (C) 2014 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 VENDOR_TAG_DESCRIPTOR_H + +#include <utils/KeyedVector.h> +#include <utils/String8.h> +#include <utils/RefBase.h> +#include <system/camera_vendor_tags.h> + +#include <stdint.h> + +namespace android { + +class Parcel; + +/** + * VendorTagDescriptor objects are parcelable containers for the vendor tag + * definitions provided, and are typically used to pass the vendor tag + * information enumerated by the HAL to clients of the camera service. + */ +class VendorTagDescriptor + : public LightRefBase<VendorTagDescriptor> { + public: + virtual ~VendorTagDescriptor(); + + /** + * The following 'get*' methods implement the corresponding + * functions defined in + * system/media/camera/include/system/camera_vendor_tags.h + */ + + // Returns the number of vendor tags defined. + int getTagCount() const; + + // Returns an array containing the id's of vendor tags defined. + void getTagArray(uint32_t* tagArray) const; + + // Returns the section name string for a given vendor tag id. + const char* getSectionName(uint32_t tag) const; + + // Returns the tag name string for a given vendor tag id. + const char* getTagName(uint32_t tag) const; + + // Returns the tag type for a given vendor tag id. + int getTagType(uint32_t tag) const; + + /** + * Write the VendorTagDescriptor object into the given parcel. + * + * Returns OK on success, or a negative error code. + */ + status_t writeToParcel( + /*out*/ + Parcel* parcel) const; + + // Static methods: + + /** + * Create a VendorTagDescriptor object from the given parcel. + * + * Returns OK on success, or a negative error code. + */ + static status_t createFromParcel(const Parcel* parcel, + /*out*/ + sp<VendorTagDescriptor>& descriptor); + + /** + * Create a VendorTagDescriptor object from the given vendor_tag_ops_t + * struct. + * + * Returns OK on success, or a negative error code. + */ + static status_t createDescriptorFromOps(const vendor_tag_ops_t* vOps, + /*out*/ + sp<VendorTagDescriptor>& descriptor); + + /** + * Sets the global vendor tag descriptor to use for this process. + * Camera metadata operations that access vendor tags will use the + * vendor tag definitions set this way. + * + * Returns OK on success, or a negative error code. + */ + static status_t setAsGlobalVendorTagDescriptor(const sp<VendorTagDescriptor>& desc); + + /** + * Clears the global vendor tag descriptor used by this process. + */ + static void clearGlobalVendorTagDescriptor(); + + /** + * Returns the global vendor tag descriptor used by this process. + * This will contain NULL if no vendor tags are defined. + */ + static sp<VendorTagDescriptor> getGlobalVendorTagDescriptor(); + protected: + VendorTagDescriptor(); + KeyedVector<uint32_t, String8> mTagToNameMap; + KeyedVector<uint32_t, String8> mTagToSectionMap; + KeyedVector<uint32_t, int32_t> mTagToTypeMap; + // must be int32_t to be compatible with Parcel::writeInt32 + int32_t mTagCount; + private: + vendor_tag_ops mVendorOps; +}; + +} /* namespace android */ + +#define VENDOR_TAG_DESCRIPTOR_H +#endif /* VENDOR_TAG_DESCRIPTOR_H */ diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index 716eaa1..647748b 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -748,7 +748,6 @@ protected: sp<AudioTrackClientProxy> mProxy; // primary owner of the memory bool mInUnderrun; // whether track is currently in underrun state - String8 mName; // server's name for this IAudioTrack uint32_t mPausedPosition; private: diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h index 7c5f33a..9101f06 100644 --- a/include/media/IAudioFlinger.h +++ b/include/media/IAudioFlinger.h @@ -73,10 +73,6 @@ public: audio_io_handle_t output, pid_t tid, // -1 means unused, otherwise must be valid non-0 int *sessionId, - // input: ignored - // output: server's description of IAudioTrack for display in logs. - // Don't attempt to parse, as the format could change. - String8& name, int clientUid, status_t *status) = 0; diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h index 36f2a67..863a7d5 100644 --- a/include/media/stagefright/ACodec.h +++ b/include/media/stagefright/ACodec.h @@ -67,8 +67,6 @@ struct ACodec : public AHierarchicalStateMachine { void signalRequestIDRFrame(); - bool isConfiguredForAdaptivePlayback() { return mIsConfiguredForAdaptivePlayback; } - struct PortDescription : public RefBase { size_t countBuffers(); IOMX::buffer_id bufferIDAt(size_t index) const; @@ -178,6 +176,8 @@ private: sp<MemoryDealer> mDealer[2]; sp<ANativeWindow> mNativeWindow; + sp<AMessage> mInputFormat; + sp<AMessage> mOutputFormat; Vector<BufferInfo> mBuffers[2]; bool mPortEOS[2]; @@ -189,7 +189,6 @@ private: bool mIsEncoder; bool mUseMetadataOnEncoderOutput; bool mShutdownInProgress; - bool mIsConfiguredForAdaptivePlayback; // If "mKeepComponentAllocated" we only transition back to Loaded state // and do not release the component instance. @@ -203,6 +202,7 @@ private: unsigned mDequeueCounter; bool mStoreMetaDataInOutputBuffers; int32_t mMetaDataBuffersToSubmit; + size_t mNumUndequeuedBuffers; int64_t mRepeatFrameDelayUs; int64_t mMaxPtsGapUs; @@ -305,6 +305,7 @@ private: void processDeferredMessages(); void sendFormatChange(const sp<AMessage> &reply); + status_t getPortFormat(OMX_U32 portIndex, sp<AMessage> ¬ify); void signalError( OMX_ERRORTYPE error = OMX_ErrorUndefined, diff --git a/include/media/stagefright/MediaCodec.h b/include/media/stagefright/MediaCodec.h index 76aa503..276543b 100644 --- a/include/media/stagefright/MediaCodec.h +++ b/include/media/stagefright/MediaCodec.h @@ -106,6 +106,7 @@ struct MediaCodec : public AHandler { status_t signalEndOfInputStream(); status_t getOutputFormat(sp<AMessage> *format) const; + status_t getInputFormat(sp<AMessage> *format) const; status_t getInputBuffers(Vector<sp<ABuffer> > *buffers) const; status_t getOutputBuffers(Vector<sp<ABuffer> > *buffers) const; @@ -159,6 +160,7 @@ private: kWhatGetBuffers = 'getB', kWhatFlush = 'flus', kWhatGetOutputFormat = 'getO', + kWhatGetInputFormat = 'getI', kWhatDequeueInputTimedOut = 'dITO', kWhatDequeueOutputTimedOut = 'dOTO', kWhatCodecNotify = 'codc', @@ -199,6 +201,7 @@ private: sp<Surface> mNativeWindow; SoftwareRenderer *mSoftRenderer; sp<AMessage> mOutputFormat; + sp<AMessage> mInputFormat; List<size_t> mAvailPortBuffers[2]; Vector<BufferInfo> mPortBuffers[2]; diff --git a/libvideoeditor/lvpp/Android.mk b/libvideoeditor/lvpp/Android.mk index 860d351..77a21ac 100755 --- a/libvideoeditor/lvpp/Android.mk +++ b/libvideoeditor/lvpp/Android.mk @@ -99,6 +99,8 @@ LOCAL_CFLAGS += -Wno-multichar \ -DUSE_STAGEFRIGHT_READERS \ -DUSE_STAGEFRIGHT_3GPP_READER +LOCAL_32_BIT_ONLY := true + include $(BUILD_SHARED_LIBRARY) #include $(call all-makefiles-under,$(LOCAL_PATH)) diff --git a/libvideoeditor/vss/src/Android.mk b/libvideoeditor/vss/src/Android.mk index 0caa15b..8856c41 100755 --- a/libvideoeditor/vss/src/Android.mk +++ b/libvideoeditor/vss/src/Android.mk @@ -96,4 +96,6 @@ LOCAL_CFLAGS += -Wno-multichar \ -DM4xVSS_RESERVED_MOOV_DISK_SPACEno \ -DDECODE_GIF_ON_SAVING +LOCAL_32_BIT_ONLY := true + include $(BUILD_SHARED_LIBRARY) diff --git a/libvideoeditor/vss/stagefrightshells/src/Android.mk b/libvideoeditor/vss/stagefrightshells/src/Android.mk index 9188942..a060c0d 100755 --- a/libvideoeditor/vss/stagefrightshells/src/Android.mk +++ b/libvideoeditor/vss/stagefrightshells/src/Android.mk @@ -64,4 +64,6 @@ LOCAL_MODULE:= libvideoeditor_stagefrightshells LOCAL_MODULE_TAGS := optional +LOCAL_32_BIT_ONLY := true + include $(BUILD_STATIC_LIBRARY) diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index 60ed626..20c1cdb 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -1024,7 +1024,6 @@ status_t AudioTrack::createTrack_l(size_t epoch) output, tid, &mSessionId, - mName, mClientUid, &status); @@ -1281,8 +1280,7 @@ void AudioTrack::releaseBuffer(Buffer* audioBuffer) if (mState == STATE_ACTIVE) { audio_track_cblk_t* cblk = mCblk; if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { - ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", - this, mName.string()); + ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); // FIXME ignoring status mAudioTrack->start(); } diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp index a9a9f1a..762681e 100644 --- a/media/libmedia/IAudioFlinger.cpp +++ b/media/libmedia/IAudioFlinger.cpp @@ -95,7 +95,6 @@ public: audio_io_handle_t output, pid_t tid, int *sessionId, - String8& name, int clientUid, status_t *status) { @@ -140,7 +139,6 @@ public: if (sessionId != NULL) { *sessionId = lSessionId; } - name = reply.readString8(); lStatus = reply.readInt32(); track = interface_cast<IAudioTrack>(reply.readStrongBinder()); if (lStatus == NO_ERROR) { @@ -808,7 +806,6 @@ status_t BnAudioFlinger::onTransact( pid_t tid = (pid_t) data.readInt32(); int sessionId = data.readInt32(); int clientUid = data.readInt32(); - String8 name; status_t status; sp<IAudioTrack> track; if ((haveSharedBuffer && (buffer == 0)) || @@ -819,13 +816,12 @@ status_t BnAudioFlinger::onTransact( track = createTrack( (audio_stream_type_t) streamType, sampleRate, format, channelMask, &frameCount, &flags, buffer, output, tid, - &sessionId, name, clientUid, &status); + &sessionId, clientUid, &status); LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR)); } reply->writeInt32(frameCount); reply->writeInt32(flags); reply->writeInt32(sessionId); - reply->writeString8(name); reply->writeInt32(status); reply->writeStrongBinder(track->asBinder()); return NO_ERROR; diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk index 4189a5e..caf2dfc 100644 --- a/media/libmediaplayerservice/Android.mk +++ b/media/libmediaplayerservice/Android.mk @@ -53,6 +53,8 @@ LOCAL_C_INCLUDES := \ LOCAL_MODULE:= libmediaplayerservice +LOCAL_32_BIT_ONLY := true + include $(BUILD_SHARED_LIBRARY) include $(call all-makefiles-under,$(LOCAL_PATH)) diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp index a750ad0..d8d939a 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp @@ -31,13 +31,10 @@ #include "ATSParser.h" -#include "SoftwareRenderer.h" - #include <media/stagefright/foundation/hexdump.h> #include <media/stagefright/foundation/ABuffer.h> #include <media/stagefright/foundation/ADebug.h> #include <media/stagefright/foundation/AMessage.h> -#include <media/stagefright/ACodec.h> #include <media/stagefright/MediaDefs.h> #include <media/stagefright/MediaErrors.h> #include <media/stagefright/MetaData.h> @@ -146,7 +143,6 @@ NuPlayer::NuPlayer() : mUIDValid(false), mSourceFlags(0), mVideoIsAVC(false), - mNeedsSwRenderer(false), mAudioEOS(false), mVideoEOS(false), mScanSourcesPending(false), @@ -442,7 +438,6 @@ void NuPlayer::onMessageReceived(const sp<AMessage> &msg) { ALOGV("kWhatStart"); mVideoIsAVC = false; - mNeedsSwRenderer = false; mAudioEOS = false; mVideoEOS = false; mSkipRenderingAudioUntilMediaTimeUs = -1; @@ -533,24 +528,21 @@ void NuPlayer::onMessageReceived(const sp<AMessage> &msg) { { bool audio = msg->what() == kWhatAudioNotify; - sp<AMessage> codecRequest; - CHECK(msg->findMessage("codec-request", &codecRequest)); - int32_t what; - CHECK(codecRequest->findInt32("what", &what)); + CHECK(msg->findInt32("what", &what)); - if (what == ACodec::kWhatFillThisBuffer) { + if (what == Decoder::kWhatFillThisBuffer) { status_t err = feedDecoderInputData( - audio, codecRequest); + audio, msg); if (err == -EWOULDBLOCK) { if (mSource->feedMoreTSData() == OK) { msg->post(10000ll); } } - } else if (what == ACodec::kWhatEOS) { + } else if (what == Decoder::kWhatEOS) { int32_t err; - CHECK(codecRequest->findInt32("err", &err)); + CHECK(msg->findInt32("err", &err)); if (err == ERROR_END_OF_STREAM) { ALOGV("got %s decoder EOS", audio ? "audio" : "video"); @@ -561,7 +553,7 @@ void NuPlayer::onMessageReceived(const sp<AMessage> &msg) { } mRenderer->queueEOS(audio, err); - } else if (what == ACodec::kWhatFlushCompleted) { + } else if (what == Decoder::kWhatFlushCompleted) { bool needShutdown; if (audio) { @@ -590,14 +582,17 @@ void NuPlayer::onMessageReceived(const sp<AMessage> &msg) { } finishFlushIfPossible(); - } else if (what == ACodec::kWhatOutputFormatChanged) { + } else if (what == Decoder::kWhatOutputFormatChanged) { + sp<AMessage> format; + CHECK(msg->findMessage("format", &format)); + if (audio) { int32_t numChannels; - CHECK(codecRequest->findInt32( + CHECK(format->findInt32( "channel-count", &numChannels)); int32_t sampleRate; - CHECK(codecRequest->findInt32("sample-rate", &sampleRate)); + CHECK(format->findInt32("sample-rate", &sampleRate)); ALOGV("Audio output format changed to %d Hz, %d channels", sampleRate, numChannels); @@ -621,7 +616,7 @@ void NuPlayer::onMessageReceived(const sp<AMessage> &msg) { } int32_t channelMask; - if (!codecRequest->findInt32("channel-mask", &channelMask)) { + if (!format->findInt32("channel-mask", &channelMask)) { channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER; } @@ -642,11 +637,11 @@ void NuPlayer::onMessageReceived(const sp<AMessage> &msg) { // video int32_t width, height; - CHECK(codecRequest->findInt32("width", &width)); - CHECK(codecRequest->findInt32("height", &height)); + CHECK(format->findInt32("width", &width)); + CHECK(format->findInt32("height", &height)); int32_t cropLeft, cropTop, cropRight, cropBottom; - CHECK(codecRequest->findRect( + CHECK(format->findRect( "crop", &cropLeft, &cropTop, &cropRight, &cropBottom)); @@ -679,22 +674,8 @@ void NuPlayer::onMessageReceived(const sp<AMessage> &msg) { notifyListener( MEDIA_SET_VIDEO_SIZE, displayWidth, displayHeight); - - if (mNeedsSwRenderer && mNativeWindow != NULL) { - int32_t colorFormat; - CHECK(codecRequest->findInt32("color-format", &colorFormat)); - - sp<MetaData> meta = new MetaData; - meta->setInt32(kKeyWidth, width); - meta->setInt32(kKeyHeight, height); - meta->setRect(kKeyCropRect, cropLeft, cropTop, cropRight, cropBottom); - meta->setInt32(kKeyColorFormat, colorFormat); - - mRenderer->setSoftRenderer( - new SoftwareRenderer(mNativeWindow->getNativeWindow(), meta)); - } } - } else if (what == ACodec::kWhatShutdownCompleted) { + } else if (what == Decoder::kWhatShutdownCompleted) { ALOGV("%s shutdown completed", audio ? "audio" : "video"); if (audio) { mAudioDecoder.clear(); @@ -709,22 +690,15 @@ void NuPlayer::onMessageReceived(const sp<AMessage> &msg) { } finishFlushIfPossible(); - } else if (what == ACodec::kWhatError) { + } else if (what == Decoder::kWhatError) { ALOGE("Received error from %s decoder, aborting playback.", audio ? "audio" : "video"); mRenderer->queueEOS(audio, UNKNOWN_ERROR); - } else if (what == ACodec::kWhatDrainThisBuffer) { - renderBuffer(audio, codecRequest); - } else if (what == ACodec::kWhatComponentAllocated) { - if (!audio) { - AString name; - CHECK(codecRequest->findString("componentName", &name)); - mNeedsSwRenderer = name.startsWith("OMX.google."); - } - } else if (what != ACodec::kWhatComponentConfigured - && what != ACodec::kWhatBuffersAllocated) { - ALOGV("Unhandled codec notification %d '%c%c%c%c'.", + } else if (what == Decoder::kWhatDrainThisBuffer) { + renderBuffer(audio, msg); + } else { + ALOGV("Unhandled decoder notification %d '%c%c%c%c'.", what, what >> 24, (what >> 16) & 0xff, @@ -925,8 +899,7 @@ status_t NuPlayer::instantiateDecoder(bool audio, sp<Decoder> *decoder) { *decoder = audio ? new Decoder(notify) : new Decoder(notify, mNativeWindow); - looper()->registerHandler(*decoder); - + (*decoder)->init(); (*decoder)->configure(format); return OK; diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h index 9dfe4a0..f1d3d55 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayer.h +++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h @@ -24,7 +24,6 @@ namespace android { -struct ACodec; struct MetaData; struct NuPlayerDriver; @@ -118,7 +117,6 @@ private: sp<MediaPlayerBase::AudioSink> mAudioSink; sp<Decoder> mVideoDecoder; bool mVideoIsAVC; - bool mNeedsSwRenderer; sp<Decoder> mAudioDecoder; sp<Renderer> mRenderer; diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp index 2423fd5..469c9ca 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp @@ -17,14 +17,17 @@ //#define LOG_NDEBUG 0 #define LOG_TAG "NuPlayerDecoder" #include <utils/Log.h> +#include <inttypes.h> #include "NuPlayerDecoder.h" +#include <media/ICrypto.h> #include <media/stagefright/foundation/ABuffer.h> #include <media/stagefright/foundation/ADebug.h> #include <media/stagefright/foundation/AMessage.h> -#include <media/stagefright/ACodec.h> +#include <media/stagefright/MediaCodec.h> #include <media/stagefright/MediaDefs.h> +#include <media/stagefright/MediaErrors.h> namespace android { @@ -32,122 +35,425 @@ NuPlayer::Decoder::Decoder( const sp<AMessage> ¬ify, const sp<NativeWindowWrapper> &nativeWindow) : mNotify(notify), - mNativeWindow(nativeWindow) { + mNativeWindow(nativeWindow), + mBufferGeneration(0), + mComponentName("decoder") { + // Every decoder has its own looper because MediaCodec operations + // are blocking, but NuPlayer needs asynchronous operations. + mDecoderLooper = new ALooper; + mDecoderLooper->setName("NuPlayerDecoder"); + mDecoderLooper->start(false, false, ANDROID_PRIORITY_AUDIO); + + mCodecLooper = new ALooper; + mCodecLooper->setName("NuPlayerDecoder-MC"); + mCodecLooper->start(false, false, ANDROID_PRIORITY_AUDIO); } NuPlayer::Decoder::~Decoder() { } -void NuPlayer::Decoder::configure(const sp<AMessage> &format) { +void NuPlayer::Decoder::onConfigure(const sp<AMessage> &format) { CHECK(mCodec == NULL); + ++mBufferGeneration; + AString mime; CHECK(format->findString("mime", &mime)); - sp<AMessage> notifyMsg = - new AMessage(kWhatCodecNotify, id()); + sp<Surface> surface = NULL; + if (mNativeWindow != NULL) { + surface = mNativeWindow->getSurfaceTextureClient(); + } - mCSDIndex = 0; - for (size_t i = 0;; ++i) { - sp<ABuffer> csd; - if (!format->findBuffer(StringPrintf("csd-%d", i).c_str(), &csd)) { - break; - } + mComponentName = mime; + mComponentName.append(" decoder"); + ALOGV("[%s] onConfigure (surface=%p)", mComponentName.c_str(), surface.get()); - mCSD.push(csd); + mCodec = MediaCodec::CreateByType(mCodecLooper, mime.c_str(), false /* encoder */); + if (mCodec == NULL) { + ALOGE("Failed to create %s decoder", mime.c_str()); + handleError(UNKNOWN_ERROR); + return; } + mCodec->getName(&mComponentName); + if (mNativeWindow != NULL) { - format->setObject("native-window", mNativeWindow); + // disconnect from surface as MediaCodec will reconnect + CHECK_EQ((int)NO_ERROR, + native_window_api_disconnect( + surface.get(), + NATIVE_WINDOW_API_MEDIA)); + } + status_t err = mCodec->configure( + format, surface, NULL /* crypto */, 0 /* flags */); + if (err != OK) { + ALOGE("Failed to configure %s decoder (err=%d)", mComponentName.c_str(), err); + handleError(err); + return; + } + // the following should work in configured state + CHECK_EQ((status_t)OK, mCodec->getOutputFormat(&mOutputFormat)); + CHECK_EQ((status_t)OK, mCodec->getInputFormat(&mInputFormat)); + + err = mCodec->start(); + if (err != OK) { + ALOGE("Failed to start %s decoder (err=%d)", mComponentName.c_str(), err); + handleError(err); + return; } - // Current video decoders do not return from OMX_FillThisBuffer - // quickly, violating the OpenMAX specs, until that is remedied - // we need to invest in an extra looper to free the main event - // queue. - bool needDedicatedLooper = !strncasecmp(mime.c_str(), "video/", 6); + // the following should work after start + CHECK_EQ((status_t)OK, mCodec->getInputBuffers(&mInputBuffers)); + CHECK_EQ((status_t)OK, mCodec->getOutputBuffers(&mOutputBuffers)); + ALOGV("[%s] got %zu input and %zu output buffers", + mComponentName.c_str(), + mInputBuffers.size(), + mOutputBuffers.size()); - mFormat = format; - mCodec = new ACodec; + requestCodecNotification(); +} - if (needDedicatedLooper && mCodecLooper == NULL) { - mCodecLooper = new ALooper; - mCodecLooper->setName("NuPlayerDecoder"); - mCodecLooper->start(false, false, ANDROID_PRIORITY_AUDIO); +void NuPlayer::Decoder::requestCodecNotification() { + if (mCodec != NULL) { + sp<AMessage> reply = new AMessage(kWhatCodecNotify, id()); + reply->setInt32("generation", mBufferGeneration); + mCodec->requestActivityNotification(reply); } +} - (needDedicatedLooper ? mCodecLooper : looper())->registerHandler(mCodec); +bool NuPlayer::Decoder::isStaleReply(const sp<AMessage> &msg) { + int32_t generation; + CHECK(msg->findInt32("generation", &generation)); + return generation != mBufferGeneration; +} - mCodec->setNotificationMessage(notifyMsg); - mCodec->initiateSetup(format); +void NuPlayer::Decoder::init() { + mDecoderLooper->registerHandler(this); } -void NuPlayer::Decoder::onMessageReceived(const sp<AMessage> &msg) { - switch (msg->what()) { - case kWhatCodecNotify: - { - int32_t what; - CHECK(msg->findInt32("what", &what)); - - if (what == ACodec::kWhatFillThisBuffer) { - onFillThisBuffer(msg); - } else { - sp<AMessage> notify = mNotify->dup(); - notify->setMessage("codec-request", msg); - notify->post(); - } - break; +void NuPlayer::Decoder::configure(const sp<AMessage> &format) { + sp<AMessage> msg = new AMessage(kWhatConfigure, id()); + msg->setMessage("format", format); + msg->post(); +} + +void NuPlayer::Decoder::handleError(int32_t err) +{ + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatError); + notify->setInt32("err", err); + notify->post(); +} + +bool NuPlayer::Decoder::handleAnInputBuffer() { + size_t bufferIx = -1; + status_t res = mCodec->dequeueInputBuffer(&bufferIx); + ALOGV("[%s] dequeued input: %d", + mComponentName.c_str(), res == OK ? (int)bufferIx : res); + if (res != OK) { + if (res != -EAGAIN) { + handleError(res); } + return false; + } - default: - TRESPASS(); - break; + CHECK_LT(bufferIx, mInputBuffers.size()); + + sp<AMessage> reply = new AMessage(kWhatInputBufferFilled, id()); + reply->setSize("buffer-ix", bufferIx); + reply->setInt32("generation", mBufferGeneration); + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatFillThisBuffer); + notify->setBuffer("buffer", mInputBuffers[bufferIx]); + notify->setMessage("reply", reply); + notify->post(); + return true; +} + +void android::NuPlayer::Decoder::onInputBufferFilled(const sp<AMessage> &msg) { + size_t bufferIx; + CHECK(msg->findSize("buffer-ix", &bufferIx)); + CHECK_LT(bufferIx, mInputBuffers.size()); + sp<ABuffer> codecBuffer = mInputBuffers[bufferIx]; + + sp<ABuffer> buffer; + bool hasBuffer = msg->findBuffer("buffer", &buffer); + if (buffer == NULL /* includes !hasBuffer */) { + int32_t streamErr = ERROR_END_OF_STREAM; + CHECK(msg->findInt32("err", &streamErr) || !hasBuffer); + + if (streamErr == OK) { + /* buffers are returned to hold on to */ + return; + } + + // attempt to queue EOS + status_t err = mCodec->queueInputBuffer( + bufferIx, + 0, + 0, + 0, + MediaCodec::BUFFER_FLAG_EOS); + if (streamErr == ERROR_END_OF_STREAM && err != OK) { + streamErr = err; + // err will not be ERROR_END_OF_STREAM + } + + if (streamErr != ERROR_END_OF_STREAM) { + handleError(streamErr); + } + } else { + int64_t timeUs = 0; + uint32_t flags = 0; + CHECK(buffer->meta()->findInt64("timeUs", &timeUs)); + + int32_t eos; + // we do not expect CODECCONFIG or SYNCFRAME for decoder + if (buffer->meta()->findInt32("eos", &eos) && eos) { + flags |= MediaCodec::BUFFER_FLAG_EOS; + } + + // copy into codec buffer + if (buffer != codecBuffer) { + CHECK_LE(buffer->size(), codecBuffer->capacity()); + codecBuffer->setRange(0, buffer->size()); + memcpy(codecBuffer->data(), buffer->data(), buffer->size()); + } + + status_t err = mCodec->queueInputBuffer( + bufferIx, + codecBuffer->offset(), + codecBuffer->size(), + timeUs, + flags); + if (err != OK) { + ALOGE("Failed to queue input buffer for %s (err=%d)", + mComponentName.c_str(), err); + handleError(err); + } } } -void NuPlayer::Decoder::onFillThisBuffer(const sp<AMessage> &msg) { - sp<AMessage> reply; - CHECK(msg->findMessage("reply", &reply)); +bool NuPlayer::Decoder::handleAnOutputBuffer() { + size_t bufferIx = -1; + size_t offset; + size_t size; + int64_t timeUs; + uint32_t flags; + status_t res = mCodec->dequeueOutputBuffer( + &bufferIx, &offset, &size, &timeUs, &flags); + + if (res != OK) { + ALOGV("[%s] dequeued output: %d", mComponentName.c_str(), res); + } else { + ALOGV("[%s] dequeued output: %d (time=%lld flags=%" PRIu32 ")", + mComponentName.c_str(), (int)bufferIx, timeUs, flags); + } -#if 0 - sp<ABuffer> outBuffer; - CHECK(msg->findBuffer("buffer", &outBuffer)); -#else - sp<ABuffer> outBuffer; -#endif + if (res == INFO_OUTPUT_BUFFERS_CHANGED) { + res = mCodec->getOutputBuffers(&mOutputBuffers); + if (res != OK) { + ALOGE("Failed to get output buffers for %s after INFO event (err=%d)", + mComponentName.c_str(), res); + handleError(res); + return false; + } + // NuPlayer ignores this + return true; + } else if (res == INFO_FORMAT_CHANGED) { + sp<AMessage> format = new AMessage(); + res = mCodec->getOutputFormat(&format); + if (res != OK) { + ALOGE("Failed to get output format for %s after INFO event (err=%d)", + mComponentName.c_str(), res); + handleError(res); + return false; + } - if (mCSDIndex < mCSD.size()) { - outBuffer = mCSD.editItemAt(mCSDIndex++); - outBuffer->meta()->setInt64("timeUs", 0); + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatOutputFormatChanged); + notify->setMessage("format", format); + notify->post(); + return true; + } else if (res == INFO_DISCONTINUITY) { + // nothing to do + return true; + } else if (res != OK) { + if (res != -EAGAIN) { + handleError(res); + } + return false; + } - reply->setBuffer("buffer", outBuffer); - reply->post(); - return; + CHECK_LT(bufferIx, mOutputBuffers.size()); + sp<ABuffer> buffer = mOutputBuffers[bufferIx]; + buffer->setRange(offset, size); + buffer->meta()->clear(); + buffer->meta()->setInt64("timeUs", timeUs); + if (flags & MediaCodec::BUFFER_FLAG_EOS) { + buffer->meta()->setInt32("eos", true); } + // we do not expect CODECCONFIG or SYNCFRAME for decoder + + sp<AMessage> reply = new AMessage(kWhatRenderBuffer, id()); + reply->setSize("buffer-ix", bufferIx); + reply->setInt32("generation", mBufferGeneration); sp<AMessage> notify = mNotify->dup(); - notify->setMessage("codec-request", msg); + notify->setInt32("what", kWhatDrainThisBuffer); + notify->setBuffer("buffer", buffer); + notify->setMessage("reply", reply); notify->post(); + + // FIXME: This should be handled after rendering is complete, + // but Renderer needs it now + if (flags & MediaCodec::BUFFER_FLAG_EOS) { + ALOGV("queueing eos [%s]", mComponentName.c_str()); + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatEOS); + notify->setInt32("err", ERROR_END_OF_STREAM); + notify->post(); + } + return true; } -void NuPlayer::Decoder::signalFlush() { - if (mCodec != NULL) { - mCodec->signalFlush(); +void NuPlayer::Decoder::onRenderBuffer(const sp<AMessage> &msg) { + status_t err; + int32_t render; + size_t bufferIx; + CHECK(msg->findSize("buffer-ix", &bufferIx)); + if (msg->findInt32("render", &render) && render) { + err = mCodec->renderOutputBufferAndRelease(bufferIx); + } else { + err = mCodec->releaseOutputBuffer(bufferIx); + } + if (err != OK) { + ALOGE("failed to release output buffer for %s (err=%d)", + mComponentName.c_str(), err); + handleError(err); } } -void NuPlayer::Decoder::signalResume() { +void NuPlayer::Decoder::onFlush() { + status_t err = OK; if (mCodec != NULL) { - mCodec->signalResume(); + err = mCodec->flush(); + ++mBufferGeneration; } + + if (err != OK) { + ALOGE("failed to flush %s (err=%d)", mComponentName.c_str(), err); + handleError(err); + return; + } + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatFlushCompleted); + notify->post(); } -void NuPlayer::Decoder::initiateShutdown() { +void NuPlayer::Decoder::onShutdown() { + status_t err = OK; if (mCodec != NULL) { - mCodec->initiateShutdown(); + err = mCodec->release(); + mCodec = NULL; + ++mBufferGeneration; + + if (mNativeWindow != NULL) { + // reconnect to surface as MediaCodec disconnected from it + CHECK_EQ((int)NO_ERROR, + native_window_api_connect( + mNativeWindow->getNativeWindow().get(), + NATIVE_WINDOW_API_MEDIA)); + } + mComponentName = "decoder"; + } + + if (err != OK) { + ALOGE("failed to release %s (err=%d)", mComponentName.c_str(), err); + handleError(err); + return; + } + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatShutdownCompleted); + notify->post(); +} + +void NuPlayer::Decoder::onMessageReceived(const sp<AMessage> &msg) { + ALOGV("[%s] onMessage: %s", mComponentName.c_str(), msg->debugString().c_str()); + + switch (msg->what()) { + case kWhatConfigure: + { + sp<AMessage> format; + CHECK(msg->findMessage("format", &format)); + onConfigure(format); + break; + } + + case kWhatCodecNotify: + { + if (!isStaleReply(msg)) { + while (handleAnInputBuffer()) { + } + + while (handleAnOutputBuffer()) { + } + } + + requestCodecNotification(); + break; + } + + case kWhatInputBufferFilled: + { + if (!isStaleReply(msg)) { + onInputBufferFilled(msg); + } + break; + } + + case kWhatRenderBuffer: + { + if (!isStaleReply(msg)) { + onRenderBuffer(msg); + } + break; + } + + case kWhatFlush: + { + onFlush(); + break; + } + + case kWhatShutdown: + { + onShutdown(); + break; + } + + default: + TRESPASS(); + break; } } +void NuPlayer::Decoder::signalFlush() { + (new AMessage(kWhatFlush, id()))->post(); +} + +void NuPlayer::Decoder::signalResume() { + // nothing to do +} + +void NuPlayer::Decoder::initiateShutdown() { + (new AMessage(kWhatShutdown, id()))->post(); +} + bool NuPlayer::Decoder::supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const { if (targetFormat == NULL) { return true; @@ -163,14 +469,16 @@ bool NuPlayer::Decoder::supportsSeamlessAudioFormatChange(const sp<AMessage> &ta const char * keys[] = { "channel-count", "sample-rate", "is-adts" }; for (unsigned int i = 0; i < sizeof(keys) / sizeof(keys[0]); i++) { int32_t oldVal, newVal; - if (!mFormat->findInt32(keys[i], &oldVal) || !targetFormat->findInt32(keys[i], &newVal) - || oldVal != newVal) { + if (!mOutputFormat->findInt32(keys[i], &oldVal) || + !targetFormat->findInt32(keys[i], &newVal) || + oldVal != newVal) { return false; } } sp<ABuffer> oldBuf, newBuf; - if (mFormat->findBuffer("csd-0", &oldBuf) && targetFormat->findBuffer("csd-0", &newBuf)) { + if (mOutputFormat->findBuffer("csd-0", &oldBuf) && + targetFormat->findBuffer("csd-0", &newBuf)) { if (oldBuf->size() != newBuf->size()) { return false; } @@ -181,7 +489,7 @@ bool NuPlayer::Decoder::supportsSeamlessAudioFormatChange(const sp<AMessage> &ta } bool NuPlayer::Decoder::supportsSeamlessFormatChange(const sp<AMessage> &targetFormat) const { - if (mFormat == NULL) { + if (mOutputFormat == NULL) { return false; } @@ -190,7 +498,7 @@ bool NuPlayer::Decoder::supportsSeamlessFormatChange(const sp<AMessage> &targetF } AString oldMime, newMime; - if (!mFormat->findString("mime", &oldMime) + if (!mOutputFormat->findString("mime", &oldMime) || !targetFormat->findString("mime", &newMime) || !(oldMime == newMime)) { return false; @@ -201,7 +509,10 @@ bool NuPlayer::Decoder::supportsSeamlessFormatChange(const sp<AMessage> &targetF if (audio) { seamless = supportsSeamlessAudioFormatChange(targetFormat); } else { - seamless = mCodec != NULL && mCodec->isConfiguredForAdaptivePlayback(); + int32_t isAdaptive; + seamless = (mCodec != NULL && + mInputFormat->findInt32("adaptive-playback", &isAdaptive) && + isAdaptive); } ALOGV("%s seamless support for %s", seamless ? "yes" : "no", oldMime.c_str()); diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h index 78ea74a..94243fc 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h +++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h @@ -25,12 +25,14 @@ namespace android { struct ABuffer; +struct MediaCodec; struct NuPlayer::Decoder : public AHandler { Decoder(const sp<AMessage> ¬ify, const sp<NativeWindowWrapper> &nativeWindow = NULL); void configure(const sp<AMessage> &format); + void init(); void signalFlush(); void signalResume(); @@ -38,7 +40,18 @@ struct NuPlayer::Decoder : public AHandler { bool supportsSeamlessFormatChange(const sp<AMessage> &to) const; + enum { + kWhatFillThisBuffer = 'flTB', + kWhatDrainThisBuffer = 'drTB', + kWhatOutputFormatChanged = 'fmtC', + kWhatFlushCompleted = 'flsC', + kWhatShutdownCompleted = 'shDC', + kWhatEOS = 'eos ', + kWhatError = 'err ', + }; + protected: + virtual ~Decoder(); virtual void onMessageReceived(const sp<AMessage> &msg); @@ -46,21 +59,40 @@ protected: private: enum { kWhatCodecNotify = 'cdcN', + kWhatConfigure = 'conf', + kWhatInputBufferFilled = 'inpF', + kWhatRenderBuffer = 'rndr', + kWhatFlush = 'flus', + kWhatShutdown = 'shuD', }; sp<AMessage> mNotify; sp<NativeWindowWrapper> mNativeWindow; - sp<AMessage> mFormat; - sp<ACodec> mCodec; + sp<AMessage> mInputFormat; + sp<AMessage> mOutputFormat; + sp<MediaCodec> mCodec; sp<ALooper> mCodecLooper; + sp<ALooper> mDecoderLooper; + + Vector<sp<ABuffer> > mInputBuffers; + Vector<sp<ABuffer> > mOutputBuffers; + + void handleError(int32_t err); + bool handleAnInputBuffer(); + bool handleAnOutputBuffer(); - Vector<sp<ABuffer> > mCSD; - size_t mCSDIndex; + void requestCodecNotification(); + bool isStaleReply(const sp<AMessage> &msg); - sp<AMessage> makeFormat(const sp<MetaData> &meta); + void onConfigure(const sp<AMessage> &format); + void onFlush(); + void onInputBufferFilled(const sp<AMessage> &msg); + void onRenderBuffer(const sp<AMessage> &msg); + void onShutdown(); - void onFillThisBuffer(const sp<AMessage> &msg); + int32_t mBufferGeneration; + AString mComponentName; bool supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const; diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp index bf5271e..a070c1a 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp @@ -20,8 +20,6 @@ #include "NuPlayerRenderer.h" -#include "SoftwareRenderer.h" - #include <media/stagefright/foundation/ABuffer.h> #include <media/stagefright/foundation/ADebug.h> #include <media/stagefright/foundation/AMessage.h> @@ -36,7 +34,6 @@ NuPlayer::Renderer::Renderer( const sp<AMessage> ¬ify, uint32_t flags) : mAudioSink(sink), - mSoftRenderer(NULL), mNotify(notify), mFlags(flags), mNumFramesWritten(0), @@ -60,12 +57,6 @@ NuPlayer::Renderer::Renderer( } NuPlayer::Renderer::~Renderer() { - delete mSoftRenderer; -} - -void NuPlayer::Renderer::setSoftRenderer(SoftwareRenderer *softRenderer) { - delete mSoftRenderer; - mSoftRenderer = softRenderer; } void NuPlayer::Renderer::queueBuffer( @@ -425,9 +416,6 @@ void NuPlayer::Renderer::onDrainVideoQueue() { ALOGV("rendering video at media time %.2f secs", (mFlags & FLAG_REAL_TIME ? realTimeUs : (realTimeUs + mAnchorTimeMediaUs - mAnchorTimeRealUs)) / 1E6); - if (mSoftRenderer != NULL) { - mSoftRenderer->render(entry->mBuffer->data(), entry->mBuffer->size(), NULL); - } } entry->mNotifyConsumed->setInt32("render", !tooLate); diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h index 9124e03..94a05ea 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h +++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h @@ -23,7 +23,6 @@ namespace android { struct ABuffer; -class SoftwareRenderer; struct NuPlayer::Renderer : public AHandler { enum Flags { @@ -57,8 +56,6 @@ struct NuPlayer::Renderer : public AHandler { kWhatMediaRenderingStart = 'mdrd', }; - void setSoftRenderer(SoftwareRenderer *softRenderer); - protected: virtual ~Renderer(); @@ -86,7 +83,6 @@ private: static const int64_t kMinPositionUpdateDelayUs; sp<MediaPlayerBase::AudioSink> mAudioSink; - SoftwareRenderer *mSoftRenderer; sp<AMessage> mNotify; uint32_t mFlags; List<QueueEntry> mAudioQueue; diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp index 9c48587..e9e96d1 100644 --- a/media/libstagefright/ACodec.cpp +++ b/media/libstagefright/ACodec.cpp @@ -365,7 +365,6 @@ ACodec::ACodec() mIsEncoder(false), mUseMetadataOnEncoderOutput(false), mShutdownInProgress(false), - mIsConfiguredForAdaptivePlayback(false), mEncoderDelay(0), mEncoderPadding(0), mChannelMaskPresent(false), @@ -643,18 +642,33 @@ status_t ACodec::configureOutputBuffersFromNativeWindow( return err; } - // XXX: Is this the right logic to use? It's not clear to me what the OMX - // buffer counts refer to - how do they account for the renderer holding on - // to buffers? - if (def.nBufferCountActual < def.nBufferCountMin + *minUndequeuedBuffers) { - OMX_U32 newBufferCount = def.nBufferCountMin + *minUndequeuedBuffers; + // FIXME: assume that surface is controlled by app (native window + // returns the number for the case when surface is not controlled by app) + (*minUndequeuedBuffers)++; + + + // Use conservative allocation while also trying to reduce starvation + // + // 1. allocate at least nBufferCountMin + minUndequeuedBuffers - that is the + // minimum needed for the consumer to be able to work + // 2. try to allocate two (2) additional buffers to reduce starvation from + // the consumer + for (OMX_U32 extraBuffers = 2; /* condition inside loop */; extraBuffers--) { + OMX_U32 newBufferCount = + def.nBufferCountMin + *minUndequeuedBuffers + extraBuffers; def.nBufferCountActual = newBufferCount; err = mOMX->setParameter( mNode, OMX_IndexParamPortDefinition, &def, sizeof(def)); - if (err != OK) { - ALOGE("[%s] setting nBufferCountActual to %lu failed: %d", - mComponentName.c_str(), newBufferCount, err); + if (err == OK) { + *minUndequeuedBuffers += extraBuffers; + break; + } + + ALOGW("[%s] setting nBufferCountActual to %lu failed: %d", + mComponentName.c_str(), newBufferCount, err); + /* exit condition */ + if (extraBuffers == 0) { return err; } } @@ -679,6 +693,7 @@ status_t ACodec::allocateOutputBuffersFromNativeWindow() { &bufferCount, &bufferSize, &minUndequeuedBuffers); if (err != 0) return err; + mNumUndequeuedBuffers = minUndequeuedBuffers; ALOGV("[%s] Allocating %lu buffers from a native window of size %lu on " "output port", @@ -744,6 +759,7 @@ status_t ACodec::allocateOutputMetaDataBuffers() { &bufferCount, &bufferSize, &minUndequeuedBuffers); if (err != 0) return err; + mNumUndequeuedBuffers = minUndequeuedBuffers; ALOGV("[%s] Allocating %lu meta buffers on output port", mComponentName.c_str(), bufferCount); @@ -1041,6 +1057,9 @@ status_t ACodec::configureCodec( encoder = false; } + sp<AMessage> inputFormat = new AMessage(); + sp<AMessage> outputFormat = new AMessage(); + mIsEncoder = encoder; status_t err = setComponentRole(encoder /* isEncoder */, mime); @@ -1142,7 +1161,9 @@ status_t ACodec::configureCodec( int32_t haveNativeWindow = msg->findObject("native-window", &obj) && obj != NULL; mStoreMetaDataInOutputBuffers = false; - mIsConfiguredForAdaptivePlayback = false; + if (video && !encoder) { + inputFormat->setInt32("adaptive-playback", false); + } if (!encoder && video && haveNativeWindow) { err = mOMX->storeMetaDataInBuffers(mNode, kPortIndexOutput, OMX_TRUE); if (err != OK) { @@ -1187,14 +1208,19 @@ status_t ACodec::configureCodec( ALOGW_IF(err != OK, "[%s] prepareForAdaptivePlayback failed w/ err %d", mComponentName.c_str(), err); - mIsConfiguredForAdaptivePlayback = (err == OK); + + if (err == OK) { + inputFormat->setInt32("max-width", maxWidth); + inputFormat->setInt32("max-height", maxHeight); + inputFormat->setInt32("adaptive-playback", true); + } } // allow failure err = OK; } else { ALOGV("[%s] storeMetaDataInBuffers succeeded", mComponentName.c_str()); mStoreMetaDataInOutputBuffers = true; - mIsConfiguredForAdaptivePlayback = true; + inputFormat->setInt32("adaptive-playback", true); } int32_t push; @@ -1334,6 +1360,11 @@ status_t ACodec::configureCodec( err = setMinBufferSize(kPortIndexInput, 8192); // XXX } + CHECK_EQ(getPortFormat(kPortIndexInput, inputFormat), (status_t)OK); + CHECK_EQ(getPortFormat(kPortIndexOutput, outputFormat), (status_t)OK); + mInputFormat = inputFormat; + mOutputFormat = outputFormat; + return err; } @@ -2498,19 +2529,7 @@ void ACodec::waitUntilAllPossibleNativeWindowBuffersAreReturnedToUs() { return; } - int minUndequeuedBufs = 0; - status_t err = mNativeWindow->query( - mNativeWindow.get(), NATIVE_WINDOW_MIN_UNDEQUEUED_BUFFERS, - &minUndequeuedBufs); - - if (err != OK) { - ALOGE("[%s] NATIVE_WINDOW_MIN_UNDEQUEUED_BUFFERS query failed: %s (%d)", - mComponentName.c_str(), strerror(-err), -err); - - minUndequeuedBufs = 0; - } - - while (countBuffersOwnedByNativeWindow() > (size_t)minUndequeuedBufs + while (countBuffersOwnedByNativeWindow() > mNumUndequeuedBuffers && dequeueBufferFromNativeWindow() != NULL) { // these buffers will be submitted as regular buffers; account for this if (mStoreMetaDataInOutputBuffers && mMetaDataBuffersToSubmit > 0) { @@ -2556,79 +2575,78 @@ void ACodec::processDeferredMessages() { } } -void ACodec::sendFormatChange(const sp<AMessage> &reply) { - sp<AMessage> notify = mNotify->dup(); - notify->setInt32("what", kWhatOutputFormatChanged); - +status_t ACodec::getPortFormat(OMX_U32 portIndex, sp<AMessage> ¬ify) { + // TODO: catch errors an return them instead of using CHECK OMX_PARAM_PORTDEFINITIONTYPE def; InitOMXParams(&def); - def.nPortIndex = kPortIndexOutput; + def.nPortIndex = portIndex; CHECK_EQ(mOMX->getParameter( mNode, OMX_IndexParamPortDefinition, &def, sizeof(def)), (status_t)OK); - CHECK_EQ((int)def.eDir, (int)OMX_DirOutput); + CHECK_EQ((int)def.eDir, + (int)(portIndex == kPortIndexOutput ? OMX_DirOutput : OMX_DirInput)); switch (def.eDomain) { case OMX_PortDomainVideo: { OMX_VIDEO_PORTDEFINITIONTYPE *videoDef = &def.format.video; + switch ((int)videoDef->eCompressionFormat) { + case OMX_VIDEO_CodingUnused: + { + CHECK(mIsEncoder ^ (portIndex == kPortIndexOutput)); + notify->setString("mime", MEDIA_MIMETYPE_VIDEO_RAW); + + notify->setInt32("stride", videoDef->nStride); + notify->setInt32("slice-height", videoDef->nSliceHeight); + notify->setInt32("color-format", videoDef->eColorFormat); + + OMX_CONFIG_RECTTYPE rect; + InitOMXParams(&rect); + rect.nPortIndex = kPortIndexOutput; + + if (mOMX->getConfig( + mNode, OMX_IndexConfigCommonOutputCrop, + &rect, sizeof(rect)) != OK) { + rect.nLeft = 0; + rect.nTop = 0; + rect.nWidth = videoDef->nFrameWidth; + rect.nHeight = videoDef->nFrameHeight; + } - AString mime; - if (!mIsEncoder) { - notify->setString("mime", MEDIA_MIMETYPE_VIDEO_RAW); - } else if (GetMimeTypeForVideoCoding( - videoDef->eCompressionFormat, &mime) != OK) { - notify->setString("mime", "application/octet-stream"); - } else { - notify->setString("mime", mime.c_str()); - } - - notify->setInt32("width", videoDef->nFrameWidth); - notify->setInt32("height", videoDef->nFrameHeight); - - if (!mIsEncoder) { - notify->setInt32("stride", videoDef->nStride); - notify->setInt32("slice-height", videoDef->nSliceHeight); - notify->setInt32("color-format", videoDef->eColorFormat); - - OMX_CONFIG_RECTTYPE rect; - InitOMXParams(&rect); - rect.nPortIndex = kPortIndexOutput; - - if (mOMX->getConfig( - mNode, OMX_IndexConfigCommonOutputCrop, - &rect, sizeof(rect)) != OK) { - rect.nLeft = 0; - rect.nTop = 0; - rect.nWidth = videoDef->nFrameWidth; - rect.nHeight = videoDef->nFrameHeight; - } + CHECK_GE(rect.nLeft, 0); + CHECK_GE(rect.nTop, 0); + CHECK_GE(rect.nWidth, 0u); + CHECK_GE(rect.nHeight, 0u); + CHECK_LE(rect.nLeft + rect.nWidth - 1, videoDef->nFrameWidth); + CHECK_LE(rect.nTop + rect.nHeight - 1, videoDef->nFrameHeight); - CHECK_GE(rect.nLeft, 0); - CHECK_GE(rect.nTop, 0); - CHECK_GE(rect.nWidth, 0u); - CHECK_GE(rect.nHeight, 0u); - CHECK_LE(rect.nLeft + rect.nWidth - 1, videoDef->nFrameWidth); - CHECK_LE(rect.nTop + rect.nHeight - 1, videoDef->nFrameHeight); - - notify->setRect( - "crop", - rect.nLeft, - rect.nTop, - rect.nLeft + rect.nWidth - 1, - rect.nTop + rect.nHeight - 1); - - if (mNativeWindow != NULL) { - reply->setRect( + notify->setRect( "crop", rect.nLeft, rect.nTop, - rect.nLeft + rect.nWidth, - rect.nTop + rect.nHeight); + rect.nLeft + rect.nWidth - 1, + rect.nTop + rect.nHeight - 1); + + break; + } + default: + { + CHECK(mIsEncoder ^ (portIndex == kPortIndexInput)); + AString mime; + if (GetMimeTypeForVideoCoding( + videoDef->eCompressionFormat, &mime) != OK) { + notify->setString("mime", "application/octet-stream"); + } else { + notify->setString("mime", mime.c_str()); + } + break; } } + + notify->setInt32("width", videoDef->nFrameWidth); + notify->setInt32("height", videoDef->nFrameHeight); break; } @@ -2641,7 +2659,7 @@ void ACodec::sendFormatChange(const sp<AMessage> &reply) { { OMX_AUDIO_PARAM_PCMMODETYPE params; InitOMXParams(¶ms); - params.nPortIndex = kPortIndexOutput; + params.nPortIndex = portIndex; CHECK_EQ(mOMX->getParameter( mNode, OMX_IndexParamAudioPcm, @@ -2661,20 +2679,6 @@ void ACodec::sendFormatChange(const sp<AMessage> &reply) { notify->setString("mime", MEDIA_MIMETYPE_AUDIO_RAW); notify->setInt32("channel-count", params.nChannels); notify->setInt32("sample-rate", params.nSamplingRate); - if (mEncoderDelay + mEncoderPadding) { - size_t frameSize = params.nChannels * sizeof(int16_t); - if (mSkipCutBuffer != NULL) { - size_t prevbufsize = mSkipCutBuffer->size(); - if (prevbufsize != 0) { - ALOGW("Replacing SkipCutBuffer holding %d " - "bytes", - prevbufsize); - } - } - mSkipCutBuffer = new SkipCutBuffer( - mEncoderDelay * frameSize, - mEncoderPadding * frameSize); - } if (mChannelMaskPresent) { notify->setInt32("channel-mask", mChannelMask); @@ -2686,7 +2690,7 @@ void ACodec::sendFormatChange(const sp<AMessage> &reply) { { OMX_AUDIO_PARAM_AACPROFILETYPE params; InitOMXParams(¶ms); - params.nPortIndex = kPortIndexOutput; + params.nPortIndex = portIndex; CHECK_EQ(mOMX->getParameter( mNode, OMX_IndexParamAudioAac, @@ -2703,7 +2707,7 @@ void ACodec::sendFormatChange(const sp<AMessage> &reply) { { OMX_AUDIO_PARAM_AMRTYPE params; InitOMXParams(¶ms); - params.nPortIndex = kPortIndexOutput; + params.nPortIndex = portIndex; CHECK_EQ(mOMX->getParameter( mNode, OMX_IndexParamAudioAmr, @@ -2729,7 +2733,7 @@ void ACodec::sendFormatChange(const sp<AMessage> &reply) { { OMX_AUDIO_PARAM_FLACTYPE params; InitOMXParams(¶ms); - params.nPortIndex = kPortIndexOutput; + params.nPortIndex = portIndex; CHECK_EQ(mOMX->getParameter( mNode, OMX_IndexParamAudioFlac, @@ -2742,11 +2746,45 @@ void ACodec::sendFormatChange(const sp<AMessage> &reply) { break; } + case OMX_AUDIO_CodingMP3: + { + OMX_AUDIO_PARAM_MP3TYPE params; + InitOMXParams(¶ms); + params.nPortIndex = portIndex; + + CHECK_EQ(mOMX->getParameter( + mNode, OMX_IndexParamAudioMp3, + ¶ms, sizeof(params)), + (status_t)OK); + + notify->setString("mime", MEDIA_MIMETYPE_AUDIO_MPEG); + notify->setInt32("channel-count", params.nChannels); + notify->setInt32("sample-rate", params.nSampleRate); + break; + } + + case OMX_AUDIO_CodingVORBIS: + { + OMX_AUDIO_PARAM_VORBISTYPE params; + InitOMXParams(¶ms); + params.nPortIndex = portIndex; + + CHECK_EQ(mOMX->getParameter( + mNode, OMX_IndexParamAudioVorbis, + ¶ms, sizeof(params)), + (status_t)OK); + + notify->setString("mime", MEDIA_MIMETYPE_AUDIO_VORBIS); + notify->setInt32("channel-count", params.nChannels); + notify->setInt32("sample-rate", params.nSampleRate); + break; + } + case OMX_AUDIO_CodingAndroidAC3: { OMX_AUDIO_PARAM_ANDROID_AC3TYPE params; InitOMXParams(¶ms); - params.nPortIndex = kPortIndexOutput; + params.nPortIndex = portIndex; CHECK_EQ((status_t)OK, mOMX->getParameter( mNode, @@ -2761,6 +2799,7 @@ void ACodec::sendFormatChange(const sp<AMessage> &reply) { } default: + ALOGE("UNKNOWN AUDIO CODING: %d\n", audioDef->eEncoding); TRESPASS(); } break; @@ -2770,6 +2809,43 @@ void ACodec::sendFormatChange(const sp<AMessage> &reply) { TRESPASS(); } + return OK; +} + +void ACodec::sendFormatChange(const sp<AMessage> &reply) { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatOutputFormatChanged); + + CHECK_EQ(getPortFormat(kPortIndexOutput, notify), (status_t)OK); + + AString mime; + CHECK(notify->findString("mime", &mime)); + + int32_t left, top, right, bottom; + if (mime == MEDIA_MIMETYPE_VIDEO_RAW && + mNativeWindow != NULL && + notify->findRect("crop", &left, &top, &right, &bottom)) { + // notify renderer of the crop change + // NOTE: native window uses extended right-bottom coordinate + reply->setRect("crop", left, top, right + 1, bottom + 1); + } else if (mime == MEDIA_MIMETYPE_AUDIO_RAW && + (mEncoderDelay || mEncoderPadding)) { + int32_t channelCount; + CHECK(notify->findInt32("channel-count", &channelCount)); + size_t frameSize = channelCount * sizeof(int16_t); + if (mSkipCutBuffer != NULL) { + size_t prevbufsize = mSkipCutBuffer->size(); + if (prevbufsize != 0) { + ALOGW("Replacing SkipCutBuffer holding %d " + "bytes", + prevbufsize); + } + } + mSkipCutBuffer = new SkipCutBuffer( + mEncoderDelay * frameSize, + mEncoderPadding * frameSize); + } + notify->post(); mSentFormat = true; @@ -3799,7 +3875,8 @@ void ACodec::LoadedState::stateEntered() { mCodec->mDequeueCounter = 0; mCodec->mMetaDataBuffersToSubmit = 0; mCodec->mRepeatFrameDelayUs = -1ll; - mCodec->mIsConfiguredForAdaptivePlayback = false; + mCodec->mInputFormat.clear(); + mCodec->mOutputFormat.clear(); if (mCodec->mShutdownInProgress) { bool keepComponentAllocated = mCodec->mKeepComponentAllocated; @@ -3913,6 +3990,8 @@ bool ACodec::LoadedState::onConfigureComponent( { sp<AMessage> notify = mCodec->mNotify->dup(); notify->setInt32("what", ACodec::kWhatComponentConfigured); + notify->setMessage("input-format", mCodec->mInputFormat); + notify->setMessage("output-format", mCodec->mOutputFormat); notify->post(); } diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk index 714b5e0..a9b0c73 100644 --- a/media/libstagefright/Android.mk +++ b/media/libstagefright/Android.mk @@ -118,6 +118,8 @@ LOCAL_MODULE:= libstagefright LOCAL_MODULE_TAGS := optional +LOCAL_32_BIT_ONLY := true + include $(BUILD_SHARED_LIBRARY) include $(call all-makefiles-under,$(LOCAL_PATH)) diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp index fe21296..e0419ca 100644 --- a/media/libstagefright/MediaCodec.cpp +++ b/media/libstagefright/MediaCodec.cpp @@ -352,6 +352,20 @@ status_t MediaCodec::getOutputFormat(sp<AMessage> *format) const { return OK; } +status_t MediaCodec::getInputFormat(sp<AMessage> *format) const { + sp<AMessage> msg = new AMessage(kWhatGetInputFormat, id()); + + sp<AMessage> response; + status_t err; + if ((err = PostAndAwaitResponse(msg, &response)) != OK) { + return err; + } + + CHECK(response->findMessage("format", format)); + + return OK; +} + status_t MediaCodec::getName(AString *name) const { sp<AMessage> msg = new AMessage(kWhatGetName, id()); @@ -642,6 +656,9 @@ void MediaCodec::onMessageReceived(const sp<AMessage> &msg) { // reset input surface flag mHaveInputSurface = false; + CHECK(msg->findMessage("input-format", &mInputFormat)); + CHECK(msg->findMessage("output-format", &mOutputFormat)); + (new AMessage)->postReply(mReplyID); break; } @@ -1330,14 +1347,19 @@ void MediaCodec::onMessageReceived(const sp<AMessage> &msg) { break; } + case kWhatGetInputFormat: case kWhatGetOutputFormat: { + sp<AMessage> format = + (msg->what() == kWhatGetOutputFormat ? mOutputFormat : mInputFormat); + uint32_t replyID; CHECK(msg->senderAwaitsResponse(&replyID)); - if ((mState != STARTED && mState != FLUSHING) + if ((mState != CONFIGURED && mState != STARTING && + mState != STARTED && mState != FLUSHING) || (mFlags & kFlagStickyError) - || mOutputFormat == NULL) { + || format == NULL) { sp<AMessage> response = new AMessage; response->setInt32("err", INVALID_OPERATION); @@ -1346,7 +1368,7 @@ void MediaCodec::onMessageReceived(const sp<AMessage> &msg) { } sp<AMessage> response = new AMessage; - response->setMessage("format", mOutputFormat); + response->setMessage("format", format); response->postReply(replyID); break; } diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp index 4d3b5bd..545ca9d 100644 --- a/media/libstagefright/OMXCodec.cpp +++ b/media/libstagefright/OMXCodec.cpp @@ -94,6 +94,7 @@ static sp<MediaSource> InstantiateSoftwareEncoder( #define CODEC_LOGI(x, ...) ALOGI("[%s] "x, mComponentName, ##__VA_ARGS__) #define CODEC_LOGV(x, ...) ALOGV("[%s] "x, mComponentName, ##__VA_ARGS__) +#define CODEC_LOGW(x, ...) ALOGW("[%s] "x, mComponentName, ##__VA_ARGS__) #define CODEC_LOGE(x, ...) ALOGE("[%s] "x, mComponentName, ##__VA_ARGS__) struct OMXCodecObserver : public BnOMXObserver { @@ -1803,21 +1804,40 @@ status_t OMXCodec::allocateOutputBuffersFromNativeWindow() { strerror(-err), -err); return err; } - - // XXX: Is this the right logic to use? It's not clear to me what the OMX - // buffer counts refer to - how do they account for the renderer holding on - // to buffers? - if (def.nBufferCountActual < def.nBufferCountMin + minUndequeuedBufs) { - OMX_U32 newBufferCount = def.nBufferCountMin + minUndequeuedBufs; + // FIXME: assume that surface is controlled by app (native window + // returns the number for the case when surface is not controlled by app) + minUndequeuedBufs++; + + // Use conservative allocation while also trying to reduce starvation + // + // 1. allocate at least nBufferCountMin + minUndequeuedBuffers - that is the + // minimum needed for the consumer to be able to work + // 2. try to allocate two (2) additional buffers to reduce starvation from + // the consumer + CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d", + def.nBufferCountMin, def.nBufferCountActual, minUndequeuedBufs); + + for (OMX_U32 extraBuffers = 2; /* condition inside loop */; extraBuffers--) { + OMX_U32 newBufferCount = + def.nBufferCountMin + minUndequeuedBufs + extraBuffers; def.nBufferCountActual = newBufferCount; err = mOMX->setParameter( mNode, OMX_IndexParamPortDefinition, &def, sizeof(def)); - if (err != OK) { - CODEC_LOGE("setting nBufferCountActual to %lu failed: %d", - newBufferCount, err); + + if (err == OK) { + minUndequeuedBufs += extraBuffers; + break; + } + + CODEC_LOGW("setting nBufferCountActual to %lu failed: %d", + newBufferCount, err); + /* exit condition */ + if (extraBuffers == 0) { return err; } } + CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d", + def.nBufferCountMin, def.nBufferCountActual, minUndequeuedBufs); err = native_window_set_buffer_count( mNativeWindow.get(), def.nBufferCountActual); diff --git a/media/libstagefright/codecs/aacenc/Android.mk b/media/libstagefright/codecs/aacenc/Android.mk index 58ec3ba..04dc487 100644 --- a/media/libstagefright/codecs/aacenc/Android.mk +++ b/media/libstagefright/codecs/aacenc/Android.mk @@ -117,6 +117,7 @@ ifeq ($(AAC_LIBRARY), fraunhofer) LOCAL_MODULE := libstagefright_soft_aacenc LOCAL_MODULE_TAGS := optional + LOCAL_32_BIT_ONLY := true include $(BUILD_SHARED_LIBRARY) diff --git a/media/libstagefright/codecs/avc/enc/Android.mk b/media/libstagefright/codecs/avc/enc/Android.mk index 537ba42..c2e7b81 100644 --- a/media/libstagefright/codecs/avc/enc/Android.mk +++ b/media/libstagefright/codecs/avc/enc/Android.mk @@ -20,6 +20,7 @@ LOCAL_SRC_FILES := \ LOCAL_MODULE := libstagefright_avcenc +LOCAL_32_BIT_ONLY := true LOCAL_C_INCLUDES := \ $(LOCAL_PATH)/src \ @@ -70,6 +71,7 @@ LOCAL_SHARED_LIBRARIES := \ LOCAL_MODULE := libstagefright_soft_h264enc LOCAL_MODULE_TAGS := optional +LOCAL_32_BIT_ONLY := true LOCAL_CFLAGS += -Werror diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp index a09ab7c..5396022 100644 --- a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp +++ b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp @@ -146,6 +146,23 @@ OMX_ERRORTYPE SoftMP3::internalGetParameter( return OMX_ErrorNone; } + case OMX_IndexParamAudioMp3: + { + OMX_AUDIO_PARAM_MP3TYPE *mp3Params = + (OMX_AUDIO_PARAM_MP3TYPE *)params; + + if (mp3Params->nPortIndex > 1) { + return OMX_ErrorUndefined; + } + + mp3Params->nChannels = mNumChannels; + mp3Params->nBitRate = 0 /* unknown */; + mp3Params->nSampleRate = mSamplingRate; + // other fields are encoder-only + + return OMX_ErrorNone; + } + default: return SimpleSoftOMXComponent::internalGetParameter(index, params); } diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp index dd396e7..19db6eb 100644 --- a/media/libstagefright/httplive/LiveSession.cpp +++ b/media/libstagefright/httplive/LiveSession.cpp @@ -488,7 +488,7 @@ void LiveSession::onConnect(const sp<AMessage> &msg) { mPlaylist = fetchPlaylist(url.c_str(), NULL /* curPlaylistHash */, &dummy); if (mPlaylist == NULL) { - ALOGE("unable to fetch master playlist '%s'.", url.c_str()); + ALOGE("unable to fetch master playlist <URL suppressed>."); postPrepared(ERROR_IO); return; diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp index 87918c8..dacdd40 100644 --- a/media/libstagefright/httplive/M3UParser.cpp +++ b/media/libstagefright/httplive/M3UParser.cpp @@ -798,8 +798,7 @@ status_t M3UParser::parseCipherInfo( if (MakeURL(baseURI.c_str(), val.c_str(), &absURI)) { val = absURI; } else { - ALOGE("failed to make absolute url for '%s'.", - val.c_str()); + ALOGE("failed to make absolute url for <URL suppressed>."); } } diff --git a/media/libstagefright/rtsp/ARTSPConnection.cpp b/media/libstagefright/rtsp/ARTSPConnection.cpp index 4054da6..cc3b63c 100644 --- a/media/libstagefright/rtsp/ARTSPConnection.cpp +++ b/media/libstagefright/rtsp/ARTSPConnection.cpp @@ -239,7 +239,7 @@ void ARTSPConnection::onConnect(const sp<AMessage> &msg) { // right here, since we currently have no way of asking the user // for this information. - ALOGE("Malformed rtsp url %s", url.c_str()); + ALOGE("Malformed rtsp url <URL suppressed>"); reply->setInt32("result", ERROR_MALFORMED); reply->post(); diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h index 45470a3..f3dfc59 100644 --- a/media/libstagefright/rtsp/MyHandler.h +++ b/media/libstagefright/rtsp/MyHandler.h @@ -159,7 +159,7 @@ struct MyHandler : public AHandler { mSessionURL.append(StringPrintf("%u", port)); mSessionURL.append(path); - ALOGI("rewritten session url: '%s'", mSessionURL.c_str()); + ALOGV("rewritten session url: '%s'", mSessionURL.c_str()); } mSessionHost = host; @@ -488,21 +488,32 @@ struct MyHandler : public AHandler { sp<ARTSPResponse> response = static_cast<ARTSPResponse *>(obj.get()); - if (response->mStatusCode == 302) { + if (response->mStatusCode == 301 || response->mStatusCode == 302) { ssize_t i = response->mHeaders.indexOfKey("location"); CHECK_GE(i, 0); - mSessionURL = response->mHeaders.valueAt(i); - - AString request; - request = "DESCRIBE "; - request.append(mSessionURL); - request.append(" RTSP/1.0\r\n"); - request.append("Accept: application/sdp\r\n"); - request.append("\r\n"); + mOriginalSessionURL = response->mHeaders.valueAt(i); + mSessionURL = mOriginalSessionURL; + + // Strip any authentication info from the session url, we don't + // want to transmit user/pass in cleartext. + AString host, path, user, pass; + unsigned port; + if (ARTSPConnection::ParseURL( + mSessionURL.c_str(), &host, &port, &path, &user, &pass) + && user.size() > 0) { + mSessionURL.clear(); + mSessionURL.append("rtsp://"); + mSessionURL.append(host); + mSessionURL.append(":"); + mSessionURL.append(StringPrintf("%u", port)); + mSessionURL.append(path); + + ALOGI("rewritten session url: '%s'", mSessionURL.c_str()); + } - sp<AMessage> reply = new AMessage('desc', id()); - mConn->sendRequest(request.c_str(), reply); + sp<AMessage> reply = new AMessage('conn', id()); + mConn->connect(mOriginalSessionURL.c_str(), reply); break; } diff --git a/media/libstagefright/rtsp/SDPLoader.cpp b/media/libstagefright/rtsp/SDPLoader.cpp index ce1e89d..13e8da3 100644 --- a/media/libstagefright/rtsp/SDPLoader.cpp +++ b/media/libstagefright/rtsp/SDPLoader.cpp @@ -90,7 +90,7 @@ void SDPLoader::onLoad(const sp<AMessage> &msg) { msg->findPointer("headers", (void **)&headers); if (!(mFlags & kFlagIncognito)) { - ALOGI("onLoad '%s'", url.c_str()); + ALOGV("onLoad '%s'", url.c_str()); } else { ALOGI("onLoad <URL suppressed>"); } diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk index f848054..d3e546a 100644 --- a/media/mediaserver/Android.mk +++ b/media/mediaserver/Android.mk @@ -15,6 +15,7 @@ LOCAL_SRC_FILES:= \ LOCAL_SHARED_LIBRARIES := \ libaudioflinger \ + libaudiopolicy \ libcamera_metadata\ libcameraservice \ libmedialogservice \ @@ -33,8 +34,10 @@ LOCAL_C_INCLUDES := \ frameworks/av/media/libmediaplayerservice \ frameworks/av/services/medialog \ frameworks/av/services/audioflinger \ + frameworks/av/services/audiopolicy \ frameworks/av/services/camera/libcameraservice LOCAL_MODULE:= mediaserver +LOCAL_32_BIT_ONLY := true include $(BUILD_EXECUTABLE) diff --git a/media/mtp/MtpProperty.cpp b/media/mtp/MtpProperty.cpp index 375ed9a..3838ce8 100644 --- a/media/mtp/MtpProperty.cpp +++ b/media/mtp/MtpProperty.cpp @@ -190,9 +190,9 @@ void MtpProperty::write(MtpDataPacket& packet) { if (deviceProp) writeValue(packet, mCurrentValue); } - packet.putUInt32(mGroupCode); if (!deviceProp) - packet.putUInt8(mFormFlag); + packet.putUInt32(mGroupCode); + packet.putUInt8(mFormFlag); if (mFormFlag == kFormRange) { writeValue(packet, mMinimumValue); writeValue(packet, mMaximumValue); diff --git a/media/mtp/MtpServer.cpp b/media/mtp/MtpServer.cpp index df87db4..dadfb54 100644 --- a/media/mtp/MtpServer.cpp +++ b/media/mtp/MtpServer.cpp @@ -93,6 +93,7 @@ static const MtpEventCode kSupportedEventCodes[] = { MTP_EVENT_OBJECT_REMOVED, MTP_EVENT_STORE_ADDED, MTP_EVENT_STORE_REMOVED, + MTP_EVENT_DEVICE_PROP_CHANGED, }; MtpServer::MtpServer(int fd, MtpDatabase* database, bool ptp, @@ -261,6 +262,11 @@ void MtpServer::sendStoreRemoved(MtpStorageID id) { sendEvent(MTP_EVENT_STORE_REMOVED, id); } +void MtpServer::sendDevicePropertyChanged(MtpDeviceProperty property) { + ALOGV("sendDevicePropertyChanged %d\n", property); + sendEvent(MTP_EVENT_DEVICE_PROP_CHANGED, property); +} + void MtpServer::sendEvent(MtpEventCode code, uint32_t param1) { if (mSessionOpen) { mEvent.setEventCode(code); diff --git a/media/mtp/MtpServer.h b/media/mtp/MtpServer.h index dfa8258..b3a11e0 100644 --- a/media/mtp/MtpServer.h +++ b/media/mtp/MtpServer.h @@ -104,6 +104,7 @@ public: void sendObjectAdded(MtpObjectHandle handle); void sendObjectRemoved(MtpObjectHandle handle); + void sendDevicePropertyChanged(MtpDeviceProperty property); private: void sendStoreAdded(MtpStorageID id); diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk index 4524d3c..de7e3c3 100644 --- a/services/audioflinger/Android.mk +++ b/services/audioflinger/Android.mk @@ -13,6 +13,16 @@ include $(BUILD_STATIC_LIBRARY) include $(CLEAR_VARS) +LOCAL_SRC_FILES := \ + ServiceUtilities.cpp + +# FIXME Move this library to frameworks/native +LOCAL_MODULE := libserviceutility + +include $(BUILD_STATIC_LIBRARY) + +include $(CLEAR_VARS) + LOCAL_SRC_FILES:= \ AudioFlinger.cpp \ Threads.cpp \ @@ -20,8 +30,6 @@ LOCAL_SRC_FILES:= \ Effects.cpp \ AudioMixer.cpp.arm \ AudioResampler.cpp.arm \ - AudioPolicyService.cpp \ - ServiceUtilities.cpp \ AudioResamplerCubic.cpp.arm \ AudioResamplerSinc.cpp.arm \ AudioResamplerDyn.cpp.arm @@ -29,6 +37,7 @@ LOCAL_SRC_FILES:= \ LOCAL_SRC_FILES += StateQueue.cpp LOCAL_C_INCLUDES := \ + $(TOPDIR)frameworks/av/services/audiopolicy \ $(call include-path-for, audio-effects) \ $(call include-path-for, audio-utils) @@ -50,9 +59,11 @@ LOCAL_SHARED_LIBRARIES := \ LOCAL_STATIC_LIBRARIES := \ libscheduling_policy \ libcpustats \ - libmedia_helper + libmedia_helper \ + libserviceutility LOCAL_MODULE:= libaudioflinger +LOCAL_32_BIT_ONLY := true LOCAL_SRC_FILES += FastMixer.cpp FastMixerState.cpp AudioWatchdog.cpp diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 92ee30e..50179c5 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -509,7 +509,6 @@ sp<IAudioTrack> AudioFlinger::createTrack( audio_io_handle_t output, pid_t tid, int *sessionId, - String8& name, int clientUid, status_t *status) { @@ -559,7 +558,6 @@ sp<IAudioTrack> AudioFlinger::createTrack( { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); - PlaybackThread *effectThread = NULL; if (thread == NULL) { ALOGE("no playback thread found for output handle %d", output); lStatus = BAD_VALUE; @@ -567,24 +565,23 @@ sp<IAudioTrack> AudioFlinger::createTrack( } pid_t pid = IPCThreadState::self()->getCallingPid(); - client = registerPid_l(pid); - ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); + PlaybackThread *effectThread = NULL; if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { + lSessionId = *sessionId; // check if an effect chain with the same session ID is present on another // output thread and move it here. for (size_t i = 0; i < mPlaybackThreads.size(); i++) { sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); if (mPlaybackThreads.keyAt(i) != output) { - uint32_t sessions = t->hasAudioSession(*sessionId); + uint32_t sessions = t->hasAudioSession(lSessionId); if (sessions & PlaybackThread::EFFECT_SESSION) { effectThread = t.get(); break; } } } - lSessionId = *sessionId; } else { // if no audio session id is provided, create one here lSessionId = nextUniqueId(); @@ -625,18 +622,17 @@ sp<IAudioTrack> AudioFlinger::createTrack( } - if (lStatus == NO_ERROR) { - // s for server's pid, n for normal mixer name, f for fast index - name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0, - track->fastIndex()); - trackHandle = new TrackHandle(track); - } else { - // remove local strong reference to Client before deleting the Track so that the Client - // destructor is called by the TrackBase destructor with mLock held + if (lStatus != NO_ERROR) { + // remove local strong reference to Client before deleting the Track so that the + // Client destructor is called by the TrackBase destructor with mLock held client.clear(); track.clear(); + goto Exit; } + // return handle to client + trackHandle = new TrackHandle(track); + Exit: *status = lStatus; return trackHandle; @@ -1324,8 +1320,6 @@ sp<IAudioRecord> AudioFlinger::openRecord( sp<RecordHandle> recordHandle; sp<Client> client; status_t lStatus; - RecordThread *thread; - size_t inFrameCount; int lSessionId; // check calling permissions @@ -1342,9 +1336,9 @@ sp<IAudioRecord> AudioFlinger::openRecord( goto Exit; } - // FIXME when we support more formats, add audio_is_valid_format(format) - // and any explicit restrictions if audio_is_linear_pcm(format) - if (format != AUDIO_FORMAT_PCM_16_BIT) { + // we don't yet support anything other than 16-bit PCM + if (!(audio_is_valid_format(format) && + audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { ALOGE("openRecord() invalid format %#x", format); lStatus = BAD_VALUE; goto Exit; @@ -1357,10 +1351,9 @@ sp<IAudioRecord> AudioFlinger::openRecord( goto Exit; } - // add client to list - { // scope for mLock + { Mutex::Autolock _l(mLock); - thread = checkRecordThread_l(input); + RecordThread *thread = checkRecordThread_l(input); if (thread == NULL) { ALOGE("openRecord() checkRecordThread_l failed"); lStatus = BAD_VALUE; @@ -1377,17 +1370,17 @@ sp<IAudioRecord> AudioFlinger::openRecord( pid_t pid = IPCThreadState::self()->getCallingPid(); client = registerPid_l(pid); - // If no audio session id is provided, create one here if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { lSessionId = *sessionId; } else { + // if no audio session id is provided, create one here lSessionId = nextUniqueId(); if (sessionId != NULL) { *sessionId = lSessionId; } } - // create new record track. - // The record track uses one track in mHardwareMixerThread by convention. + ALOGV("openRecord() lSessionId: %d", lSessionId); + // TODO: the uid should be passed in as a parameter to openRecord recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, frameCount, lSessionId, diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index c2b516b..2367d7d 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -108,7 +108,6 @@ public: audio_io_handle_t output, pid_t tid, int *sessionId, - String8& name, int clientUid, status_t *status /*non-NULL*/); diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 8aee194..12d453e 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -1340,7 +1340,9 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac } *pFrameCount = frameCount; - if (mType == DIRECT) { + switch (mType) { + + case DIRECT: if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " @@ -1350,7 +1352,9 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac goto Exit; } } - } else if (mType == OFFLOAD) { + break; + + case OFFLOAD: if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" "for output %p with format %#x", @@ -1358,7 +1362,9 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac lStatus = BAD_VALUE; goto Exit; } - } else { + break; + + default: if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { ALOGE("createTrack_l() Bad parameter: format %#x \"" "for output %p with format %#x", @@ -1372,11 +1378,13 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac lStatus = BAD_VALUE; goto Exit; } + break; + } lStatus = initCheck(); if (lStatus != NO_ERROR) { - ALOGE("Audio driver not initialized."); + ALOGE("createTrack_l() audio driver not initialized"); goto Exit; } @@ -1416,7 +1424,6 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac // track must be cleared from the caller as the caller has the AF lock goto Exit; } - mTracks.add(track); sp<EffectChain> chain = getEffectChain_l(sessionId); @@ -1786,8 +1793,9 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l() // Originally this was int16_t[] array, need to remove legacy implications. free(mSinkBuffer); mSinkBuffer = NULL; - const size_t sinkBufferSize = mNormalFrameCount * mChannelCount - * audio_bytes_per_sample(mFormat); + // For sink buffer size, we use the frame size from the downstream sink to avoid problems + // with non PCM formats for compressed music, e.g. AAC, and Offload threads. + const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); // We resize the mMixerBuffer according to the requirements of the sink buffer which @@ -5052,6 +5060,7 @@ void AudioFlinger::RecordThread::inputStandBy() mInput->stream->common.standby(&mInput->stream->common); } +// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( const sp<AudioFlinger::Client>& client, uint32_t sampleRate, @@ -5068,12 +5077,6 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe sp<RecordTrack> track; status_t lStatus; - lStatus = initCheck(); - if (lStatus != NO_ERROR) { - ALOGE("createRecordTrack_l() audio driver not initialized"); - goto Exit; - } - // client expresses a preference for FAST, but we get the final say if (*flags & IAudioFlinger::TRACK_FAST) { if ( @@ -5126,7 +5129,11 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe } *pFrameCount = frameCount; - // FIXME use flags and tid similar to createTrack_l() + lStatus = initCheck(); + if (lStatus != NO_ERROR) { + ALOGE("createRecordTrack_l() audio driver not initialized"); + goto Exit; + } { // scope for mLock Mutex::Autolock _l(mLock); @@ -5155,6 +5162,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); } } + lStatus = NO_ERROR; Exit: diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp index 92ed46a..f19cd88 100644 --- a/services/audioflinger/Tracks.cpp +++ b/services/audioflinger/Tracks.cpp @@ -350,39 +350,39 @@ AudioFlinger::PlaybackThread::Track::Track( mResumeToStopping(false), mFlushHwPending(false) { - if (mCblk != NULL) { - if (sharedBuffer == 0) { - mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, - mFrameSize); - } else { - mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, - mFrameSize); - } - mServerProxy = mAudioTrackServerProxy; - // to avoid leaking a track name, do not allocate one unless there is an mCblk - mName = thread->getTrackName_l(channelMask, sessionId); - if (mName < 0) { - ALOGE("no more track names available"); - return; - } - // only allocate a fast track index if we were able to allocate a normal track name - if (flags & IAudioFlinger::TRACK_FAST) { - mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); - ALOG_ASSERT(thread->mFastTrackAvailMask != 0); - int i = __builtin_ctz(thread->mFastTrackAvailMask); - ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); - // FIXME This is too eager. We allocate a fast track index before the - // fast track becomes active. Since fast tracks are a scarce resource, - // this means we are potentially denying other more important fast tracks from - // being created. It would be better to allocate the index dynamically. - mFastIndex = i; - // Read the initial underruns because this field is never cleared by the fast mixer - mObservedUnderruns = thread->getFastTrackUnderruns(i); - thread->mFastTrackAvailMask &= ~(1 << i); - } + if (mCblk == NULL) { + return; + } + + if (sharedBuffer == 0) { + mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, + mFrameSize); + } else { + mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, + mFrameSize); + } + mServerProxy = mAudioTrackServerProxy; + + mName = thread->getTrackName_l(channelMask, sessionId); + if (mName < 0) { + ALOGE("no more track names available"); + return; + } + // only allocate a fast track index if we were able to allocate a normal track name + if (flags & IAudioFlinger::TRACK_FAST) { + mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); + ALOG_ASSERT(thread->mFastTrackAvailMask != 0); + int i = __builtin_ctz(thread->mFastTrackAvailMask); + ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); + // FIXME This is too eager. We allocate a fast track index before the + // fast track becomes active. Since fast tracks are a scarce resource, + // this means we are potentially denying other more important fast tracks from + // being created. It would be better to allocate the index dynamically. + mFastIndex = i; + // Read the initial underruns because this field is never cleared by the fast mixer + mObservedUnderruns = thread->getFastTrackUnderruns(i); + thread->mFastTrackAvailMask &= ~(1 << i); } - ALOGV("Track constructor name %d, calling pid %d", mName, - IPCThreadState::self()->getCallingPid()); } AudioFlinger::PlaybackThread::Track::~Track() @@ -1773,7 +1773,7 @@ status_t AudioFlinger::RecordHandle::onTransact( // ---------------------------------------------------------------------------- -// RecordTrack constructor must be called with AudioFlinger::mLock held +// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held AudioFlinger::RecordThread::RecordTrack::RecordTrack( RecordThread *thread, const sp<Client>& client, @@ -1789,11 +1789,12 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack( // See real initialization of mRsmpInFront at RecordThread::start() mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL) { - ALOGV("RecordTrack constructor"); - if (mCblk != NULL) { - mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize); + if (mCblk == NULL) { + return; } + mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize); + uint32_t channelCount = popcount(channelMask); // FIXME I don't understand either of the channel count checks if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 && diff --git a/services/audiopolicy/Android.mk b/services/audiopolicy/Android.mk new file mode 100644 index 0000000..84565bb --- /dev/null +++ b/services/audiopolicy/Android.mk @@ -0,0 +1,32 @@ +LOCAL_PATH:= $(call my-dir) + +include $(CLEAR_VARS) + +LOCAL_SRC_FILES:= \ + AudioPolicyService.cpp \ + AudioPolicyInterfaceImpl.cpp \ + AudioPolicyClientImpl.cpp + +LOCAL_C_INCLUDES := \ + $(TOPDIR)frameworks/av/services/audioflinger \ + $(call include-path-for, audio-effects) \ + $(call include-path-for, audio-utils) + +LOCAL_SHARED_LIBRARIES := \ + libcutils \ + libutils \ + liblog \ + libbinder \ + libmedia \ + libhardware \ + libhardware_legacy + +LOCAL_STATIC_LIBRARIES := \ + libmedia_helper \ + libserviceutility + +LOCAL_MODULE:= libaudiopolicy + +LOCAL_CFLAGS += -fvisibility=hidden + +include $(BUILD_SHARED_LIBRARY) diff --git a/services/audiopolicy/AudioPolicyClientImpl.cpp b/services/audiopolicy/AudioPolicyClientImpl.cpp new file mode 100644 index 0000000..53f3e2d --- /dev/null +++ b/services/audiopolicy/AudioPolicyClientImpl.cpp @@ -0,0 +1,261 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "AudioPolicyService" +//#define LOG_NDEBUG 0 + +#include "Configuration.h" +#undef __STRICT_ANSI__ +#define __STDINT_LIMITS +#define __STDC_LIMIT_MACROS +#include <stdint.h> + +#include <sys/time.h> +#include <binder/IServiceManager.h> +#include <utils/Log.h> +#include <cutils/properties.h> +#include <binder/IPCThreadState.h> +#include <utils/String16.h> +#include <utils/threads.h> +#include "AudioPolicyService.h" +#include "ServiceUtilities.h" +#include <hardware_legacy/power.h> +#include <media/AudioEffect.h> +#include <media/EffectsFactoryApi.h> +//#include <media/IAudioFlinger.h> + +#include <hardware/hardware.h> +#include <system/audio.h> +#include <system/audio_policy.h> +#include <hardware/audio_policy.h> +#include <audio_effects/audio_effects_conf.h> +#include <media/AudioParameter.h> + + +namespace android { + +/* implementation of the interface to the policy manager */ +extern "C" { + +audio_module_handle_t aps_load_hw_module(void *service __unused, + const char *name) +{ + sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return 0; + } + + return af->loadHwModule(name); +} + +// deprecated: replaced by aps_open_output_on_module() +audio_io_handle_t aps_open_output(void *service __unused, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask, + uint32_t *pLatencyMs, + audio_output_flags_t flags) +{ + sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return 0; + } + + return af->openOutput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask, + pLatencyMs, flags); +} + +audio_io_handle_t aps_open_output_on_module(void *service __unused, + audio_module_handle_t module, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask, + uint32_t *pLatencyMs, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo) +{ + sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return 0; + } + return af->openOutput(module, pDevices, pSamplingRate, pFormat, pChannelMask, + pLatencyMs, flags, offloadInfo); +} + +audio_io_handle_t aps_open_dup_output(void *service __unused, + audio_io_handle_t output1, + audio_io_handle_t output2) +{ + sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return 0; + } + return af->openDuplicateOutput(output1, output2); +} + +int aps_close_output(void *service __unused, audio_io_handle_t output) +{ + sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); + if (af == 0) { + return PERMISSION_DENIED; + } + + return af->closeOutput(output); +} + +int aps_suspend_output(void *service __unused, audio_io_handle_t output) +{ + sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return PERMISSION_DENIED; + } + + return af->suspendOutput(output); +} + +int aps_restore_output(void *service __unused, audio_io_handle_t output) +{ + sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return PERMISSION_DENIED; + } + + return af->restoreOutput(output); +} + +// deprecated: replaced by aps_open_input_on_module(), and acoustics parameter is ignored +audio_io_handle_t aps_open_input(void *service __unused, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask, + audio_in_acoustics_t acoustics __unused) +{ + sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return 0; + } + + return af->openInput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask); +} + +audio_io_handle_t aps_open_input_on_module(void *service __unused, + audio_module_handle_t module, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask) +{ + sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); + if (af == 0) { + ALOGW("%s: could not get AudioFlinger", __func__); + return 0; + } + + return af->openInput(module, pDevices, pSamplingRate, pFormat, pChannelMask); +} + +int aps_close_input(void *service __unused, audio_io_handle_t input) +{ + sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); + if (af == 0) { + return PERMISSION_DENIED; + } + + return af->closeInput(input); +} + +int aps_invalidate_stream(void *service __unused, audio_stream_type_t stream) +{ + sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); + if (af == 0) { + return PERMISSION_DENIED; + } + + return af->invalidateStream(stream); +} + +int aps_move_effects(void *service __unused, int session, + audio_io_handle_t src_output, + audio_io_handle_t dst_output) +{ + sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); + if (af == 0) { + return PERMISSION_DENIED; + } + + return af->moveEffects(session, src_output, dst_output); +} + +char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle, + const char *keys) +{ + String8 result = AudioSystem::getParameters(io_handle, String8(keys)); + return strdup(result.string()); +} + +void aps_set_parameters(void *service, audio_io_handle_t io_handle, + const char *kv_pairs, int delay_ms) +{ + AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; + + audioPolicyService->setParameters(io_handle, kv_pairs, delay_ms); +} + +int aps_set_stream_volume(void *service, audio_stream_type_t stream, + float volume, audio_io_handle_t output, + int delay_ms) +{ + AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; + + return audioPolicyService->setStreamVolume(stream, volume, output, + delay_ms); +} + +int aps_start_tone(void *service, audio_policy_tone_t tone, + audio_stream_type_t stream) +{ + AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; + + return audioPolicyService->startTone(tone, stream); +} + +int aps_stop_tone(void *service) +{ + AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; + + return audioPolicyService->stopTone(); +} + +int aps_set_voice_volume(void *service, float volume, int delay_ms) +{ + AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; + + return audioPolicyService->setVoiceVolume(volume, delay_ms); +} + +}; // extern "C" + +}; // namespace android diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h new file mode 100644 index 0000000..da03ee3 --- /dev/null +++ b/services/audiopolicy/AudioPolicyInterface.h @@ -0,0 +1,261 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef ANDROID_AUDIOPOLICYINTERFACE_H +#define ANDROID_AUDIOPOLICYINTERFACE_H + +#include <media/AudioSystem.h> +#include <media/ToneGenerator.h> +#include <utils/String8.h> + +#include <hardware_legacy/AudioSystemLegacy.h> +#include <hardware/audio_policy.h> + +namespace android_audio_legacy { + using android::Vector; + using android::String8; + using android::ToneGenerator; + +// ---------------------------------------------------------------------------- + +// The AudioPolicyInterface and AudioPolicyClientInterface classes define the communication interfaces +// between the platform specific audio policy manager and Android generic audio policy manager. +// The platform specific audio policy manager must implement methods of the AudioPolicyInterface class. +// This implementation makes use of the AudioPolicyClientInterface to control the activity and +// configuration of audio input and output streams. +// +// The platform specific audio policy manager is in charge of the audio routing and volume control +// policies for a given platform. +// The main roles of this module are: +// - keep track of current system state (removable device connections, phone state, user requests...). +// System state changes and user actions are notified to audio policy manager with methods of the AudioPolicyInterface. +// - process getOutput() queries received when AudioTrack objects are created: Those queries +// return a handler on an output that has been selected, configured and opened by the audio policy manager and that +// must be used by the AudioTrack when registering to the AudioFlinger with the createTrack() method. +// When the AudioTrack object is released, a putOutput() query is received and the audio policy manager can decide +// to close or reconfigure the output depending on other streams using this output and current system state. +// - similarly process getInput() and putInput() queries received from AudioRecord objects and configure audio inputs. +// - process volume control requests: the stream volume is converted from an index value (received from UI) to a float value +// applicable to each output as a function of platform specific settings and current output route (destination device). It +// also make sure that streams are not muted if not allowed (e.g. camera shutter sound in some countries). +// +// The platform specific audio policy manager is provided as a shared library by platform vendors (as for libaudio.so) +// and is linked with libaudioflinger.so + + +// Audio Policy Manager Interface +class AudioPolicyInterface +{ + +public: + virtual ~AudioPolicyInterface() {} + // + // configuration functions + // + + // indicate a change in device connection status + virtual status_t setDeviceConnectionState(audio_devices_t device, + AudioSystem::device_connection_state state, + const char *device_address) = 0; + // retrieve a device connection status + virtual AudioSystem::device_connection_state getDeviceConnectionState(audio_devices_t device, + const char *device_address) = 0; + // indicate a change in phone state. Valid phones states are defined by AudioSystem::audio_mode + virtual void setPhoneState(int state) = 0; + // force using a specific device category for the specified usage + virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) = 0; + // retrieve current device category forced for a given usage + virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage) = 0; + // set a system property (e.g. camera sound always audible) + virtual void setSystemProperty(const char* property, const char* value) = 0; + // check proper initialization + virtual status_t initCheck() = 0; + + // + // Audio routing query functions + // + + // request an output appropriate for playback of the supplied stream type and parameters + virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + AudioSystem::output_flags flags, + const audio_offload_info_t *offloadInfo) = 0; + // indicates to the audio policy manager that the output starts being used by corresponding stream. + virtual status_t startOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int session = 0) = 0; + // indicates to the audio policy manager that the output stops being used by corresponding stream. + virtual status_t stopOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int session = 0) = 0; + // releases the output. + virtual void releaseOutput(audio_io_handle_t output) = 0; + + // request an input appropriate for record from the supplied device with supplied parameters. + virtual audio_io_handle_t getInput(int inputSource, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + AudioSystem::audio_in_acoustics acoustics) = 0; + // indicates to the audio policy manager that the input starts being used. + virtual status_t startInput(audio_io_handle_t input) = 0; + // indicates to the audio policy manager that the input stops being used. + virtual status_t stopInput(audio_io_handle_t input) = 0; + // releases the input. + virtual void releaseInput(audio_io_handle_t input) = 0; + + // + // volume control functions + // + + // initialises stream volume conversion parameters by specifying volume index range. + virtual void initStreamVolume(AudioSystem::stream_type stream, + int indexMin, + int indexMax) = 0; + + // sets the new stream volume at a level corresponding to the supplied index for the + // supplied device. By convention, specifying AUDIO_DEVICE_OUT_DEFAULT means + // setting volume for all devices + virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, + int index, + audio_devices_t device) = 0; + + // retrieve current volume index for the specified stream and the + // specified device. By convention, specifying AUDIO_DEVICE_OUT_DEFAULT means + // querying the volume of the active device. + virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, + int *index, + audio_devices_t device) = 0; + + // return the strategy corresponding to a given stream type + virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream) = 0; + + // return the enabled output devices for the given stream type + virtual audio_devices_t getDevicesForStream(AudioSystem::stream_type stream) = 0; + + // Audio effect management + virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc) = 0; + virtual status_t registerEffect(const effect_descriptor_t *desc, + audio_io_handle_t io, + uint32_t strategy, + int session, + int id) = 0; + virtual status_t unregisterEffect(int id) = 0; + virtual status_t setEffectEnabled(int id, bool enabled) = 0; + + virtual bool isStreamActive(int stream, uint32_t inPastMs = 0) const = 0; + virtual bool isStreamActiveRemotely(int stream, uint32_t inPastMs = 0) const = 0; + virtual bool isSourceActive(audio_source_t source) const = 0; + + //dump state + virtual status_t dump(int fd) = 0; + + virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo) = 0; +}; + + +// Audio Policy client Interface +class AudioPolicyClientInterface +{ +public: + virtual ~AudioPolicyClientInterface() {} + + // + // Audio HW module functions + // + + // loads a HW module. + virtual audio_module_handle_t loadHwModule(const char *name) = 0; + + // + // Audio output Control functions + // + + // opens an audio output with the requested parameters. The parameter values can indicate to use the default values + // in case the audio policy manager has no specific requirements for the output being opened. + // When the function returns, the parameter values reflect the actual values used by the audio hardware output stream. + // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly. + virtual audio_io_handle_t openOutput(audio_module_handle_t module, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask, + uint32_t *pLatencyMs, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo = NULL) = 0; + // creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by + // a special mixer thread in the AudioFlinger. + virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2) = 0; + // closes the output stream + virtual status_t closeOutput(audio_io_handle_t output) = 0; + // suspends the output. When an output is suspended, the corresponding audio hardware output stream is placed in + // standby and the AudioTracks attached to the mixer thread are still processed but the output mix is discarded. + virtual status_t suspendOutput(audio_io_handle_t output) = 0; + // restores a suspended output. + virtual status_t restoreOutput(audio_io_handle_t output) = 0; + + // + // Audio input Control functions + // + + // opens an audio input + virtual audio_io_handle_t openInput(audio_module_handle_t module, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask) = 0; + // closes an audio input + virtual status_t closeInput(audio_io_handle_t input) = 0; + // + // misc control functions + // + + // set a stream volume for a particular output. For the same user setting, a given stream type can have different volumes + // for each output (destination device) it is attached to. + virtual status_t setStreamVolume(AudioSystem::stream_type stream, float volume, audio_io_handle_t output, int delayMs = 0) = 0; + + // invalidate a stream type, causing a reroute to an unspecified new output + virtual status_t invalidateStream(AudioSystem::stream_type stream) = 0; + + // function enabling to send proprietary informations directly from audio policy manager to audio hardware interface. + virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs, int delayMs = 0) = 0; + // function enabling to receive proprietary informations directly from audio hardware interface to audio policy manager. + virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) = 0; + + // request the playback of a tone on the specified stream: used for instance to replace notification sounds when playing + // over a telephony device during a phone call. + virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream) = 0; + virtual status_t stopTone() = 0; + + // set down link audio volume. + virtual status_t setVoiceVolume(float volume, int delayMs = 0) = 0; + + // move effect to the specified output + virtual status_t moveEffects(int session, + audio_io_handle_t srcOutput, + audio_io_handle_t dstOutput) = 0; + +}; + +extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface); +extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface); + + +}; // namespace android + +#endif // ANDROID_AUDIOPOLICYINTERFACE_H diff --git a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/AudioPolicyInterfaceImpl.cpp new file mode 100644 index 0000000..bb62ab3 --- /dev/null +++ b/services/audiopolicy/AudioPolicyInterfaceImpl.cpp @@ -0,0 +1,489 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "AudioPolicyService" +//#define LOG_NDEBUG 0 + +#include <utils/Log.h> +#include "AudioPolicyService.h" +#include "ServiceUtilities.h" + +#include <system/audio.h> +#include <system/audio_policy.h> +#include <hardware/audio_policy.h> + +namespace android { + + +// ---------------------------------------------------------------------------- + +status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device, + audio_policy_dev_state_t state, + const char *device_address) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (!audio_is_output_device(device) && !audio_is_input_device(device)) { + return BAD_VALUE; + } + if (state != AUDIO_POLICY_DEVICE_STATE_AVAILABLE && + state != AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) { + return BAD_VALUE; + } + + ALOGV("setDeviceConnectionState()"); + Mutex::Autolock _l(mLock); + return mpAudioPolicy->set_device_connection_state(mpAudioPolicy, device, + state, device_address); +} + +audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState( + audio_devices_t device, + const char *device_address) +{ + if (mpAudioPolicy == NULL) { + return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; + } + return mpAudioPolicy->get_device_connection_state(mpAudioPolicy, device, + device_address); +} + +status_t AudioPolicyService::setPhoneState(audio_mode_t state) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (uint32_t(state) >= AUDIO_MODE_CNT) { + return BAD_VALUE; + } + + ALOGV("setPhoneState()"); + + // TODO: check if it is more appropriate to do it in platform specific policy manager + AudioSystem::setMode(state); + + Mutex::Autolock _l(mLock); + mpAudioPolicy->set_phone_state(mpAudioPolicy, state); + return NO_ERROR; +} + +status_t AudioPolicyService::setForceUse(audio_policy_force_use_t usage, + audio_policy_forced_cfg_t config) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) { + return BAD_VALUE; + } + if (config < 0 || config >= AUDIO_POLICY_FORCE_CFG_CNT) { + return BAD_VALUE; + } + ALOGV("setForceUse()"); + Mutex::Autolock _l(mLock); + mpAudioPolicy->set_force_use(mpAudioPolicy, usage, config); + return NO_ERROR; +} + +audio_policy_forced_cfg_t AudioPolicyService::getForceUse(audio_policy_force_use_t usage) +{ + if (mpAudioPolicy == NULL) { + return AUDIO_POLICY_FORCE_NONE; + } + if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) { + return AUDIO_POLICY_FORCE_NONE; + } + return mpAudioPolicy->get_force_use(mpAudioPolicy, usage); +} + +audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo) +{ + if (mpAudioPolicy == NULL) { + return 0; + } + ALOGV("getOutput()"); + Mutex::Autolock _l(mLock); + return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, + format, channelMask, flags, offloadInfo); +} + +status_t AudioPolicyService::startOutput(audio_io_handle_t output, + audio_stream_type_t stream, + int session) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + ALOGV("startOutput()"); + Mutex::Autolock _l(mLock); + return mpAudioPolicy->start_output(mpAudioPolicy, output, stream, session); +} + +status_t AudioPolicyService::stopOutput(audio_io_handle_t output, + audio_stream_type_t stream, + int session) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + ALOGV("stopOutput()"); + mOutputCommandThread->stopOutputCommand(output, stream, session); + return NO_ERROR; +} + +status_t AudioPolicyService::doStopOutput(audio_io_handle_t output, + audio_stream_type_t stream, + int session) +{ + ALOGV("doStopOutput from tid %d", gettid()); + Mutex::Autolock _l(mLock); + return mpAudioPolicy->stop_output(mpAudioPolicy, output, stream, session); +} + +void AudioPolicyService::releaseOutput(audio_io_handle_t output) +{ + if (mpAudioPolicy == NULL) { + return; + } + ALOGV("releaseOutput()"); + mOutputCommandThread->releaseOutputCommand(output); +} + +void AudioPolicyService::doReleaseOutput(audio_io_handle_t output) +{ + ALOGV("doReleaseOutput from tid %d", gettid()); + Mutex::Autolock _l(mLock); + mpAudioPolicy->release_output(mpAudioPolicy, output); +} + +audio_io_handle_t AudioPolicyService::getInput(audio_source_t inputSource, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + int audioSession) +{ + if (mpAudioPolicy == NULL) { + return 0; + } + // already checked by client, but double-check in case the client wrapper is bypassed + if (inputSource >= AUDIO_SOURCE_CNT && inputSource != AUDIO_SOURCE_HOTWORD) { + return 0; + } + + if ((inputSource == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) { + return 0; + } + + Mutex::Autolock _l(mLock); + // the audio_in_acoustics_t parameter is ignored by get_input() + audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate, + format, channelMask, (audio_in_acoustics_t) 0); + + if (input == 0) { + return input; + } + // create audio pre processors according to input source + audio_source_t aliasSource = (inputSource == AUDIO_SOURCE_HOTWORD) ? + AUDIO_SOURCE_VOICE_RECOGNITION : inputSource; + + ssize_t index = mInputSources.indexOfKey(aliasSource); + if (index < 0) { + return input; + } + ssize_t idx = mInputs.indexOfKey(input); + InputDesc *inputDesc; + if (idx < 0) { + inputDesc = new InputDesc(audioSession); + mInputs.add(input, inputDesc); + } else { + inputDesc = mInputs.valueAt(idx); + } + + Vector <EffectDesc *> effects = mInputSources.valueAt(index)->mEffects; + for (size_t i = 0; i < effects.size(); i++) { + EffectDesc *effect = effects[i]; + sp<AudioEffect> fx = new AudioEffect(NULL, &effect->mUuid, -1, 0, 0, audioSession, input); + status_t status = fx->initCheck(); + if (status != NO_ERROR && status != ALREADY_EXISTS) { + ALOGW("Failed to create Fx %s on input %d", effect->mName, input); + // fx goes out of scope and strong ref on AudioEffect is released + continue; + } + for (size_t j = 0; j < effect->mParams.size(); j++) { + fx->setParameter(effect->mParams[j]); + } + inputDesc->mEffects.add(fx); + } + setPreProcessorEnabled(inputDesc, true); + return input; +} + +status_t AudioPolicyService::startInput(audio_io_handle_t input) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + Mutex::Autolock _l(mLock); + + return mpAudioPolicy->start_input(mpAudioPolicy, input); +} + +status_t AudioPolicyService::stopInput(audio_io_handle_t input) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + Mutex::Autolock _l(mLock); + + return mpAudioPolicy->stop_input(mpAudioPolicy, input); +} + +void AudioPolicyService::releaseInput(audio_io_handle_t input) +{ + if (mpAudioPolicy == NULL) { + return; + } + Mutex::Autolock _l(mLock); + mpAudioPolicy->release_input(mpAudioPolicy, input); + + ssize_t index = mInputs.indexOfKey(input); + if (index < 0) { + return; + } + InputDesc *inputDesc = mInputs.valueAt(index); + setPreProcessorEnabled(inputDesc, false); + delete inputDesc; + mInputs.removeItemsAt(index); +} + +status_t AudioPolicyService::initStreamVolume(audio_stream_type_t stream, + int indexMin, + int indexMax) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (uint32_t(stream) >= AUDIO_STREAM_CNT) { + return BAD_VALUE; + } + Mutex::Autolock _l(mLock); + mpAudioPolicy->init_stream_volume(mpAudioPolicy, stream, indexMin, indexMax); + return NO_ERROR; +} + +status_t AudioPolicyService::setStreamVolumeIndex(audio_stream_type_t stream, + int index, + audio_devices_t device) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if (uint32_t(stream) >= AUDIO_STREAM_CNT) { + return BAD_VALUE; + } + Mutex::Autolock _l(mLock); + if (mpAudioPolicy->set_stream_volume_index_for_device) { + return mpAudioPolicy->set_stream_volume_index_for_device(mpAudioPolicy, + stream, + index, + device); + } else { + return mpAudioPolicy->set_stream_volume_index(mpAudioPolicy, stream, index); + } +} + +status_t AudioPolicyService::getStreamVolumeIndex(audio_stream_type_t stream, + int *index, + audio_devices_t device) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + if (uint32_t(stream) >= AUDIO_STREAM_CNT) { + return BAD_VALUE; + } + Mutex::Autolock _l(mLock); + if (mpAudioPolicy->get_stream_volume_index_for_device) { + return mpAudioPolicy->get_stream_volume_index_for_device(mpAudioPolicy, + stream, + index, + device); + } else { + return mpAudioPolicy->get_stream_volume_index(mpAudioPolicy, stream, index); + } +} + +uint32_t AudioPolicyService::getStrategyForStream(audio_stream_type_t stream) +{ + if (mpAudioPolicy == NULL) { + return 0; + } + return mpAudioPolicy->get_strategy_for_stream(mpAudioPolicy, stream); +} + +//audio policy: use audio_device_t appropriately + +audio_devices_t AudioPolicyService::getDevicesForStream(audio_stream_type_t stream) +{ + if (mpAudioPolicy == NULL) { + return (audio_devices_t)0; + } + return mpAudioPolicy->get_devices_for_stream(mpAudioPolicy, stream); +} + +audio_io_handle_t AudioPolicyService::getOutputForEffect(const effect_descriptor_t *desc) +{ + // FIXME change return type to status_t, and return NO_INIT here + if (mpAudioPolicy == NULL) { + return 0; + } + Mutex::Autolock _l(mLock); + return mpAudioPolicy->get_output_for_effect(mpAudioPolicy, desc); +} + +status_t AudioPolicyService::registerEffect(const effect_descriptor_t *desc, + audio_io_handle_t io, + uint32_t strategy, + int session, + int id) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + return mpAudioPolicy->register_effect(mpAudioPolicy, desc, io, strategy, session, id); +} + +status_t AudioPolicyService::unregisterEffect(int id) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + return mpAudioPolicy->unregister_effect(mpAudioPolicy, id); +} + +status_t AudioPolicyService::setEffectEnabled(int id, bool enabled) +{ + if (mpAudioPolicy == NULL) { + return NO_INIT; + } + return mpAudioPolicy->set_effect_enabled(mpAudioPolicy, id, enabled); +} + +bool AudioPolicyService::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const +{ + if (mpAudioPolicy == NULL) { + return 0; + } + Mutex::Autolock _l(mLock); + return mpAudioPolicy->is_stream_active(mpAudioPolicy, stream, inPastMs); +} + +bool AudioPolicyService::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const +{ + if (mpAudioPolicy == NULL) { + return 0; + } + Mutex::Autolock _l(mLock); + return mpAudioPolicy->is_stream_active_remotely(mpAudioPolicy, stream, inPastMs); +} + +bool AudioPolicyService::isSourceActive(audio_source_t source) const +{ + if (mpAudioPolicy == NULL) { + return false; + } + if (mpAudioPolicy->is_source_active == 0) { + return false; + } + Mutex::Autolock _l(mLock); + return mpAudioPolicy->is_source_active(mpAudioPolicy, source); +} + +status_t AudioPolicyService::queryDefaultPreProcessing(int audioSession, + effect_descriptor_t *descriptors, + uint32_t *count) +{ + + if (mpAudioPolicy == NULL) { + *count = 0; + return NO_INIT; + } + Mutex::Autolock _l(mLock); + status_t status = NO_ERROR; + + size_t index; + for (index = 0; index < mInputs.size(); index++) { + if (mInputs.valueAt(index)->mSessionId == audioSession) { + break; + } + } + if (index == mInputs.size()) { + *count = 0; + return BAD_VALUE; + } + Vector< sp<AudioEffect> > effects = mInputs.valueAt(index)->mEffects; + + for (size_t i = 0; i < effects.size(); i++) { + effect_descriptor_t desc = effects[i]->descriptor(); + if (i < *count) { + descriptors[i] = desc; + } + } + if (effects.size() > *count) { + status = NO_MEMORY; + } + *count = effects.size(); + return status; +} + +bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info) +{ + if (mpAudioPolicy == NULL) { + ALOGV("mpAudioPolicy == NULL"); + return false; + } + + if (mpAudioPolicy->is_offload_supported == NULL) { + ALOGV("HAL does not implement is_offload_supported"); + return false; + } + + return mpAudioPolicy->is_offload_supported(mpAudioPolicy, &info); +} + + +}; // namespace android diff --git a/services/audiopolicy/AudioPolicyManagerBase.cpp b/services/audiopolicy/AudioPolicyManagerBase.cpp new file mode 100644 index 0000000..6f58cf7 --- /dev/null +++ b/services/audiopolicy/AudioPolicyManagerBase.cpp @@ -0,0 +1,4091 @@ +/* + * 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 + +//#define VERY_VERBOSE_LOGGING +#ifdef VERY_VERBOSE_LOGGING +#define ALOGVV ALOGV +#else +#define ALOGVV(a...) do { } while(0) +#endif + +// A device mask for all audio input devices that are considered "virtual" when evaluating +// active inputs in getActiveInput() +#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL AUDIO_DEVICE_IN_REMOTE_SUBMIX +// A device mask for all audio output devices that are considered "remote" when evaluating +// active output devices in isStreamActiveRemotely() +#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX + +#include <utils/Log.h> +#include <hardware_legacy/AudioPolicyManagerBase.h> +#include <hardware/audio_effect.h> +#include <hardware/audio.h> +#include <math.h> +#include <hardware_legacy/audio_policy_conf.h> +#include <cutils/properties.h> + +namespace android_audio_legacy { + +// ---------------------------------------------------------------------------- +// AudioPolicyInterface implementation +// ---------------------------------------------------------------------------- + + +status_t AudioPolicyManagerBase::setDeviceConnectionState(audio_devices_t device, + AudioSystem::device_connection_state state, + const char *device_address) +{ + SortedVector <audio_io_handle_t> outputs; + + ALOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address); + + // connect/disconnect only 1 device at a time + if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; + + if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) { + ALOGE("setDeviceConnectionState() invalid address: %s", device_address); + return BAD_VALUE; + } + + // handle output devices + if (audio_is_output_device(device)) { + + if (!mHasA2dp && audio_is_a2dp_device(device)) { + ALOGE("setDeviceConnectionState() invalid A2DP device: %x", device); + return BAD_VALUE; + } + if (!mHasUsb && audio_is_usb_device(device)) { + ALOGE("setDeviceConnectionState() invalid USB audio device: %x", device); + return BAD_VALUE; + } + if (!mHasRemoteSubmix && audio_is_remote_submix_device((audio_devices_t)device)) { + ALOGE("setDeviceConnectionState() invalid remote submix audio device: %x", device); + return BAD_VALUE; + } + + // save a copy of the opened output descriptors before any output is opened or closed + // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies() + mPreviousOutputs = mOutputs; + String8 paramStr; + switch (state) + { + // handle output device connection + case AudioSystem::DEVICE_STATE_AVAILABLE: + if (mAvailableOutputDevices & device) { + ALOGW("setDeviceConnectionState() device already connected: %x", device); + return INVALID_OPERATION; + } + ALOGV("setDeviceConnectionState() connecting device %x", device); + + if (mHasA2dp && audio_is_a2dp_device(device)) { + // handle A2DP device connection + AudioParameter param; + param.add(String8(AUDIO_PARAMETER_A2DP_SINK_ADDRESS), String8(device_address)); + paramStr = param.toString(); + } else if (mHasUsb && audio_is_usb_device(device)) { + // handle USB device connection + paramStr = String8(device_address, MAX_DEVICE_ADDRESS_LEN); + } + + if (checkOutputsForDevice(device, state, outputs, paramStr) != NO_ERROR) { + return INVALID_OPERATION; + } + ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %d outputs", + outputs.size()); + // register new device as available + mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | device); + + if (mHasA2dp && audio_is_a2dp_device(device)) { + // handle A2DP device connection + mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN); + mA2dpSuspended = false; + } else if (audio_is_bluetooth_sco_device(device)) { + // handle SCO device connection + mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN); + } else if (mHasUsb && audio_is_usb_device(device)) { + // handle USB device connection + mUsbCardAndDevice = String8(device_address, MAX_DEVICE_ADDRESS_LEN); + } + + break; + // handle output device disconnection + case AudioSystem::DEVICE_STATE_UNAVAILABLE: { + if (!(mAvailableOutputDevices & device)) { + ALOGW("setDeviceConnectionState() device not connected: %x", device); + return INVALID_OPERATION; + } + + ALOGV("setDeviceConnectionState() disconnecting device %x", device); + // remove device from available output devices + mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices & ~device); + + checkOutputsForDevice(device, state, outputs, paramStr); + if (mHasA2dp && audio_is_a2dp_device(device)) { + // handle A2DP device disconnection + mA2dpDeviceAddress = ""; + mA2dpSuspended = false; + } else if (audio_is_bluetooth_sco_device(device)) { + // handle SCO device disconnection + mScoDeviceAddress = ""; + } else if (mHasUsb && audio_is_usb_device(device)) { + // handle USB device disconnection + mUsbCardAndDevice = ""; + } + // not currently handling multiple simultaneous submixes: ignoring remote submix + // case and address + } break; + + default: + ALOGE("setDeviceConnectionState() invalid state: %x", state); + return BAD_VALUE; + } + + checkA2dpSuspend(); + checkOutputForAllStrategies(); + // outputs must be closed after checkOutputForAllStrategies() is executed + if (!outputs.isEmpty()) { + for (size_t i = 0; i < outputs.size(); i++) { + AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]); + // close unused outputs after device disconnection or direct outputs that have been + // opened by checkOutputsForDevice() to query dynamic parameters + if ((state == AudioSystem::DEVICE_STATE_UNAVAILABLE) || + (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) && + (desc->mDirectOpenCount == 0))) { + closeOutput(outputs[i]); + } + } + } + + updateDevicesAndOutputs(); + for (size_t i = 0; i < mOutputs.size(); i++) { + // do not force device change on duplicated output because if device is 0, it will + // also force a device 0 for the two outputs it is duplicated to which may override + // a valid device selection on those outputs. + setOutputDevice(mOutputs.keyAt(i), + getNewDevice(mOutputs.keyAt(i), true /*fromCache*/), + !mOutputs.valueAt(i)->isDuplicated(), + 0); + } + + if (device == AUDIO_DEVICE_OUT_WIRED_HEADSET) { + device = AUDIO_DEVICE_IN_WIRED_HEADSET; + } else if (device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO || + device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET || + device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) { + device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; + } else { + return NO_ERROR; + } + } + // handle input devices + if (audio_is_input_device(device)) { + + switch (state) + { + // handle input device connection + case AudioSystem::DEVICE_STATE_AVAILABLE: { + if (mAvailableInputDevices & device) { + ALOGW("setDeviceConnectionState() device already connected: %d", device); + return INVALID_OPERATION; + } + mAvailableInputDevices = mAvailableInputDevices | (device & ~AUDIO_DEVICE_BIT_IN); + } + break; + + // handle input device disconnection + case AudioSystem::DEVICE_STATE_UNAVAILABLE: { + if (!(mAvailableInputDevices & device)) { + ALOGW("setDeviceConnectionState() device not connected: %d", device); + return INVALID_OPERATION; + } + mAvailableInputDevices = (audio_devices_t) (mAvailableInputDevices & ~device); + } break; + + default: + ALOGE("setDeviceConnectionState() invalid state: %x", state); + return BAD_VALUE; + } + + audio_io_handle_t activeInput = getActiveInput(); + if (activeInput != 0) { + AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput); + audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource); + if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) { + ALOGV("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; + } + + ALOGW("setDeviceConnectionState() invalid device: %x", device); + return BAD_VALUE; +} + +AudioSystem::device_connection_state AudioPolicyManagerBase::getDeviceConnectionState(audio_devices_t device, + const char *device_address) +{ + AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE; + String8 address = String8(device_address); + if (audio_is_output_device(device)) { + if (device & mAvailableOutputDevices) { + if (audio_is_a2dp_device(device) && + (!mHasA2dp || (address != "" && mA2dpDeviceAddress != address))) { + return state; + } + if (audio_is_bluetooth_sco_device(device) && + address != "" && mScoDeviceAddress != address) { + return state; + } + if (audio_is_usb_device(device) && + (!mHasUsb || (address != "" && mUsbCardAndDevice != address))) { + ALOGE("getDeviceConnectionState() invalid device: %x", device); + return state; + } + if (audio_is_remote_submix_device((audio_devices_t)device) && !mHasRemoteSubmix) { + return state; + } + state = AudioSystem::DEVICE_STATE_AVAILABLE; + } + } else if (audio_is_input_device(device)) { + if (device & mAvailableInputDevices) { + state = AudioSystem::DEVICE_STATE_AVAILABLE; + } + } + + return state; +} + +void AudioPolicyManagerBase::setPhoneState(int state) +{ + ALOGV("setPhoneState() state %d", state); + audio_devices_t newDevice = AUDIO_DEVICE_NONE; + if (state < 0 || state >= AudioSystem::NUM_MODES) { + ALOGW("setPhoneState() invalid state %d", state); + return; + } + + if (state == mPhoneState ) { + ALOGW("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()) { + ALOGV("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)) { + ALOGV(" Entering call in setPhoneState()"); + // force routing command to audio hardware when starting a call + // even if no device change is needed + force = true; + for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { + mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = + sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j]; + } + } else if (isStateInCall(oldState) && !isStateInCall(state)) { + ALOGV(" Exiting call in setPhoneState()"); + // force routing command to audio hardware when exiting a call + // even if no device change is needed + force = true; + for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { + mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = + sVolumeProfiles[AUDIO_STREAM_DTMF][j]; + } + } else if (isStateInCall(state) && (state != oldState)) { + ALOGV(" 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(mPrimaryOutput, false /*fromCache*/); + checkA2dpSuspend(); + checkOutputForAllStrategies(); + updateDevicesAndOutputs(); + + AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); + + // force routing command to audio hardware when ending call + // even if no device change is needed + if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) { + newDevice = hwOutputDesc->device(); + } + + int delayMs = 0; + if (isStateInCall(state)) { + nsecs_t sysTime = systemTime(); + for (size_t i = 0; i < mOutputs.size(); i++) { + AudioOutputDescriptor *desc = mOutputs.valueAt(i); + // mute media and sonification strategies and delay device switch by the largest + // latency of any output where either strategy is active. + // This avoid sending the ring tone or music tail into the earpiece or headset. + if ((desc->isStrategyActive(STRATEGY_MEDIA, + SONIFICATION_HEADSET_MUSIC_DELAY, + sysTime) || + desc->isStrategyActive(STRATEGY_SONIFICATION, + SONIFICATION_HEADSET_MUSIC_DELAY, + sysTime)) && + (delayMs < (int)desc->mLatency*2)) { + delayMs = desc->mLatency*2; + } + setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i)); + setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS, + getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/)); + setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i)); + setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS, + getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/)); + } + } + + // change routing is necessary + setOutputDevice(mPrimaryOutput, newDevice, force, delayMs); + + // if entering in call state, handle special case of active streams + // pertaining to sonification strategy see handleIncallSonification() + if (isStateInCall(state)) { + ALOGV("setPhoneState() in call state management: new state is %d", state); + 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::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) +{ + ALOGV("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) { + ALOGW("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 && + config != AudioSystem::FORCE_NO_BT_A2DP) { + ALOGW("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) { + ALOGW("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) { + ALOGW("setForceUse() invalid config %d for FOR_DOCK", config); + } + forceVolumeReeval = true; + mForceUse[usage] = config; + break; + case AudioSystem::FOR_SYSTEM: + if (config != AudioSystem::FORCE_NONE && + config != AudioSystem::FORCE_SYSTEM_ENFORCED) { + ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config); + } + forceVolumeReeval = true; + mForceUse[usage] = config; + break; + default: + ALOGW("setForceUse() invalid usage %d", usage); + break; + } + + // check for device and output changes triggered by new force usage + checkA2dpSuspend(); + checkOutputForAllStrategies(); + updateDevicesAndOutputs(); + for (size_t i = 0; i < mOutputs.size(); i++) { + audio_io_handle_t output = mOutputs.keyAt(i); + audio_devices_t newDevice = getNewDevice(output, true /*fromCache*/); + setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE)); + if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) { + applyStreamVolumes(output, newDevice, 0, true); + } + } + + audio_io_handle_t activeInput = getActiveInput(); + if (activeInput != 0) { + AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput); + audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource); + if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) { + ALOGV("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) +{ + ALOGV("setSystemProperty() property %s, value %s", property, value); +} + +// Find a direct output profile compatible with the parameters passed, even if the input flags do +// not explicitly request a direct output +AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::getProfileForDirectOutput( + audio_devices_t device, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags) +{ + for (size_t i = 0; i < mHwModules.size(); i++) { + if (mHwModules[i]->mHandle == 0) { + continue; + } + for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { + IOProfile *profile = mHwModules[i]->mOutputProfiles[j]; + if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { + if (profile->isCompatibleProfile(device, samplingRate, format, + channelMask, + AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) { + if (mAvailableOutputDevices & profile->mSupportedDevices) { + return mHwModules[i]->mOutputProfiles[j]; + } + } + } else { + if (profile->isCompatibleProfile(device, samplingRate, format, + channelMask, + AUDIO_OUTPUT_FLAG_DIRECT)) { + if (mAvailableOutputDevices & profile->mSupportedDevices) { + return mHwModules[i]->mOutputProfiles[j]; + } + } + } + } + } + return 0; +} + +audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + AudioSystem::output_flags flags, + const audio_offload_info_t *offloadInfo) +{ + audio_io_handle_t output = 0; + uint32_t latency = 0; + routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream); + audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); + ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x", + device, stream, samplingRate, format, channelMask, flags); + +#ifdef AUDIO_POLICY_TEST + if (mCurOutput != 0) { + ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d", + mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput); + + if (mTestOutputs[mCurOutput] == 0) { + ALOGV("getOutput() opening test output"); + AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL); + outputDesc->mDevice = mTestDevice; + outputDesc->mSamplingRate = mTestSamplingRate; + outputDesc->mFormat = mTestFormat; + outputDesc->mChannelMask = mTestChannels; + outputDesc->mLatency = mTestLatencyMs; + outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0); + outputDesc->mRefCount[stream] = 0; + mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice, + &outputDesc->mSamplingRate, + &outputDesc->mFormat, + &outputDesc->mChannelMask, + &outputDesc->mLatency, + outputDesc->mFlags, + offloadInfo); + 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 + //force direct flag if offload flag is set: offloading implies a direct output stream + // and all common behaviors are driven by checking only the direct flag + // this should normally be set appropriately in the policy configuration file + if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { + flags = (AudioSystem::output_flags)(flags | AUDIO_OUTPUT_FLAG_DIRECT); + } + + // Do not allow offloading if one non offloadable effect is enabled. This prevents from + // creating an offloaded track and tearing it down immediately after start when audioflinger + // detects there is an active non offloadable effect. + // FIXME: We should check the audio session here but we do not have it in this context. + // This may prevent offloading in rare situations where effects are left active by apps + // in the background. + IOProfile *profile = NULL; + if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) || + !isNonOffloadableEffectEnabled()) { + profile = getProfileForDirectOutput(device, + samplingRate, + format, + channelMask, + (audio_output_flags_t)flags); + } + + if (profile != NULL) { + AudioOutputDescriptor *outputDesc = NULL; + + for (size_t i = 0; i < mOutputs.size(); i++) { + AudioOutputDescriptor *desc = mOutputs.valueAt(i); + if (!desc->isDuplicated() && (profile == desc->mProfile)) { + outputDesc = desc; + // reuse direct output if currently open and configured with same parameters + if ((samplingRate == outputDesc->mSamplingRate) && + (format == outputDesc->mFormat) && + (channelMask == outputDesc->mChannelMask)) { + outputDesc->mDirectOpenCount++; + ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i)); + return mOutputs.keyAt(i); + } + } + } + // close direct output if currently open and configured with different parameters + if (outputDesc != NULL) { + closeOutput(outputDesc->mId); + } + outputDesc = new AudioOutputDescriptor(profile); + outputDesc->mDevice = device; + outputDesc->mSamplingRate = samplingRate; + outputDesc->mFormat = format; + outputDesc->mChannelMask = channelMask; + outputDesc->mLatency = 0; + outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags); + outputDesc->mRefCount[stream] = 0; + outputDesc->mStopTime[stream] = 0; + outputDesc->mDirectOpenCount = 1; + output = mpClientInterface->openOutput(profile->mModule->mHandle, + &outputDesc->mDevice, + &outputDesc->mSamplingRate, + &outputDesc->mFormat, + &outputDesc->mChannelMask, + &outputDesc->mLatency, + outputDesc->mFlags, + offloadInfo); + + // only accept an output with the requested parameters + if (output == 0 || + (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) || + (format != AUDIO_FORMAT_DEFAULT && format != outputDesc->mFormat) || + (channelMask != 0 && channelMask != outputDesc->mChannelMask)) { + ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d," + "format %d %d, channelMask %04x %04x", output, samplingRate, + outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask, + outputDesc->mChannelMask); + if (output != 0) { + mpClientInterface->closeOutput(output); + } + delete outputDesc; + return 0; + } + audio_io_handle_t srcOutput = getOutputForEffect(); + addOutput(output, outputDesc); + audio_io_handle_t dstOutput = getOutputForEffect(); + if (dstOutput == output) { + mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput); + } + mPreviousOutputs = mOutputs; + ALOGV("getOutput() returns new direct output %d", output); + return output; + } + + // ignoring channel mask due to downmix capability in mixer + + // open a non direct output + + // for non direct outputs, only PCM is supported + if (audio_is_linear_pcm(format)) { + // get which output is suitable for the specified stream. The actual + // routing change will happen when startOutput() will be called + SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs); + + output = selectOutput(outputs, flags); + } + ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d," + "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags); + + ALOGV("getOutput() returns output %d", output); + + return output; +} + +audio_io_handle_t AudioPolicyManagerBase::selectOutput(const SortedVector<audio_io_handle_t>& outputs, + AudioSystem::output_flags flags) +{ + // select one output among several that provide a path to a particular device or set of + // devices (the list was previously build by getOutputsForDevice()). + // The priority is as follows: + // 1: the output with the highest number of requested policy flags + // 2: the primary output + // 3: the first output in the list + + if (outputs.size() == 0) { + return 0; + } + if (outputs.size() == 1) { + return outputs[0]; + } + + int maxCommonFlags = 0; + audio_io_handle_t outputFlags = 0; + audio_io_handle_t outputPrimary = 0; + + for (size_t i = 0; i < outputs.size(); i++) { + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(outputs[i]); + if (!outputDesc->isDuplicated()) { + int commonFlags = (int)AudioSystem::popCount(outputDesc->mProfile->mFlags & flags); + if (commonFlags > maxCommonFlags) { + outputFlags = outputs[i]; + maxCommonFlags = commonFlags; + ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags); + } + if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { + outputPrimary = outputs[i]; + } + } + } + + if (outputFlags != 0) { + return outputFlags; + } + if (outputPrimary != 0) { + return outputPrimary; + } + + return outputs[0]; +} + +status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int session) +{ + ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session); + ssize_t index = mOutputs.indexOfKey(output); + if (index < 0) { + ALOGW("startOutput() unknown output %d", output); + return BAD_VALUE; + } + + AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); + + // increment 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 + // necessary for a correct control of hardware output routing by startOutput() and stopOutput() + outputDesc->changeRefCount(stream, 1); + + if (outputDesc->mRefCount[stream] == 1) { + audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/); + routing_strategy strategy = getStrategy(stream); + bool shouldWait = (strategy == STRATEGY_SONIFICATION) || + (strategy == STRATEGY_SONIFICATION_RESPECTFUL); + uint32_t waitMs = 0; + bool force = false; + for (size_t i = 0; i < mOutputs.size(); i++) { + AudioOutputDescriptor *desc = mOutputs.valueAt(i); + if (desc != outputDesc) { + // force a device change if any other output is managed by the same hw + // module and has a current device selection that differs from selected device. + // In this case, the audio HAL must receive the new device selection so that it can + // change the device currently selected by the other active output. + if (outputDesc->sharesHwModuleWith(desc) && + desc->device() != newDevice) { + force = true; + } + // wait for audio on other active outputs to be presented when starting + // a notification so that audio focus effect can propagate. + uint32_t latency = desc->latency(); + if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) { + waitMs = latency; + } + } + } + uint32_t muteWaitMs = setOutputDevice(output, newDevice, force); + + // 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].getVolumeIndex(newDevice), + output, + newDevice); + + // update the outputs if starting an output with a stream that can affect notification + // routing + handleNotificationRoutingForStream(stream); + if (waitMs > muteWaitMs) { + usleep((waitMs - muteWaitMs) * 2 * 1000); + } + } + return NO_ERROR; +} + + +status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int session) +{ + ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session); + ssize_t index = mOutputs.indexOfKey(output); + if (index < 0) { + ALOGW("stopOutput() unknown output %d", output); + return BAD_VALUE; + } + + AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); + + // 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() + if (outputDesc->mRefCount[stream] == 0) { + outputDesc->mStopTime[stream] = systemTime(); + audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/); + // delay the device switch by twice the latency because stopOutput() is executed when + // the track stop() command is received and at that time the audio track buffer can + // still contain data that needs to be drained. The latency only covers the audio HAL + // and kernel buffers. Also the latency does not always include additional delay in the + // audio path (audio DSP, CODEC ...) + setOutputDevice(output, newDevice, false, outputDesc->mLatency*2); + + // force restoring the device selection on other active outputs if it differs from the + // one being selected for this output + for (size_t i = 0; i < mOutputs.size(); i++) { + audio_io_handle_t curOutput = mOutputs.keyAt(i); + AudioOutputDescriptor *desc = mOutputs.valueAt(i); + if (curOutput != output && + desc->isActive() && + outputDesc->sharesHwModuleWith(desc) && + (newDevice != desc->device())) { + setOutputDevice(curOutput, + getNewDevice(curOutput, false /*fromCache*/), + true, + outputDesc->mLatency*2); + } + } + // update the outputs if stopping one with a stream that can affect notification routing + handleNotificationRoutingForStream(stream); + } + return NO_ERROR; + } else { + ALOGW("stopOutput() refcount is already 0 for output %d", output); + return INVALID_OPERATION; + } +} + +void AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output) +{ + ALOGV("releaseOutput() %d", output); + ssize_t index = mOutputs.indexOfKey(output); + if (index < 0) { + ALOGW("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->isActive()) { + mpClientInterface->closeOutput(output); + delete mOutputs.valueAt(index); + mOutputs.removeItem(output); + mTestOutputs[testIndex] = 0; + } + return; + } +#endif //AUDIO_POLICY_TEST + + AudioOutputDescriptor *desc = mOutputs.valueAt(index); + if (desc->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) { + if (desc->mDirectOpenCount <= 0) { + ALOGW("releaseOutput() invalid open count %d for output %d", + desc->mDirectOpenCount, output); + return; + } + if (--desc->mDirectOpenCount == 0) { + closeOutput(output); + // If effects where present on the output, audioflinger moved them to the primary + // output by default: move them back to the appropriate output. + audio_io_handle_t dstOutput = getOutputForEffect(); + if (dstOutput != mPrimaryOutput) { + mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput); + } + } + } +} + + +audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + AudioSystem::audio_in_acoustics acoustics) +{ + audio_io_handle_t input = 0; + audio_devices_t device = getDeviceForInputSource(inputSource); + + ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x", + inputSource, samplingRate, format, channelMask, acoustics); + + if (device == AUDIO_DEVICE_NONE) { + ALOGW("getInput() could not find device for inputSource %d", inputSource); + return 0; + } + + // adapt channel selection to input source + switch(inputSource) { + case AUDIO_SOURCE_VOICE_UPLINK: + channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK; + break; + case AUDIO_SOURCE_VOICE_DOWNLINK: + channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK; + break; + case AUDIO_SOURCE_VOICE_CALL: + channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK; + break; + default: + break; + } + + IOProfile *profile = getInputProfile(device, + samplingRate, + format, + channelMask); + if (profile == NULL) { + ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d, " + "channelMask %04x", + device, samplingRate, format, channelMask); + return 0; + } + + if (profile->mModule->mHandle == 0) { + ALOGE("getInput(): HW module %s not opened", profile->mModule->mName); + return 0; + } + + AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile); + + inputDesc->mInputSource = inputSource; + inputDesc->mDevice = device; + inputDesc->mSamplingRate = samplingRate; + inputDesc->mFormat = format; + inputDesc->mChannelMask = channelMask; + inputDesc->mRefCount = 0; + input = mpClientInterface->openInput(profile->mModule->mHandle, + &inputDesc->mDevice, + &inputDesc->mSamplingRate, + &inputDesc->mFormat, + &inputDesc->mChannelMask); + + // only accept input with the exact requested set of parameters + if (input == 0 || + (samplingRate != inputDesc->mSamplingRate) || + (format != inputDesc->mFormat) || + (channelMask != inputDesc->mChannelMask)) { + ALOGI("getInput() failed opening input: samplingRate %d, format %d, channelMask %x", + samplingRate, format, channelMask); + 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) +{ + ALOGV("startInput() input %d", input); + ssize_t index = mInputs.indexOfKey(input); + if (index < 0) { + ALOGW("startInput() unknown 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 except if the active input + // uses AUDIO_SOURCE_HOTWORD in which case it is closed. + audio_io_handle_t activeInput = getActiveInput(); + if (!isVirtualInputDevice(inputDesc->mDevice) && activeInput != 0) { + AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput); + if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) { + ALOGW("startInput() preempting already started low-priority input %d", activeInput); + stopInput(activeInput); + releaseInput(activeInput); + } else { + ALOGW("startInput() input %d failed: other input already started", input); + return INVALID_OPERATION; + } + } + } + + audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource); + if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) { + inputDesc->mDevice = newDevice; + } + + // automatically enable the remote submix output when input is started + if (audio_is_remote_submix_device(inputDesc->mDevice)) { + setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, + AudioSystem::DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS); + } + + AudioParameter param = AudioParameter(); + param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice); + + int aliasSource = (inputDesc->mInputSource == AUDIO_SOURCE_HOTWORD) ? + AUDIO_SOURCE_VOICE_RECOGNITION : inputDesc->mInputSource; + + param.addInt(String8(AudioParameter::keyInputSource), aliasSource); + ALOGV("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) +{ + ALOGV("stopInput() input %d", input); + ssize_t index = mInputs.indexOfKey(input); + if (index < 0) { + ALOGW("stopInput() unknown input %d", input); + return BAD_VALUE; + } + AudioInputDescriptor *inputDesc = mInputs.valueAt(index); + + if (inputDesc->mRefCount == 0) { + ALOGW("stopInput() input %d already stopped", input); + return INVALID_OPERATION; + } else { + // automatically disable the remote submix output when input is stopped + if (audio_is_remote_submix_device(inputDesc->mDevice)) { + setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, + AudioSystem::DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS); + } + + 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) +{ + ALOGV("releaseInput() %d", input); + ssize_t index = mInputs.indexOfKey(input); + if (index < 0) { + ALOGW("releaseInput() releasing unknown input %d", input); + return; + } + mpClientInterface->closeInput(input); + delete mInputs.valueAt(index); + mInputs.removeItem(input); + ALOGV("releaseInput() exit"); +} + +void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type stream, + int indexMin, + int indexMax) +{ + ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax); + if (indexMin < 0 || indexMin >= indexMax) { + ALOGW("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, + audio_devices_t device) +{ + + if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) { + return BAD_VALUE; + } + if (!audio_is_output_device(device)) { + return BAD_VALUE; + } + + // Force max volume if stream cannot be muted + if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax; + + ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d", + stream, device, index); + + // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and + // clear all device specific values + if (device == AUDIO_DEVICE_OUT_DEFAULT) { + mStreams[stream].mIndexCur.clear(); + } + mStreams[stream].mIndexCur.add(device, 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++) { + audio_devices_t curDevice = + getDeviceForVolume(mOutputs.valueAt(i)->device()); + if ((device == AUDIO_DEVICE_OUT_DEFAULT) || (device == curDevice)) { + status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice); + if (volStatus != NO_ERROR) { + status = volStatus; + } + } + } + return status; +} + +status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type stream, + int *index, + audio_devices_t device) +{ + if (index == NULL) { + return BAD_VALUE; + } + if (!audio_is_output_device(device)) { + return BAD_VALUE; + } + // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to + // the strategy the stream belongs to. + if (device == AUDIO_DEVICE_OUT_DEFAULT) { + device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/); + } + device = getDeviceForVolume(device); + + *index = mStreams[stream].getVolumeIndex(device); + ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index); + return NO_ERROR; +} + +audio_io_handle_t AudioPolicyManagerBase::selectOutputForEffects( + const SortedVector<audio_io_handle_t>& outputs) +{ + // select one output among several suitable for global effects. + // The priority is as follows: + // 1: An offloaded output. If the effect ends up not being offloadable, + // AudioFlinger will invalidate the track and the offloaded output + // will be closed causing the effect to be moved to a PCM output. + // 2: A deep buffer output + // 3: the first output in the list + + if (outputs.size() == 0) { + return 0; + } + + audio_io_handle_t outputOffloaded = 0; + audio_io_handle_t outputDeepBuffer = 0; + + for (size_t i = 0; i < outputs.size(); i++) { + AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]); + ALOGV("selectOutputForEffects outputs[%d] flags %x", i, desc->mFlags); + if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { + outputOffloaded = outputs[i]; + } + if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) { + outputDeepBuffer = outputs[i]; + } + } + + ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d", + outputOffloaded, outputDeepBuffer); + if (outputOffloaded != 0) { + return outputOffloaded; + } + if (outputDeepBuffer != 0) { + return outputDeepBuffer; + } + + return outputs[0]; +} + +audio_io_handle_t AudioPolicyManagerBase::getOutputForEffect(const effect_descriptor_t *desc) +{ + // apply simple rule where global effects are attached to the same output as MUSIC streams + + routing_strategy strategy = getStrategy(AudioSystem::MUSIC); + audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); + SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs); + + audio_io_handle_t output = selectOutputForEffects(dstOutputs); + ALOGV("getOutputForEffect() got output %d for fx %s flags %x", + output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags); + + return output; +} + +status_t AudioPolicyManagerBase::registerEffect(const effect_descriptor_t *desc, + audio_io_handle_t io, + uint32_t strategy, + int session, + int id) +{ + ssize_t index = mOutputs.indexOfKey(io); + if (index < 0) { + index = mInputs.indexOfKey(io); + if (index < 0) { + ALOGW("registerEffect() unknown io %d", io); + return INVALID_OPERATION; + } + } + + if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) { + ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB", + desc->name, desc->memoryUsage); + return INVALID_OPERATION; + } + mTotalEffectsMemory += desc->memoryUsage; + ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d", + desc->name, io, strategy, session, id); + ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory); + + EffectDescriptor *pDesc = new EffectDescriptor(); + memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t)); + pDesc->mIo = io; + pDesc->mStrategy = (routing_strategy)strategy; + pDesc->mSession = session; + pDesc->mEnabled = false; + + mEffects.add(id, pDesc); + + return NO_ERROR; +} + +status_t AudioPolicyManagerBase::unregisterEffect(int id) +{ + ssize_t index = mEffects.indexOfKey(id); + if (index < 0) { + ALOGW("unregisterEffect() unknown effect ID %d", id); + return INVALID_OPERATION; + } + + EffectDescriptor *pDesc = mEffects.valueAt(index); + + setEffectEnabled(pDesc, false); + + if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) { + ALOGW("unregisterEffect() memory %d too big for total %d", + pDesc->mDesc.memoryUsage, mTotalEffectsMemory); + pDesc->mDesc.memoryUsage = mTotalEffectsMemory; + } + mTotalEffectsMemory -= pDesc->mDesc.memoryUsage; + ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d", + pDesc->mDesc.name, id, pDesc->mDesc.memoryUsage, mTotalEffectsMemory); + + mEffects.removeItem(id); + delete pDesc; + + return NO_ERROR; +} + +status_t AudioPolicyManagerBase::setEffectEnabled(int id, bool enabled) +{ + ssize_t index = mEffects.indexOfKey(id); + if (index < 0) { + ALOGW("unregisterEffect() unknown effect ID %d", id); + return INVALID_OPERATION; + } + + return setEffectEnabled(mEffects.valueAt(index), enabled); +} + +status_t AudioPolicyManagerBase::setEffectEnabled(EffectDescriptor *pDesc, bool enabled) +{ + if (enabled == pDesc->mEnabled) { + ALOGV("setEffectEnabled(%s) effect already %s", + enabled?"true":"false", enabled?"enabled":"disabled"); + return INVALID_OPERATION; + } + + if (enabled) { + if (mTotalEffectsCpuLoad + pDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) { + ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS", + pDesc->mDesc.name, (float)pDesc->mDesc.cpuLoad/10); + return INVALID_OPERATION; + } + mTotalEffectsCpuLoad += pDesc->mDesc.cpuLoad; + ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad); + } else { + if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) { + ALOGW("setEffectEnabled(false) CPU load %d too high for total %d", + pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad); + pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad; + } + mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad; + ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad); + } + pDesc->mEnabled = enabled; + return NO_ERROR; +} + +bool AudioPolicyManagerBase::isNonOffloadableEffectEnabled() +{ + for (size_t i = 0; i < mEffects.size(); i++) { + const EffectDescriptor * const pDesc = mEffects.valueAt(i); + if (pDesc->mEnabled && (pDesc->mStrategy == STRATEGY_MEDIA) && + ((pDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) { + ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d", + pDesc->mDesc.name, pDesc->mSession); + return true; + } + } + return false; +} + +bool AudioPolicyManagerBase::isStreamActive(int stream, uint32_t inPastMs) const +{ + nsecs_t sysTime = systemTime(); + for (size_t i = 0; i < mOutputs.size(); i++) { + const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i); + if (outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) { + return true; + } + } + return false; +} + +bool AudioPolicyManagerBase::isStreamActiveRemotely(int stream, uint32_t inPastMs) const +{ + nsecs_t sysTime = systemTime(); + for (size_t i = 0; i < mOutputs.size(); i++) { + const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i); + if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) && + outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) { + return true; + } + } + return false; +} + +bool AudioPolicyManagerBase::isSourceActive(audio_source_t source) const +{ + for (size_t i = 0; i < mInputs.size(); i++) { + const AudioInputDescriptor * inputDescriptor = mInputs.valueAt(i); + if ((inputDescriptor->mInputSource == (int)source || + (source == (audio_source_t)AUDIO_SOURCE_VOICE_RECOGNITION && + inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) + && (inputDescriptor->mRefCount > 0)) { + 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, " Primary Output: %d\n", mPrimaryOutput); + result.append(buffer); + snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string()); + result.append(buffer); + snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string()); + result.append(buffer); + snprintf(buffer, SIZE, " USB audio ALSA %s\n", mUsbCardAndDevice.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, " 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); + snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AudioSystem::FOR_SYSTEM]); + result.append(buffer); + write(fd, result.string(), result.size()); + + + snprintf(buffer, SIZE, "\nHW Modules dump:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < mHwModules.size(); i++) { + snprintf(buffer, SIZE, "- HW Module %d:\n", i + 1); + write(fd, buffer, strlen(buffer)); + mHwModules[i]->dump(fd); + } + + 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 Can be muted Index Min Index Max Index Cur [device : index]...\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { + snprintf(buffer, SIZE, " %02d ", i); + write(fd, buffer, strlen(buffer)); + mStreams[i].dump(fd); + } + + 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; +} + +// This function checks for the parameters which can be offloaded. +// This can be enhanced depending on the capability of the DSP and policy +// of the system. +bool AudioPolicyManagerBase::isOffloadSupported(const audio_offload_info_t& offloadInfo) +{ + ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d," + " BitRate=%u, duration=%lld us, has_video=%d", + offloadInfo.sample_rate, offloadInfo.channel_mask, + offloadInfo.format, + offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us, + offloadInfo.has_video); + + // Check if offload has been disabled + char propValue[PROPERTY_VALUE_MAX]; + if (property_get("audio.offload.disable", propValue, "0")) { + if (atoi(propValue) != 0) { + ALOGV("offload disabled by audio.offload.disable=%s", propValue ); + return false; + } + } + + // Check if stream type is music, then only allow offload as of now. + if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC) + { + ALOGV("isOffloadSupported: stream_type != MUSIC, returning false"); + return false; + } + + //TODO: enable audio offloading with video when ready + if (offloadInfo.has_video) + { + ALOGV("isOffloadSupported: has_video == true, returning false"); + return false; + } + + //If duration is less than minimum value defined in property, return false + if (property_get("audio.offload.min.duration.secs", propValue, NULL)) { + if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) { + ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue); + return false; + } + } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) { + ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS); + return false; + } + + // Do not allow offloading if one non offloadable effect is enabled. This prevents from + // creating an offloaded track and tearing it down immediately after start when audioflinger + // detects there is an active non offloadable effect. + // FIXME: We should check the audio session here but we do not have it in this context. + // This may prevent offloading in rare situations where effects are left active by apps + // in the background. + if (isNonOffloadableEffectEnabled()) { + return false; + } + + // See if there is a profile to support this. + // AUDIO_DEVICE_NONE + IOProfile *profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */, + offloadInfo.sample_rate, + offloadInfo.format, + offloadInfo.channel_mask, + AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); + ALOGV("isOffloadSupported() profile %sfound", profile != NULL ? "" : "NOT "); + return (profile != NULL); +} + +// ---------------------------------------------------------------------------- +// AudioPolicyManagerBase +// ---------------------------------------------------------------------------- + +AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface) + : +#ifdef AUDIO_POLICY_TEST + Thread(false), +#endif //AUDIO_POLICY_TEST + mPrimaryOutput((audio_io_handle_t)0), + mAvailableOutputDevices(AUDIO_DEVICE_NONE), + mPhoneState(AudioSystem::MODE_NORMAL), + mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), + mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0), + mA2dpSuspended(false), mHasA2dp(false), mHasUsb(false), mHasRemoteSubmix(false), + mSpeakerDrcEnabled(false) +{ + mpClientInterface = clientInterface; + + for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) { + mForceUse[i] = AudioSystem::FORCE_NONE; + } + + mA2dpDeviceAddress = String8(""); + mScoDeviceAddress = String8(""); + mUsbCardAndDevice = String8(""); + + if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) { + if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) { + ALOGE("could not load audio policy configuration file, setting defaults"); + defaultAudioPolicyConfig(); + } + } + + // must be done after reading the policy + initializeVolumeCurves(); + + // open all output streams needed to access attached devices + for (size_t i = 0; i < mHwModules.size(); i++) { + mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName); + if (mHwModules[i]->mHandle == 0) { + ALOGW("could not open HW module %s", mHwModules[i]->mName); + continue; + } + // open all output streams needed to access attached devices + // except for direct output streams that are only opened when they are actually + // required by an app. + for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) + { + const IOProfile *outProfile = mHwModules[i]->mOutputProfiles[j]; + + if ((outProfile->mSupportedDevices & mAttachedOutputDevices) && + ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) { + AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile); + outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice & + outProfile->mSupportedDevices); + audio_io_handle_t output = mpClientInterface->openOutput( + outProfile->mModule->mHandle, + &outputDesc->mDevice, + &outputDesc->mSamplingRate, + &outputDesc->mFormat, + &outputDesc->mChannelMask, + &outputDesc->mLatency, + outputDesc->mFlags); + if (output == 0) { + delete outputDesc; + } else { + mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | + (outProfile->mSupportedDevices & mAttachedOutputDevices)); + if (mPrimaryOutput == 0 && + outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { + mPrimaryOutput = output; + } + addOutput(output, outputDesc); + setOutputDevice(output, + (audio_devices_t)(mDefaultOutputDevice & + outProfile->mSupportedDevices), + true); + } + } + } + } + + ALOGE_IF((mAttachedOutputDevices & ~mAvailableOutputDevices), + "Not output found for attached devices %08x", + (mAttachedOutputDevices & ~mAvailableOutputDevices)); + + ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output"); + + updateDevicesAndOutputs(); + +#ifdef AUDIO_POLICY_TEST + if (mPrimaryOutput != 0) { + AudioParameter outputCmd = AudioParameter(); + outputCmd.addInt(String8("set_id"), 0); + mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); + + mTestDevice = AUDIO_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); + } + for (size_t i = 0; i < mInputs.size(); i++) { + mpClientInterface->closeInput(mInputs.keyAt(i)); + delete mInputs.valueAt(i); + } + for (size_t i = 0; i < mHwModules.size(); i++) { + delete mHwModules[i]; + } +} + +status_t AudioPolicyManagerBase::initCheck() +{ + return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR; +} + +#ifdef AUDIO_POLICY_TEST +bool AudioPolicyManagerBase::threadLoop() +{ + ALOGV("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) { + ALOGV("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")); + + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput); + mpClientInterface->closeOutput(mPrimaryOutput); + + audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle; + + delete mOutputs.valueFor(mPrimaryOutput); + mOutputs.removeItem(mPrimaryOutput); + + AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL); + outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER; + mPrimaryOutput = mpClientInterface->openOutput(moduleHandle, + &outputDesc->mDevice, + &outputDesc->mSamplingRate, + &outputDesc->mFormat, + &outputDesc->mChannelMask, + &outputDesc->mLatency, + outputDesc->mFlags); + if (mPrimaryOutput == 0) { + ALOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d", + outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask); + } else { + AudioParameter outputCmd = AudioParameter(); + outputCmd.addInt(String8("set_id"), 0); + mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); + addOutput(mPrimaryOutput, 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); +} + + +status_t AudioPolicyManagerBase::checkOutputsForDevice(audio_devices_t device, + AudioSystem::device_connection_state state, + SortedVector<audio_io_handle_t>& outputs, + const String8 paramStr) +{ + AudioOutputDescriptor *desc; + + if (state == AudioSystem::DEVICE_STATE_AVAILABLE) { + // first list already open outputs that can be routed to this device + for (size_t i = 0; i < mOutputs.size(); i++) { + desc = mOutputs.valueAt(i); + if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices & device)) { + ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i)); + outputs.add(mOutputs.keyAt(i)); + } + } + // then look for output profiles that can be routed to this device + SortedVector<IOProfile *> profiles; + for (size_t i = 0; i < mHwModules.size(); i++) + { + if (mHwModules[i]->mHandle == 0) { + continue; + } + for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) + { + if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices & device) { + ALOGV("checkOutputsForDevice(): adding profile %d from module %d", j, i); + profiles.add(mHwModules[i]->mOutputProfiles[j]); + } + } + } + + if (profiles.isEmpty() && outputs.isEmpty()) { + ALOGW("checkOutputsForDevice(): No output available for device %04x", device); + return BAD_VALUE; + } + + // open outputs for matching profiles if needed. Direct outputs are also opened to + // query for dynamic parameters and will be closed later by setDeviceConnectionState() + for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { + IOProfile *profile = profiles[profile_index]; + + // nothing to do if one output is already opened for this profile + size_t j; + for (j = 0; j < mOutputs.size(); j++) { + desc = mOutputs.valueAt(j); + if (!desc->isDuplicated() && desc->mProfile == profile) { + break; + } + } + if (j != mOutputs.size()) { + continue; + } + + ALOGV("opening output for device %08x with params %s", device, paramStr.string()); + desc = new AudioOutputDescriptor(profile); + desc->mDevice = device; + audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER; + offloadInfo.sample_rate = desc->mSamplingRate; + offloadInfo.format = desc->mFormat; + offloadInfo.channel_mask = desc->mChannelMask; + + audio_io_handle_t output = mpClientInterface->openOutput(profile->mModule->mHandle, + &desc->mDevice, + &desc->mSamplingRate, + &desc->mFormat, + &desc->mChannelMask, + &desc->mLatency, + desc->mFlags, + &offloadInfo); + if (output != 0) { + if (!paramStr.isEmpty()) { + mpClientInterface->setParameters(output, paramStr); + } + + if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { + String8 reply; + char *value; + if (profile->mSamplingRates[0] == 0) { + reply = mpClientInterface->getParameters(output, + String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)); + ALOGV("checkOutputsForDevice() direct output sup sampling rates %s", + reply.string()); + value = strpbrk((char *)reply.string(), "="); + if (value != NULL) { + loadSamplingRates(value + 1, profile); + } + } + if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { + reply = mpClientInterface->getParameters(output, + String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS)); + ALOGV("checkOutputsForDevice() direct output sup formats %s", + reply.string()); + value = strpbrk((char *)reply.string(), "="); + if (value != NULL) { + loadFormats(value + 1, profile); + } + } + if (profile->mChannelMasks[0] == 0) { + reply = mpClientInterface->getParameters(output, + String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS)); + ALOGV("checkOutputsForDevice() direct output sup channel masks %s", + reply.string()); + value = strpbrk((char *)reply.string(), "="); + if (value != NULL) { + loadOutChannels(value + 1, profile); + } + } + if (((profile->mSamplingRates[0] == 0) && + (profile->mSamplingRates.size() < 2)) || + ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) && + (profile->mFormats.size() < 2)) || + ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) && + (profile->mChannelMasks.size() < 2))) { + ALOGW("checkOutputsForDevice() direct output missing param"); + mpClientInterface->closeOutput(output); + output = 0; + } else { + addOutput(output, desc); + } + } else { + audio_io_handle_t duplicatedOutput = 0; + // add output descriptor + addOutput(output, desc); + // set initial stream volume for device + applyStreamVolumes(output, device, 0, true); + + //TODO: configure audio effect output stage here + + // open a duplicating output thread for the new output and the primary output + duplicatedOutput = mpClientInterface->openDuplicateOutput(output, + mPrimaryOutput); + if (duplicatedOutput != 0) { + // add duplicated output descriptor + AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(NULL); + dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput); + dupOutputDesc->mOutput2 = mOutputs.valueFor(output); + dupOutputDesc->mSamplingRate = desc->mSamplingRate; + dupOutputDesc->mFormat = desc->mFormat; + dupOutputDesc->mChannelMask = desc->mChannelMask; + dupOutputDesc->mLatency = desc->mLatency; + addOutput(duplicatedOutput, dupOutputDesc); + applyStreamVolumes(duplicatedOutput, device, 0, true); + } else { + ALOGW("checkOutputsForDevice() could not open dup output for %d and %d", + mPrimaryOutput, output); + mpClientInterface->closeOutput(output); + mOutputs.removeItem(output); + output = 0; + } + } + } + if (output == 0) { + ALOGW("checkOutputsForDevice() could not open output for device %x", device); + delete desc; + profiles.removeAt(profile_index); + profile_index--; + } else { + outputs.add(output); + ALOGV("checkOutputsForDevice(): adding output %d", output); + } + } + + if (profiles.isEmpty()) { + ALOGW("checkOutputsForDevice(): No output available for device %04x", device); + return BAD_VALUE; + } + } else { + // check if one opened output is not needed any more after disconnecting one device + for (size_t i = 0; i < mOutputs.size(); i++) { + desc = mOutputs.valueAt(i); + if (!desc->isDuplicated() && + !(desc->mProfile->mSupportedDevices & mAvailableOutputDevices)) { + ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i)); + outputs.add(mOutputs.keyAt(i)); + } + } + for (size_t i = 0; i < mHwModules.size(); i++) + { + if (mHwModules[i]->mHandle == 0) { + continue; + } + for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) + { + IOProfile *profile = mHwModules[i]->mOutputProfiles[j]; + if ((profile->mSupportedDevices & device) && + (profile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { + ALOGV("checkOutputsForDevice(): clearing direct output profile %d on module %d", + j, i); + if (profile->mSamplingRates[0] == 0) { + profile->mSamplingRates.clear(); + profile->mSamplingRates.add(0); + } + if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { + profile->mFormats.clear(); + profile->mFormats.add(AUDIO_FORMAT_DEFAULT); + } + if (profile->mChannelMasks[0] == 0) { + profile->mChannelMasks.clear(); + profile->mChannelMasks.add(0); + } + } + } + } + } + return NO_ERROR; +} + +void AudioPolicyManagerBase::closeOutput(audio_io_handle_t output) +{ + ALOGV("closeOutput(%d)", output); + + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); + if (outputDesc == NULL) { + ALOGW("closeOutput() unknown output %d", output); + return; + } + + // look for duplicated outputs connected to the output being removed. + for (size_t i = 0; i < mOutputs.size(); i++) { + AudioOutputDescriptor *dupOutputDesc = mOutputs.valueAt(i); + if (dupOutputDesc->isDuplicated() && + (dupOutputDesc->mOutput1 == outputDesc || + dupOutputDesc->mOutput2 == outputDesc)) { + AudioOutputDescriptor *outputDesc2; + if (dupOutputDesc->mOutput1 == outputDesc) { + outputDesc2 = dupOutputDesc->mOutput2; + } else { + outputDesc2 = dupOutputDesc->mOutput1; + } + // As all active tracks on duplicated output will be deleted, + // and as they were also referenced on the other output, the reference + // count for their stream type must be adjusted accordingly on + // the other output. + for (int j = 0; j < (int)AudioSystem::NUM_STREAM_TYPES; j++) { + int refCount = dupOutputDesc->mRefCount[j]; + outputDesc2->changeRefCount((AudioSystem::stream_type)j,-refCount); + } + audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i); + ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput); + + mpClientInterface->closeOutput(duplicatedOutput); + delete mOutputs.valueFor(duplicatedOutput); + mOutputs.removeItem(duplicatedOutput); + } + } + + AudioParameter param; + param.add(String8("closing"), String8("true")); + mpClientInterface->setParameters(output, param.toString()); + + mpClientInterface->closeOutput(output); + delete outputDesc; + mOutputs.removeItem(output); + mPreviousOutputs = mOutputs; +} + +SortedVector<audio_io_handle_t> AudioPolicyManagerBase::getOutputsForDevice(audio_devices_t device, + DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs) +{ + SortedVector<audio_io_handle_t> outputs; + + ALOGVV("getOutputsForDevice() device %04x", device); + for (size_t i = 0; i < openOutputs.size(); i++) { + ALOGVV("output %d isDuplicated=%d device=%04x", + i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices()); + if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) { + ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i)); + outputs.add(openOutputs.keyAt(i)); + } + } + return outputs; +} + +bool AudioPolicyManagerBase::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1, + SortedVector<audio_io_handle_t>& outputs2) +{ + if (outputs1.size() != outputs2.size()) { + return false; + } + for (size_t i = 0; i < outputs1.size(); i++) { + if (outputs1[i] != outputs2[i]) { + return false; + } + } + return true; +} + +void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy) +{ + audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/); + audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/); + SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs); + SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs); + + if (!vectorsEqual(srcOutputs,dstOutputs)) { + ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d", + strategy, srcOutputs[0], dstOutputs[0]); + // mute strategy while moving tracks from one output to another + for (size_t i = 0; i < srcOutputs.size(); i++) { + AudioOutputDescriptor *desc = mOutputs.valueFor(srcOutputs[i]); + if (desc->isStrategyActive(strategy)) { + setStrategyMute(strategy, true, srcOutputs[i]); + setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice); + } + } + + // Move effects associated to this strategy from previous output to new output + if (strategy == STRATEGY_MEDIA) { + audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs); + SortedVector<audio_io_handle_t> moved; + for (size_t i = 0; i < mEffects.size(); i++) { + EffectDescriptor *desc = mEffects.valueAt(i); + if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX && + desc->mIo != fxOutput) { + if (moved.indexOf(desc->mIo) < 0) { + ALOGV("checkOutputForStrategy() moving effect %d to output %d", + mEffects.keyAt(i), fxOutput); + mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, desc->mIo, + fxOutput); + moved.add(desc->mIo); + } + desc->mIo = fxOutput; + } + } + } + // 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->invalidateStream((AudioSystem::stream_type)i); + } + } + } +} + +void AudioPolicyManagerBase::checkOutputForAllStrategies() +{ + checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); + checkOutputForStrategy(STRATEGY_PHONE); + checkOutputForStrategy(STRATEGY_SONIFICATION); + checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); + checkOutputForStrategy(STRATEGY_MEDIA); + checkOutputForStrategy(STRATEGY_DTMF); +} + +audio_io_handle_t AudioPolicyManagerBase::getA2dpOutput() +{ + if (!mHasA2dp) { + return 0; + } + + for (size_t i = 0; i < mOutputs.size(); i++) { + AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i); + if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) { + return mOutputs.keyAt(i); + } + } + + return 0; +} + +void AudioPolicyManagerBase::checkA2dpSuspend() +{ + if (!mHasA2dp) { + return; + } + audio_io_handle_t a2dpOutput = getA2dpOutput(); + if (a2dpOutput == 0) { + return; + } + + // 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 (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(a2dpOutput); + 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(a2dpOutput); + mA2dpSuspended = true; + } + } +} + +audio_devices_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache) +{ + audio_devices_t device = AUDIO_DEVICE_NONE; + + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); + // check the following by order of priority to request a routing change if necessary: + // 1: the strategy enforced audible is active on the output: + // use device for strategy enforced audible + // 2: we are in call or the strategy phone is active on the output: + // use device for strategy phone + // 3: the strategy sonification is active on the output: + // use device for strategy sonification + // 4: the strategy "respectful" sonification is active on the output: + // use device for strategy "respectful" sonification + // 5: the strategy media is active on the output: + // use device for strategy media + // 6: the strategy DTMF is active on the output: + // use device for strategy DTMF + if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) { + device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); + } else if (isInCall() || + outputDesc->isStrategyActive(STRATEGY_PHONE)) { + device = getDeviceForStrategy(STRATEGY_PHONE, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) { + device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) { + device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) { + device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) { + device = getDeviceForStrategy(STRATEGY_DTMF, fromCache); + } + + ALOGV("getNewDevice() selected device %x", device); + return device; +} + +uint32_t AudioPolicyManagerBase::getStrategyForStream(AudioSystem::stream_type stream) { + return (uint32_t)getStrategy(stream); +} + +audio_devices_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) { + audio_devices_t devices; + // By checking the range of stream before calling getStrategy, we avoid + // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE + // 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 = AUDIO_DEVICE_NONE; + } else { + AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream); + devices = getDeviceForStrategy(strategy, true /*fromCache*/); + } + 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::ALARM: + return STRATEGY_SONIFICATION; + case AudioSystem::NOTIFICATION: + return STRATEGY_SONIFICATION_RESPECTFUL; + case AudioSystem::DTMF: + return STRATEGY_DTMF; + default: + ALOGE("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; + case AudioSystem::ENFORCED_AUDIBLE: + return STRATEGY_ENFORCED_AUDIBLE; + } +} + +void AudioPolicyManagerBase::handleNotificationRoutingForStream(AudioSystem::stream_type stream) { + switch(stream) { + case AudioSystem::MUSIC: + checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); + updateDevicesAndOutputs(); + break; + default: + break; + } +} + +audio_devices_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy, + bool fromCache) +{ + uint32_t device = AUDIO_DEVICE_NONE; + + if (fromCache) { + ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x", + strategy, mDeviceForStrategy[strategy]); + return mDeviceForStrategy[strategy]; + } + + switch (strategy) { + + case STRATEGY_SONIFICATION_RESPECTFUL: + if (isInCall()) { + device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); + } else if (isStreamActiveRemotely(AudioSystem::MUSIC, + SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { + // while media is playing on a remote device, use the the sonification behavior. + // Note that we test this usecase before testing if media is playing because + // the isStreamActive() method only informs about the activity of a stream, not + // if it's for local playback. Note also that we use the same delay between both tests + device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); + } else if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { + // while media is playing (or has recently played), use the same device + device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); + } else { + // when media is not playing anymore, fall back on the sonification behavior + device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); + } + + break; + + case STRATEGY_DTMF: + if (!isInCall()) { + // when off call, DTMF strategy follows the same rules as MEDIA strategy + device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); + 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 & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT; + if (device) break; + } + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET; + if (device) break; + device = mAvailableOutputDevices & AUDIO_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 + // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP + if (mHasA2dp && !isInCall() && + (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) && + (getA2dpOutput() != 0) && !mA2dpSuspended) { + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; + if (device) break; + } + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET; + if (device) break; + if (mPhoneState != AudioSystem::MODE_IN_CALL) { + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; + if (device) break; + } + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_EARPIECE; + if (device) break; + device = mDefaultOutputDevice; + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE"); + } + break; + + case AudioSystem::FORCE_SPEAKER: + // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to + // A2DP speaker when forcing to speaker output + if (mHasA2dp && !isInCall() && + (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) && + (getA2dpOutput() != 0) && !mA2dpSuspended) { + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; + if (device) break; + } + if (mPhoneState != AudioSystem::MODE_IN_CALL) { + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; + if (device) break; + } + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER; + if (device) break; + device = mDefaultOutputDevice; + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER"); + } + 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 /*fromCache*/); + break; + } + // FALL THROUGH + + case STRATEGY_ENFORCED_AUDIBLE: + // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION + // except: + // - when in call where it doesn't default to STRATEGY_PHONE behavior + // - in countries where not enforced in which case it follows STRATEGY_MEDIA + + if ((strategy == STRATEGY_SONIFICATION) || + (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_SYSTEM_ENFORCED)) { + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER; + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION"); + } + } + // The second device used for sonification is the same as the device used by media strategy + // FALL THROUGH + + case STRATEGY_MEDIA: { + uint32_t device2 = AUDIO_DEVICE_NONE; + if (strategy != STRATEGY_SONIFICATION) { + // no sonification on remote submix (e.g. WFD) + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_REMOTE_SUBMIX; + } + if ((device2 == AUDIO_DEVICE_NONE) && + mHasA2dp && (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) && + (getA2dpOutput() != 0) && !mA2dpSuspended) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; + } + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; + } + if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) { + // no sonification on aux digital (e.g. HDMI) + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL; + } + if ((device2 == AUDIO_DEVICE_NONE) && + (mForceUse[AudioSystem::FOR_DOCK] == AudioSystem::FORCE_ANALOG_DOCK)) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER; + } + + // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or + // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise + device |= device2; + if (device) break; + device = mDefaultOutputDevice; + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA"); + } + } break; + + default: + ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy); + break; + } + + ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device); + return device; +} + +void AudioPolicyManagerBase::updateDevicesAndOutputs() +{ + for (int i = 0; i < NUM_STRATEGIES; i++) { + mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); + } + mPreviousOutputs = mOutputs; +} + +uint32_t AudioPolicyManagerBase::checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc, + audio_devices_t prevDevice, + uint32_t delayMs) +{ + // mute/unmute strategies using an incompatible device combination + // if muting, wait for the audio in pcm buffer to be drained before proceeding + // if unmuting, unmute only after the specified delay + if (outputDesc->isDuplicated()) { + return 0; + } + + uint32_t muteWaitMs = 0; + audio_devices_t device = outputDesc->device(); + bool shouldMute = outputDesc->isActive() && (AudioSystem::popCount(device) >= 2); + // temporary mute output if device selection changes to avoid volume bursts due to + // different per device volumes + bool tempMute = outputDesc->isActive() && (device != prevDevice); + + for (size_t i = 0; i < NUM_STRATEGIES; i++) { + audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); + bool mute = shouldMute && (curDevice & device) && (curDevice != device); + bool doMute = false; + + if (mute && !outputDesc->mStrategyMutedByDevice[i]) { + doMute = true; + outputDesc->mStrategyMutedByDevice[i] = true; + } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){ + doMute = true; + outputDesc->mStrategyMutedByDevice[i] = false; + } + if (doMute || tempMute) { + for (size_t j = 0; j < mOutputs.size(); j++) { + AudioOutputDescriptor *desc = mOutputs.valueAt(j); + // skip output if it does not share any device with current output + if ((desc->supportedDevices() & outputDesc->supportedDevices()) + == AUDIO_DEVICE_NONE) { + continue; + } + audio_io_handle_t curOutput = mOutputs.keyAt(j); + ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d", + mute ? "muting" : "unmuting", i, curDevice, curOutput); + setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs); + if (desc->isStrategyActive((routing_strategy)i)) { + // do tempMute only for current output + if (tempMute && (desc == outputDesc)) { + setStrategyMute((routing_strategy)i, true, curOutput); + setStrategyMute((routing_strategy)i, false, curOutput, + desc->latency() * 2, device); + } + if ((tempMute && (desc == outputDesc)) || mute) { + if (muteWaitMs < desc->latency()) { + muteWaitMs = desc->latency(); + } + } + } + } + } + } + + // FIXME: should not need to double latency if volume could be applied immediately by the + // audioflinger mixer. We must account for the delay between now and the next time + // the audioflinger thread for this output will process a buffer (which corresponds to + // one buffer size, usually 1/2 or 1/4 of the latency). + muteWaitMs *= 2; + // wait for the PCM output buffers to empty before proceeding with the rest of the command + if (muteWaitMs > delayMs) { + muteWaitMs -= delayMs; + usleep(muteWaitMs * 1000); + return muteWaitMs; + } + return 0; +} + +uint32_t AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output, + audio_devices_t device, + bool force, + int delayMs) +{ + ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs); + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); + AudioParameter param; + uint32_t muteWaitMs; + + if (outputDesc->isDuplicated()) { + muteWaitMs = setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs); + muteWaitMs += setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs); + return muteWaitMs; + } + // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current + // output profile + if ((device != AUDIO_DEVICE_NONE) && + ((device & outputDesc->mProfile->mSupportedDevices) == 0)) { + return 0; + } + + // filter devices according to output selected + device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices); + + audio_devices_t prevDevice = outputDesc->mDevice; + + ALOGV("setOutputDevice() prevDevice %04x", prevDevice); + + if (device != AUDIO_DEVICE_NONE) { + outputDesc->mDevice = device; + } + muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs); + + // Do not change the routing if: + // - the requested device is AUDIO_DEVICE_NONE + // - 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 == AUDIO_DEVICE_NONE || device == prevDevice) && !force) { + ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output); + return muteWaitMs; + } + + ALOGV("setOutputDevice() changing device"); + // do the routing + param.addInt(String8(AudioParameter::keyRouting), (int)device); + mpClientInterface->setParameters(output, param.toString(), delayMs); + + // update stream volumes according to new device + applyStreamVolumes(output, device, delayMs); + + return muteWaitMs; +} + +AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::getInputProfile(audio_devices_t device, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask) +{ + // Choose an input profile based on the requested capture parameters: select the first available + // profile supporting all requested parameters. + + for (size_t i = 0; i < mHwModules.size(); i++) + { + if (mHwModules[i]->mHandle == 0) { + continue; + } + for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) + { + IOProfile *profile = mHwModules[i]->mInputProfiles[j]; + if (profile->isCompatibleProfile(device, samplingRate, format, + channelMask, AUDIO_OUTPUT_FLAG_NONE)) { + return profile; + } + } + } + return NULL; +} + +audio_devices_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource) +{ + uint32_t device = AUDIO_DEVICE_NONE; + + switch (inputSource) { + case AUDIO_SOURCE_VOICE_UPLINK: + if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) { + device = AUDIO_DEVICE_IN_VOICE_CALL; + break; + } + // FALL THROUGH + + case AUDIO_SOURCE_DEFAULT: + case AUDIO_SOURCE_MIC: + case AUDIO_SOURCE_VOICE_RECOGNITION: + case AUDIO_SOURCE_HOTWORD: + case AUDIO_SOURCE_VOICE_COMMUNICATION: + if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO && + mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { + device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; + } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) { + device = AUDIO_DEVICE_IN_WIRED_HEADSET; + } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + case AUDIO_SOURCE_CAMCORDER: + if (mAvailableInputDevices & AUDIO_DEVICE_IN_BACK_MIC) { + device = AUDIO_DEVICE_IN_BACK_MIC; + } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + case AUDIO_SOURCE_VOICE_DOWNLINK: + case AUDIO_SOURCE_VOICE_CALL: + if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) { + device = AUDIO_DEVICE_IN_VOICE_CALL; + } + break; + case AUDIO_SOURCE_REMOTE_SUBMIX: + if (mAvailableInputDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { + device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; + } + break; + default: + ALOGW("getDeviceForInputSource() invalid input source %d", inputSource); + break; + } + ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device); + return device; +} + +bool AudioPolicyManagerBase::isVirtualInputDevice(audio_devices_t device) +{ + if ((device & AUDIO_DEVICE_BIT_IN) != 0) { + device &= ~AUDIO_DEVICE_BIT_IN; + if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0)) + return true; + } + return false; +} + +audio_io_handle_t AudioPolicyManagerBase::getActiveInput(bool ignoreVirtualInputs) +{ + for (size_t i = 0; i < mInputs.size(); i++) { + const AudioInputDescriptor * input_descriptor = mInputs.valueAt(i); + if ((input_descriptor->mRefCount > 0) + && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) { + return mInputs.keyAt(i); + } + } + return 0; +} + + +audio_devices_t AudioPolicyManagerBase::getDeviceForVolume(audio_devices_t device) +{ + if (device == AUDIO_DEVICE_NONE) { + // this happens when forcing a route update and no track is active on an output. + // In this case the returned category is not important. + device = AUDIO_DEVICE_OUT_SPEAKER; + } else if (AudioSystem::popCount(device) > 1) { + // Multiple device selection is either: + // - speaker + one other device: give priority to speaker in this case. + // - one A2DP device + another device: happens with duplicated output. In this case + // retain the device on the A2DP output as the other must not correspond to an active + // selection if not the speaker. + if (device & AUDIO_DEVICE_OUT_SPEAKER) { + device = AUDIO_DEVICE_OUT_SPEAKER; + } else { + device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP); + } + } + + ALOGW_IF(AudioSystem::popCount(device) != 1, + "getDeviceForVolume() invalid device combination: %08x", + device); + + return device; +} + +AudioPolicyManagerBase::device_category AudioPolicyManagerBase::getDeviceCategory(audio_devices_t device) +{ + switch(getDeviceForVolume(device)) { + case AUDIO_DEVICE_OUT_EARPIECE: + return DEVICE_CATEGORY_EARPIECE; + case AUDIO_DEVICE_OUT_WIRED_HEADSET: + case AUDIO_DEVICE_OUT_WIRED_HEADPHONE: + case AUDIO_DEVICE_OUT_BLUETOOTH_SCO: + case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET: + case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP: + case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES: + return DEVICE_CATEGORY_HEADSET; + case AUDIO_DEVICE_OUT_SPEAKER: + case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT: + case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER: + case AUDIO_DEVICE_OUT_AUX_DIGITAL: + case AUDIO_DEVICE_OUT_USB_ACCESSORY: + case AUDIO_DEVICE_OUT_USB_DEVICE: + case AUDIO_DEVICE_OUT_REMOTE_SUBMIX: + default: + return DEVICE_CATEGORY_SPEAKER; + } +} + +float AudioPolicyManagerBase::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, + int indexInUi) +{ + device_category deviceCategory = getDeviceCategory(device); + const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory]; + + // the volume index in the UI is relative to the min and max volume indices for this stream type + int nbSteps = 1 + curve[VOLMAX].mIndex - + curve[VOLMIN].mIndex; + 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 < curve[VOLMIN].mIndex) { // out of bounds + return 0.0f; + } else if (volIdx < curve[VOLKNEE1].mIndex) { + segment = 0; + } else if (volIdx < curve[VOLKNEE2].mIndex) { + segment = 1; + } else if (volIdx <= curve[VOLMAX].mIndex) { + segment = 2; + } else { // out of bounds + return 1.0f; + } + + // linear interpolation in the attenuation table in dB + float decibels = curve[segment].mDBAttenuation + + ((float)(volIdx - curve[segment].mIndex)) * + ( (curve[segment+1].mDBAttenuation - + curve[segment].mDBAttenuation) / + ((float)(curve[segment+1].mIndex - + curve[segment].mIndex)) ); + + float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 ) + + ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f", + curve[segment].mIndex, volIdx, + curve[segment+1].mIndex, + curve[segment].mDBAttenuation, + decibels, + curve[segment+1].mDBAttenuation, + amplification); + + return amplification; +} + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT] = { + {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f} +}; + +// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks +// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets. +// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java). +// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset. + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sDefaultSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sDefaultSystemVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT] = { + {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sHeadsetSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sDefaultVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sSpeakerVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + *AudioPolicyManagerBase::sVolumeProfiles[AUDIO_STREAM_CNT] + [AudioPolicyManagerBase::DEVICE_CATEGORY_CNT] = { + { // AUDIO_STREAM_VOICE_CALL + sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_SYSTEM + sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET + sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_RING + sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_MUSIC + sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_ALARM + sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_NOTIFICATION + sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_BLUETOOTH_SCO + sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_ENFORCED_AUDIBLE + sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET + sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_DTMF + sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET + sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_TTS + sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, +}; + +void AudioPolicyManagerBase::initializeVolumeCurves() +{ + for (int i = 0; i < AUDIO_STREAM_CNT; i++) { + for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { + mStreams[i].mVolumeCurve[j] = + sVolumeProfiles[i][j]; + } + } + + // Check availability of DRC on speaker path: if available, override some of the speaker curves + if (mSpeakerDrcEnabled) { + mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = + sDefaultSystemVolumeCurveDrc; + mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = + sSpeakerSonificationVolumeCurveDrc; + mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = + sSpeakerSonificationVolumeCurveDrc; + mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = + sSpeakerSonificationVolumeCurveDrc; + } +} + +float AudioPolicyManagerBase::computeVolume(int stream, + int index, + audio_io_handle_t output, + audio_devices_t device) +{ + float volume = 1.0; + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); + StreamDescriptor &streamDesc = mStreams[stream]; + + if (device == AUDIO_DEVICE_NONE) { + 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 == AUDIO_DEVICE_OUT_AUX_DIGITAL || + device == AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET || + device == AUDIO_DEVICE_OUT_USB_ACCESSORY || + device == AUDIO_DEVICE_OUT_USB_DEVICE)) { + 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. + const routing_strategy stream_strategy = getStrategy((AudioSystem::stream_type)stream); + if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | + AUDIO_DEVICE_OUT_WIRED_HEADSET | + AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) && + ((stream_strategy == STRATEGY_SONIFICATION) + || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL) + || (stream == AudioSystem::SYSTEM) + || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) && + (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) && + 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 (isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) || + mLimitRingtoneVolume) { + audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/); + float musicVol = computeVolume(AudioSystem::MUSIC, + mStreams[AudioSystem::MUSIC].getVolumeIndex(musicDevice), + output, + musicDevice); + float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? + musicVol : SONIFICATION_HEADSET_VOLUME_MIN; + if (volume > minVol) { + volume = minVol; + ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol); + } + } + } + + return volume; +} + +status_t AudioPolicyManagerBase::checkAndSetVolume(int stream, + int index, + audio_io_handle_t output, + audio_devices_t device, + int delayMs, + bool force) +{ + + // do not change actual stream volume if the stream is muted + if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) { + ALOGVV("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)) { + ALOGV("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; + ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs); + // 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 == mPrimaryOutput) { + mpClientInterface->setVoiceVolume(voiceVolume, delayMs); + mLastVoiceVolume = voiceVolume; + } + } + + return NO_ERROR; +} + +void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output, + audio_devices_t device, + int delayMs, + bool force) +{ + ALOGVV("applyStreamVolumes() for output %d and device %x", output, device); + + for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { + checkAndSetVolume(stream, + mStreams[stream].getVolumeIndex(device), + output, + device, + delayMs, + force); + } +} + +void AudioPolicyManagerBase::setStrategyMute(routing_strategy strategy, + bool on, + audio_io_handle_t output, + int delayMs, + audio_devices_t device) +{ + ALOGVV("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, device); + } + } +} + +void AudioPolicyManagerBase::setStreamMute(int stream, + bool on, + audio_io_handle_t output, + int delayMs, + audio_devices_t device) +{ + StreamDescriptor &streamDesc = mStreams[stream]; + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); + if (device == AUDIO_DEVICE_NONE) { + device = outputDesc->device(); + } + + ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x", + stream, on, output, outputDesc->mMuteCount[stream], device); + + if (on) { + if (outputDesc->mMuteCount[stream] == 0) { + if (streamDesc.mCanBeMuted && + ((stream != AudioSystem::ENFORCED_AUDIBLE) || + (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) { + checkAndSetVolume(stream, 0, output, device, delayMs); + } + } + // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored + outputDesc->mMuteCount[stream]++; + } else { + if (outputDesc->mMuteCount[stream] == 0) { + ALOGV("setStreamMute() unmuting non muted stream!"); + return; + } + if (--outputDesc->mMuteCount[stream] == 0) { + checkAndSetVolume(stream, + streamDesc.getVolumeIndex(device), + output, + 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 + const routing_strategy stream_strategy = getStrategy((AudioSystem::stream_type)stream); + if ((stream_strategy == STRATEGY_SONIFICATION) || + ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) { + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput); + ALOGV("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)) { + ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount); + for (int i = 0; i < muteCount; i++) { + setStreamMute(stream, starting, mPrimaryOutput); + } + } else { + ALOGV("handleIncallSonification() high visibility"); + if (outputDesc->device() & + getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) { + ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount); + for (int i = 0; i < muteCount; i++) { + setStreamMute(stream, starting, mPrimaryOutput); + } + } + 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)); +} + +uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad() +{ + return MAX_EFFECTS_CPU_LOAD; +} + +uint32_t AudioPolicyManagerBase::getMaxEffectsMemory() +{ + return MAX_EFFECTS_MEMORY; +} + +// --- AudioOutputDescriptor class implementation + +AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor( + const IOProfile *profile) + : mId(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), + mChannelMask(0), mLatency(0), + mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), + mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(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; + } + for (int i = 0; i < NUM_STRATEGIES; i++) { + mStrategyMutedByDevice[i] = false; + } + if (profile != NULL) { + mSamplingRate = profile->mSamplingRates[0]; + mFormat = profile->mFormats[0]; + mChannelMask = profile->mChannelMasks[0]; + mFlags = profile->mFlags; + } +} + +audio_devices_t AudioPolicyManagerBase::AudioOutputDescriptor::device() const +{ + if (isDuplicated()) { + return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice); + } else { + return mDevice; + } +} + +uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::latency() +{ + if (isDuplicated()) { + return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency; + } else { + return mLatency; + } +} + +bool AudioPolicyManagerBase::AudioOutputDescriptor::sharesHwModuleWith( + const AudioOutputDescriptor *outputDesc) +{ + if (isDuplicated()) { + return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc); + } else if (outputDesc->isDuplicated()){ + return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2); + } else { + return (mProfile->mModule == outputDesc->mProfile->mModule); + } +} + +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) { + ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]); + mRefCount[stream] = 0; + return; + } + mRefCount[stream] += delta; + ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]); +} + +audio_devices_t AudioPolicyManagerBase::AudioOutputDescriptor::supportedDevices() +{ + if (isDuplicated()) { + return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices()); + } else { + return mProfile->mSupportedDevices ; + } +} + +bool AudioPolicyManagerBase::AudioOutputDescriptor::isActive(uint32_t inPastMs) const +{ + return isStrategyActive(NUM_STRATEGIES, inPastMs); +} + +bool AudioPolicyManagerBase::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy, + uint32_t inPastMs, + nsecs_t sysTime) const +{ + if ((sysTime == 0) && (inPastMs != 0)) { + sysTime = systemTime(); + } + for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { + if (((getStrategy((AudioSystem::stream_type)i) == strategy) || + (NUM_STRATEGIES == strategy)) && + isStreamActive((AudioSystem::stream_type)i, inPastMs, sysTime)) { + return true; + } + } + return false; +} + +bool AudioPolicyManagerBase::AudioOutputDescriptor::isStreamActive(AudioSystem::stream_type stream, + uint32_t inPastMs, + nsecs_t sysTime) const +{ + if (mRefCount[stream] != 0) { + return true; + } + if (inPastMs == 0) { + return false; + } + if (sysTime == 0) { + sysTime = systemTime(); + } + if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) { + return true; + } + return false; +} + + +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: %08x\n", mFormat); + result.append(buffer); + snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask); + 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(const IOProfile *profile) + : mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(0), + mDevice(AUDIO_DEVICE_NONE), mRefCount(0), + mInputSource(0), mProfile(profile) +{ +} + +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", mChannelMask); + 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 + +AudioPolicyManagerBase::StreamDescriptor::StreamDescriptor() + : mIndexMin(0), mIndexMax(1), mCanBeMuted(true) +{ + mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0); +} + +int AudioPolicyManagerBase::StreamDescriptor::getVolumeIndex(audio_devices_t device) +{ + device = AudioPolicyManagerBase::getDeviceForVolume(device); + // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT + if (mIndexCur.indexOfKey(device) < 0) { + device = AUDIO_DEVICE_OUT_DEFAULT; + } + return mIndexCur.valueFor(device); +} + +void AudioPolicyManagerBase::StreamDescriptor::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "%s %02d %02d ", + mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax); + result.append(buffer); + for (size_t i = 0; i < mIndexCur.size(); i++) { + snprintf(buffer, SIZE, "%04x : %02d, ", + mIndexCur.keyAt(i), + mIndexCur.valueAt(i)); + result.append(buffer); + } + result.append("\n"); + + write(fd, result.string(), result.size()); +} + +// --- EffectDescriptor class implementation + +status_t AudioPolicyManagerBase::EffectDescriptor::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " I/O: %d\n", mIo); + 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); + snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled"); + result.append(buffer); + write(fd, result.string(), result.size()); + + return NO_ERROR; +} + +// --- IOProfile class implementation + +AudioPolicyManagerBase::HwModule::HwModule(const char *name) + : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), mHandle(0) +{ +} + +AudioPolicyManagerBase::HwModule::~HwModule() +{ + for (size_t i = 0; i < mOutputProfiles.size(); i++) { + delete mOutputProfiles[i]; + } + for (size_t i = 0; i < mInputProfiles.size(); i++) { + delete mInputProfiles[i]; + } + free((void *)mName); +} + +void AudioPolicyManagerBase::HwModule::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " - name: %s\n", mName); + result.append(buffer); + snprintf(buffer, SIZE, " - handle: %d\n", mHandle); + result.append(buffer); + write(fd, result.string(), result.size()); + if (mOutputProfiles.size()) { + write(fd, " - outputs:\n", strlen(" - outputs:\n")); + for (size_t i = 0; i < mOutputProfiles.size(); i++) { + snprintf(buffer, SIZE, " output %d:\n", i); + write(fd, buffer, strlen(buffer)); + mOutputProfiles[i]->dump(fd); + } + } + if (mInputProfiles.size()) { + write(fd, " - inputs:\n", strlen(" - inputs:\n")); + for (size_t i = 0; i < mInputProfiles.size(); i++) { + snprintf(buffer, SIZE, " input %d:\n", i); + write(fd, buffer, strlen(buffer)); + mInputProfiles[i]->dump(fd); + } + } +} + +AudioPolicyManagerBase::IOProfile::IOProfile(HwModule *module) + : mFlags((audio_output_flags_t)0), mModule(module) +{ +} + +AudioPolicyManagerBase::IOProfile::~IOProfile() +{ +} + +// checks if the IO profile is compatible with specified parameters. +// Sampling rate, format and channel mask must be specified in order to +// get a valid a match +bool AudioPolicyManagerBase::IOProfile::isCompatibleProfile(audio_devices_t device, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags) const +{ + if (samplingRate == 0 || !audio_is_valid_format(format) || channelMask == 0) { + return false; + } + + if ((mSupportedDevices & device) != device) { + return false; + } + if ((mFlags & flags) != flags) { + return false; + } + size_t i; + for (i = 0; i < mSamplingRates.size(); i++) + { + if (mSamplingRates[i] == samplingRate) { + break; + } + } + if (i == mSamplingRates.size()) { + return false; + } + for (i = 0; i < mFormats.size(); i++) + { + if (mFormats[i] == format) { + break; + } + } + if (i == mFormats.size()) { + return false; + } + for (i = 0; i < mChannelMasks.size(); i++) + { + if (mChannelMasks[i] == channelMask) { + break; + } + } + if (i == mChannelMasks.size()) { + return false; + } + return true; +} + +void AudioPolicyManagerBase::IOProfile::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " - sampling rates: "); + result.append(buffer); + for (size_t i = 0; i < mSamplingRates.size(); i++) { + snprintf(buffer, SIZE, "%d", mSamplingRates[i]); + result.append(buffer); + result.append(i == (mSamplingRates.size() - 1) ? "\n" : ", "); + } + + snprintf(buffer, SIZE, " - channel masks: "); + result.append(buffer); + for (size_t i = 0; i < mChannelMasks.size(); i++) { + snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]); + result.append(buffer); + result.append(i == (mChannelMasks.size() - 1) ? "\n" : ", "); + } + + snprintf(buffer, SIZE, " - formats: "); + result.append(buffer); + for (size_t i = 0; i < mFormats.size(); i++) { + snprintf(buffer, SIZE, "0x%08x", mFormats[i]); + result.append(buffer); + result.append(i == (mFormats.size() - 1) ? "\n" : ", "); + } + + snprintf(buffer, SIZE, " - devices: 0x%04x\n", mSupportedDevices); + result.append(buffer); + snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags); + result.append(buffer); + + write(fd, result.string(), result.size()); +} + +// --- audio_policy.conf file parsing + +struct StringToEnum { + const char *name; + uint32_t value; +}; + +#define STRING_TO_ENUM(string) { #string, string } +#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) + +const struct StringToEnum sDeviceNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX), + STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC), + STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL), + STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL), + STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC), + STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX), + STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY), +}; + +const struct StringToEnum sFlagNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING), +}; + +const struct StringToEnum sFormatNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED), + STRING_TO_ENUM(AUDIO_FORMAT_MP3), + STRING_TO_ENUM(AUDIO_FORMAT_AAC), + STRING_TO_ENUM(AUDIO_FORMAT_VORBIS), +}; + +const struct StringToEnum sOutChannelsNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO), + STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO), + STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1), + STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1), +}; + +const struct StringToEnum sInChannelsNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO), + STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO), + STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK), +}; + + +uint32_t AudioPolicyManagerBase::stringToEnum(const struct StringToEnum *table, + size_t size, + const char *name) +{ + for (size_t i = 0; i < size; i++) { + if (strcmp(table[i].name, name) == 0) { + ALOGV("stringToEnum() found %s", table[i].name); + return table[i].value; + } + } + return 0; +} + +bool AudioPolicyManagerBase::stringToBool(const char *value) +{ + return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0)); +} + +audio_output_flags_t AudioPolicyManagerBase::parseFlagNames(char *name) +{ + uint32_t flag = 0; + + // it is OK to cast name to non const here as we are not going to use it after + // strtok() modifies it + char *flagName = strtok(name, "|"); + while (flagName != NULL) { + if (strlen(flagName) != 0) { + flag |= stringToEnum(sFlagNameToEnumTable, + ARRAY_SIZE(sFlagNameToEnumTable), + flagName); + } + flagName = strtok(NULL, "|"); + } + //force direct flag if offload flag is set: offloading implies a direct output stream + // and all common behaviors are driven by checking only the direct flag + // this should normally be set appropriately in the policy configuration file + if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { + flag |= AUDIO_OUTPUT_FLAG_DIRECT; + } + + return (audio_output_flags_t)flag; +} + +audio_devices_t AudioPolicyManagerBase::parseDeviceNames(char *name) +{ + uint32_t device = 0; + + char *devName = strtok(name, "|"); + while (devName != NULL) { + if (strlen(devName) != 0) { + device |= stringToEnum(sDeviceNameToEnumTable, + ARRAY_SIZE(sDeviceNameToEnumTable), + devName); + } + devName = strtok(NULL, "|"); + } + return device; +} + +void AudioPolicyManagerBase::loadSamplingRates(char *name, IOProfile *profile) +{ + char *str = strtok(name, "|"); + + // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling + // rates should be read from the output stream after it is opened for the first time + if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { + profile->mSamplingRates.add(0); + return; + } + + while (str != NULL) { + uint32_t rate = atoi(str); + if (rate != 0) { + ALOGV("loadSamplingRates() adding rate %d", rate); + profile->mSamplingRates.add(rate); + } + str = strtok(NULL, "|"); + } + return; +} + +void AudioPolicyManagerBase::loadFormats(char *name, IOProfile *profile) +{ + char *str = strtok(name, "|"); + + // by convention, "0' in the first entry in mFormats indicates the supported formats + // should be read from the output stream after it is opened for the first time + if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { + profile->mFormats.add(AUDIO_FORMAT_DEFAULT); + return; + } + + while (str != NULL) { + audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable, + ARRAY_SIZE(sFormatNameToEnumTable), + str); + if (format != AUDIO_FORMAT_DEFAULT) { + profile->mFormats.add(format); + } + str = strtok(NULL, "|"); + } + return; +} + +void AudioPolicyManagerBase::loadInChannels(char *name, IOProfile *profile) +{ + const char *str = strtok(name, "|"); + + ALOGV("loadInChannels() %s", name); + + if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { + profile->mChannelMasks.add(0); + return; + } + + while (str != NULL) { + audio_channel_mask_t channelMask = + (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable, + ARRAY_SIZE(sInChannelsNameToEnumTable), + str); + if (channelMask != 0) { + ALOGV("loadInChannels() adding channelMask %04x", channelMask); + profile->mChannelMasks.add(channelMask); + } + str = strtok(NULL, "|"); + } + return; +} + +void AudioPolicyManagerBase::loadOutChannels(char *name, IOProfile *profile) +{ + const char *str = strtok(name, "|"); + + ALOGV("loadOutChannels() %s", name); + + // by convention, "0' in the first entry in mChannelMasks indicates the supported channel + // masks should be read from the output stream after it is opened for the first time + if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { + profile->mChannelMasks.add(0); + return; + } + + while (str != NULL) { + audio_channel_mask_t channelMask = + (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable, + ARRAY_SIZE(sOutChannelsNameToEnumTable), + str); + if (channelMask != 0) { + profile->mChannelMasks.add(channelMask); + } + str = strtok(NULL, "|"); + } + return; +} + +status_t AudioPolicyManagerBase::loadInput(cnode *root, HwModule *module) +{ + cnode *node = root->first_child; + + IOProfile *profile = new IOProfile(module); + + while (node) { + if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { + loadSamplingRates((char *)node->value, profile); + } else if (strcmp(node->name, FORMATS_TAG) == 0) { + loadFormats((char *)node->value, profile); + } else if (strcmp(node->name, CHANNELS_TAG) == 0) { + loadInChannels((char *)node->value, profile); + } else if (strcmp(node->name, DEVICES_TAG) == 0) { + profile->mSupportedDevices = parseDeviceNames((char *)node->value); + } + node = node->next; + } + ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE, + "loadInput() invalid supported devices"); + ALOGW_IF(profile->mChannelMasks.size() == 0, + "loadInput() invalid supported channel masks"); + ALOGW_IF(profile->mSamplingRates.size() == 0, + "loadInput() invalid supported sampling rates"); + ALOGW_IF(profile->mFormats.size() == 0, + "loadInput() invalid supported formats"); + if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) && + (profile->mChannelMasks.size() != 0) && + (profile->mSamplingRates.size() != 0) && + (profile->mFormats.size() != 0)) { + + ALOGV("loadInput() adding input mSupportedDevices %04x", profile->mSupportedDevices); + + module->mInputProfiles.add(profile); + return NO_ERROR; + } else { + delete profile; + return BAD_VALUE; + } +} + +status_t AudioPolicyManagerBase::loadOutput(cnode *root, HwModule *module) +{ + cnode *node = root->first_child; + + IOProfile *profile = new IOProfile(module); + + while (node) { + if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { + loadSamplingRates((char *)node->value, profile); + } else if (strcmp(node->name, FORMATS_TAG) == 0) { + loadFormats((char *)node->value, profile); + } else if (strcmp(node->name, CHANNELS_TAG) == 0) { + loadOutChannels((char *)node->value, profile); + } else if (strcmp(node->name, DEVICES_TAG) == 0) { + profile->mSupportedDevices = parseDeviceNames((char *)node->value); + } else if (strcmp(node->name, FLAGS_TAG) == 0) { + profile->mFlags = parseFlagNames((char *)node->value); + } + node = node->next; + } + ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE, + "loadOutput() invalid supported devices"); + ALOGW_IF(profile->mChannelMasks.size() == 0, + "loadOutput() invalid supported channel masks"); + ALOGW_IF(profile->mSamplingRates.size() == 0, + "loadOutput() invalid supported sampling rates"); + ALOGW_IF(profile->mFormats.size() == 0, + "loadOutput() invalid supported formats"); + if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) && + (profile->mChannelMasks.size() != 0) && + (profile->mSamplingRates.size() != 0) && + (profile->mFormats.size() != 0)) { + + ALOGV("loadOutput() adding output mSupportedDevices %04x, mFlags %04x", + profile->mSupportedDevices, profile->mFlags); + + module->mOutputProfiles.add(profile); + return NO_ERROR; + } else { + delete profile; + return BAD_VALUE; + } +} + +void AudioPolicyManagerBase::loadHwModule(cnode *root) +{ + cnode *node = config_find(root, OUTPUTS_TAG); + status_t status = NAME_NOT_FOUND; + + HwModule *module = new HwModule(root->name); + + if (node != NULL) { + if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) { + mHasA2dp = true; + } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) { + mHasUsb = true; + } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) { + mHasRemoteSubmix = true; + } + + node = node->first_child; + while (node) { + ALOGV("loadHwModule() loading output %s", node->name); + status_t tmpStatus = loadOutput(node, module); + if (status == NAME_NOT_FOUND || status == NO_ERROR) { + status = tmpStatus; + } + node = node->next; + } + } + node = config_find(root, INPUTS_TAG); + if (node != NULL) { + node = node->first_child; + while (node) { + ALOGV("loadHwModule() loading input %s", node->name); + status_t tmpStatus = loadInput(node, module); + if (status == NAME_NOT_FOUND || status == NO_ERROR) { + status = tmpStatus; + } + node = node->next; + } + } + if (status == NO_ERROR) { + mHwModules.add(module); + } else { + delete module; + } +} + +void AudioPolicyManagerBase::loadHwModules(cnode *root) +{ + cnode *node = config_find(root, AUDIO_HW_MODULE_TAG); + if (node == NULL) { + return; + } + + node = node->first_child; + while (node) { + ALOGV("loadHwModules() loading module %s", node->name); + loadHwModule(node); + node = node->next; + } +} + +void AudioPolicyManagerBase::loadGlobalConfig(cnode *root) +{ + cnode *node = config_find(root, GLOBAL_CONFIG_TAG); + if (node == NULL) { + return; + } + node = node->first_child; + while (node) { + if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) { + mAttachedOutputDevices = parseDeviceNames((char *)node->value); + ALOGW_IF(mAttachedOutputDevices == AUDIO_DEVICE_NONE, + "loadGlobalConfig() no attached output devices"); + ALOGV("loadGlobalConfig() mAttachedOutputDevices %04x", mAttachedOutputDevices); + } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) { + mDefaultOutputDevice = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable, + ARRAY_SIZE(sDeviceNameToEnumTable), + (char *)node->value); + ALOGW_IF(mDefaultOutputDevice == AUDIO_DEVICE_NONE, + "loadGlobalConfig() default device not specified"); + ALOGV("loadGlobalConfig() mDefaultOutputDevice %04x", mDefaultOutputDevice); + } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) { + mAvailableInputDevices = parseDeviceNames((char *)node->value) & ~AUDIO_DEVICE_BIT_IN; + ALOGV("loadGlobalConfig() mAvailableInputDevices %04x", mAvailableInputDevices); + } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) { + mSpeakerDrcEnabled = stringToBool((char *)node->value); + ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled); + } + node = node->next; + } +} + +status_t AudioPolicyManagerBase::loadAudioPolicyConfig(const char *path) +{ + cnode *root; + char *data; + + data = (char *)load_file(path, NULL); + if (data == NULL) { + return -ENODEV; + } + root = config_node("", ""); + config_load(root, data); + + loadGlobalConfig(root); + loadHwModules(root); + + config_free(root); + free(root); + free(data); + + ALOGI("loadAudioPolicyConfig() loaded %s\n", path); + + return NO_ERROR; +} + +void AudioPolicyManagerBase::defaultAudioPolicyConfig(void) +{ + HwModule *module; + IOProfile *profile; + + mDefaultOutputDevice = AUDIO_DEVICE_OUT_SPEAKER; + mAttachedOutputDevices = AUDIO_DEVICE_OUT_SPEAKER; + mAvailableInputDevices = AUDIO_DEVICE_IN_BUILTIN_MIC & ~AUDIO_DEVICE_BIT_IN; + + module = new HwModule("primary"); + + profile = new IOProfile(module); + profile->mSamplingRates.add(44100); + profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); + profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO); + profile->mSupportedDevices = AUDIO_DEVICE_OUT_SPEAKER; + profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY; + module->mOutputProfiles.add(profile); + + profile = new IOProfile(module); + profile->mSamplingRates.add(8000); + profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); + profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO); + profile->mSupportedDevices = AUDIO_DEVICE_IN_BUILTIN_MIC; + module->mInputProfiles.add(profile); + + mHwModules.add(module); +} + +}; // namespace android diff --git a/services/audiopolicy/AudioPolicyManagerBase.h b/services/audiopolicy/AudioPolicyManagerBase.h new file mode 100644 index 0000000..1ff409e --- /dev/null +++ b/services/audiopolicy/AudioPolicyManagerBase.h @@ -0,0 +1,587 @@ +/* + * 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. + */ + + +#include <stdint.h> +#include <sys/types.h> +#include <cutils/config_utils.h> +#include <cutils/misc.h> +#include <utils/Timers.h> +#include <utils/Errors.h> +#include <utils/KeyedVector.h> +#include <utils/SortedVector.h> +#include <hardware_legacy/AudioPolicyInterface.h> + + +namespace android_audio_legacy { + using android::KeyedVector; + using android::DefaultKeyedVector; + using android::SortedVector; + +// ---------------------------------------------------------------------------- + +#define MAX_DEVICE_ADDRESS_LEN 20 +// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB +#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5 +// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB +#define SONIFICATION_HEADSET_VOLUME_MIN 0.016 +// Time in milliseconds during which we consider that music is still active after a music +// track was stopped - see computeVolume() +#define SONIFICATION_HEADSET_MUSIC_DELAY 5000 +// Time in milliseconds after media stopped playing during which we consider that the +// sonification should be as unobtrusive as during the time media was playing. +#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000 +// Time in milliseconds during witch some streams are muted while the audio path +// is switched +#define MUTE_TIME_MS 2000 + +#define NUM_TEST_OUTPUTS 5 + +#define NUM_VOL_CURVE_KNEES 2 + +// Default minimum length allowed for offloading a compressed track +// Can be overridden by the audio.offload.min.duration.secs property +#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60 + +// ---------------------------------------------------------------------------- +// AudioPolicyManagerBase implements audio policy manager behavior common to all platforms. +// Each platform must implement an AudioPolicyManager class derived from AudioPolicyManagerBase +// and override methods for which the platform specific behavior differs from the implementation +// in AudioPolicyManagerBase. Even if no specific behavior is required, the AudioPolicyManager +// class must be implemented as well as the class factory function createAudioPolicyManager() +// and provided in a shared library libaudiopolicy.so. +// ---------------------------------------------------------------------------- + +class AudioPolicyManagerBase: public AudioPolicyInterface +#ifdef AUDIO_POLICY_TEST + , public Thread +#endif //AUDIO_POLICY_TEST +{ + +public: + AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface); + virtual ~AudioPolicyManagerBase(); + + // AudioPolicyInterface + virtual status_t setDeviceConnectionState(audio_devices_t device, + AudioSystem::device_connection_state state, + const char *device_address); + virtual AudioSystem::device_connection_state getDeviceConnectionState(audio_devices_t device, + const char *device_address); + virtual void setPhoneState(int state); + virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config); + virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage); + virtual void setSystemProperty(const char* property, const char* value); + virtual status_t initCheck(); + virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + AudioSystem::output_flags flags, + const audio_offload_info_t *offloadInfo); + virtual status_t startOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int session = 0); + virtual status_t stopOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int session = 0); + virtual void releaseOutput(audio_io_handle_t output); + virtual audio_io_handle_t getInput(int inputSource, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + AudioSystem::audio_in_acoustics acoustics); + + // indicates to the audio policy manager that the input starts being used. + virtual status_t startInput(audio_io_handle_t input); + + // indicates to the audio policy manager that the input stops being used. + virtual status_t stopInput(audio_io_handle_t input); + virtual void releaseInput(audio_io_handle_t input); + virtual void initStreamVolume(AudioSystem::stream_type stream, + int indexMin, + int indexMax); + virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, + int index, + audio_devices_t device); + virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, + int *index, + audio_devices_t device); + + // return the strategy corresponding to a given stream type + virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream); + + // return the enabled output devices for the given stream type + virtual audio_devices_t getDevicesForStream(AudioSystem::stream_type stream); + + virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL); + virtual status_t registerEffect(const effect_descriptor_t *desc, + audio_io_handle_t io, + uint32_t strategy, + int session, + int id); + virtual status_t unregisterEffect(int id); + virtual status_t setEffectEnabled(int id, bool enabled); + + virtual bool isStreamActive(int stream, uint32_t inPastMs = 0) const; + // return whether a stream is playing remotely, override to change the definition of + // local/remote playback, used for instance by notification manager to not make + // media players lose audio focus when not playing locally + virtual bool isStreamActiveRemotely(int stream, uint32_t inPastMs = 0) const; + virtual bool isSourceActive(audio_source_t source) const; + + virtual status_t dump(int fd); + + virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo); + +protected: + + enum routing_strategy { + STRATEGY_MEDIA, + STRATEGY_PHONE, + STRATEGY_SONIFICATION, + STRATEGY_SONIFICATION_RESPECTFUL, + STRATEGY_DTMF, + STRATEGY_ENFORCED_AUDIBLE, + NUM_STRATEGIES + }; + + // 4 points to define the volume attenuation curve, each characterized by the volume + // index (from 0 to 100) at which they apply, and the attenuation in dB at that index. + // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl() + + enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4}; + + class VolumeCurvePoint + { + public: + int mIndex; + float mDBAttenuation; + }; + + // device categories used for volume curve management. + enum device_category { + DEVICE_CATEGORY_HEADSET, + DEVICE_CATEGORY_SPEAKER, + DEVICE_CATEGORY_EARPIECE, + DEVICE_CATEGORY_CNT + }; + + class IOProfile; + + class HwModule { + public: + HwModule(const char *name); + ~HwModule(); + + void dump(int fd); + + const char *const mName; // base name of the audio HW module (primary, a2dp ...) + audio_module_handle_t mHandle; + Vector <IOProfile *> mOutputProfiles; // output profiles exposed by this module + Vector <IOProfile *> mInputProfiles; // input profiles exposed by this module + }; + + // the IOProfile class describes the capabilities of an output or input stream. + // It is currently assumed that all combination of listed parameters are supported. + // It is used by the policy manager to determine if an output or input is suitable for + // a given use case, open/close it accordingly and connect/disconnect audio tracks + // to/from it. + class IOProfile + { + public: + IOProfile(HwModule *module); + ~IOProfile(); + + bool isCompatibleProfile(audio_devices_t device, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags) const; + + void dump(int fd); + + // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats + // indicates the supported parameters should be read from the output stream + // after it is opened for the first time + Vector <uint32_t> mSamplingRates; // supported sampling rates + Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks + Vector <audio_format_t> mFormats; // supported audio formats + audio_devices_t mSupportedDevices; // supported devices (devices this output can be + // routed to) + audio_output_flags_t mFlags; // attribute flags (e.g primary output, + // direct output...). For outputs only. + HwModule *mModule; // audio HW module exposing this I/O stream + }; + + // default volume curve + static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + // default volume curve for media strategy + static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + // volume curve for media strategy on speakers + static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + // volume curve for sonification strategy on speakers + static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT]; + static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT]; + static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + // default volume curves per stream and device category. See initializeVolumeCurves() + static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT]; + + // descriptor for audio outputs. Used to maintain current configuration of each opened audio output + // and keep track of the usage of this output by each audio stream type. + class AudioOutputDescriptor + { + public: + AudioOutputDescriptor(const IOProfile *profile); + + status_t dump(int fd); + + audio_devices_t device() const; + void changeRefCount(AudioSystem::stream_type stream, int delta); + + bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); } + audio_devices_t supportedDevices(); + uint32_t latency(); + bool sharesHwModuleWith(const AudioOutputDescriptor *outputDesc); + bool isActive(uint32_t inPastMs = 0) const; + bool isStreamActive(AudioSystem::stream_type stream, + uint32_t inPastMs = 0, + nsecs_t sysTime = 0) const; + bool isStrategyActive(routing_strategy strategy, + uint32_t inPastMs = 0, + nsecs_t sysTime = 0) const; + + audio_io_handle_t mId; // output handle + uint32_t mSamplingRate; // + audio_format_t mFormat; // + audio_channel_mask_t mChannelMask; // output configuration + uint32_t mLatency; // + audio_output_flags_t mFlags; // + audio_devices_t mDevice; // current device this output is routed to + uint32_t mRefCount[AudioSystem::NUM_STREAM_TYPES]; // number of streams of each type using this output + nsecs_t mStopTime[AudioSystem::NUM_STREAM_TYPES]; + AudioOutputDescriptor *mOutput1; // used by duplicated outputs: first output + AudioOutputDescriptor *mOutput2; // used by duplicated outputs: second output + float mCurVolume[AudioSystem::NUM_STREAM_TYPES]; // current stream volume + int mMuteCount[AudioSystem::NUM_STREAM_TYPES]; // mute request counter + const IOProfile *mProfile; // I/O profile this output derives from + bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible + // device selection. See checkDeviceMuteStrategies() + uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only) + }; + + // descriptor for audio inputs. Used to maintain current configuration of each opened audio input + // and keep track of the usage of this input. + class AudioInputDescriptor + { + public: + AudioInputDescriptor(const IOProfile *profile); + + status_t dump(int fd); + + uint32_t mSamplingRate; // + audio_format_t mFormat; // input configuration + audio_channel_mask_t mChannelMask; // + audio_devices_t mDevice; // current device this input is routed to + uint32_t mRefCount; // number of AudioRecord clients using this output + int mInputSource; // input source selected by application (mediarecorder.h) + const IOProfile *mProfile; // I/O profile this output derives from + }; + + // stream descriptor used for volume control + class StreamDescriptor + { + public: + StreamDescriptor(); + + int getVolumeIndex(audio_devices_t device); + void dump(int fd); + + int mIndexMin; // min volume index + int mIndexMax; // max volume index + KeyedVector<audio_devices_t, int> mIndexCur; // current volume index per device + bool mCanBeMuted; // true is the stream can be muted + + const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT]; + }; + + // stream descriptor used for volume control + class EffectDescriptor + { + public: + + status_t dump(int fd); + + int mIo; // io the effect is attached to + routing_strategy mStrategy; // routing strategy the effect is associated to + int mSession; // audio session the effect is on + effect_descriptor_t mDesc; // effect descriptor + bool mEnabled; // enabled state: CPU load being used or not + }; + + void addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc); + + // return the strategy corresponding to a given stream type + static routing_strategy getStrategy(AudioSystem::stream_type stream); + + // return appropriate device for streams handled by the specified strategy according to current + // phone state, connected devices... + // if fromCache is true, the device is returned from mDeviceForStrategy[], + // otherwise it is determine by current state + // (device connected,phone state, force use, a2dp output...) + // This allows to: + // 1 speed up process when the state is stable (when starting or stopping an output) + // 2 access to either current device selection (fromCache == true) or + // "future" device selection (fromCache == false) when called from a context + // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND + // before updateDevicesAndOutputs() is called. + virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy, + bool fromCache); + + // change the route of the specified output. Returns the number of ms we have slept to + // allow new routing to take effect in certain cases. + uint32_t setOutputDevice(audio_io_handle_t output, + audio_devices_t device, + bool force = false, + int delayMs = 0); + + // select input device corresponding to requested audio source + virtual audio_devices_t getDeviceForInputSource(int inputSource); + + // return io handle of active input or 0 if no input is active + // Only considers inputs from physical devices (e.g. main mic, headset mic) when + // ignoreVirtualInputs is true. + audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true); + + // initialize volume curves for each strategy and device category + void initializeVolumeCurves(); + + // compute the actual volume for a given stream according to the requested index and a particular + // device + virtual float computeVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device); + + // check that volume change is permitted, compute and send new volume to audio hardware + status_t checkAndSetVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false); + + // apply all stream volumes to the specified output and device + void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false); + + // Mute or unmute all streams handled by the specified strategy on the specified output + void setStrategyMute(routing_strategy strategy, + bool on, + audio_io_handle_t output, + int delayMs = 0, + audio_devices_t device = (audio_devices_t)0); + + // Mute or unmute the stream on the specified output + void setStreamMute(int stream, + bool on, + audio_io_handle_t output, + int delayMs = 0, + audio_devices_t device = (audio_devices_t)0); + + // handle special cases for sonification strategy while in call: mute streams or replace by + // a special tone in the device used for communication + void handleIncallSonification(int stream, bool starting, bool stateChange); + + // true if device is in a telephony or VoIP call + virtual bool isInCall(); + + // true if given state represents a device in a telephony or VoIP call + virtual bool isStateInCall(int state); + + // when a device is connected, checks if an open output can be routed + // to this device. If none is open, tries to open one of the available outputs. + // Returns an output suitable to this device or 0. + // when a device is disconnected, checks if an output is not used any more and + // returns its handle if any. + // transfers the audio tracks and effects from one output thread to another accordingly. + status_t checkOutputsForDevice(audio_devices_t device, + AudioSystem::device_connection_state state, + SortedVector<audio_io_handle_t>& outputs, + const String8 paramStr); + + // close an output and its companion duplicating output. + void closeOutput(audio_io_handle_t output); + + // checks and if necessary changes outputs used for all strategies. + // must be called every time a condition that affects the output choice for a given strategy + // changes: connected device, phone state, force use... + // Must be called before updateDevicesAndOutputs() + void checkOutputForStrategy(routing_strategy strategy); + + // Same as checkOutputForStrategy() but for a all strategies in order of priority + void checkOutputForAllStrategies(); + + // manages A2DP output suspend/restore according to phone state and BT SCO usage + void checkA2dpSuspend(); + + // returns the A2DP output handle if it is open or 0 otherwise + audio_io_handle_t getA2dpOutput(); + + // selects the most appropriate device on output for current state + // must be called every time a condition that affects the device choice for a given output is + // changed: connected device, phone state, force use, output start, output stop.. + // see getDeviceForStrategy() for the use of fromCache parameter + + audio_devices_t getNewDevice(audio_io_handle_t output, bool fromCache); + // updates cache of device used by all strategies (mDeviceForStrategy[]) + // must be called every time a condition that affects the device choice for a given strategy is + // changed: connected device, phone state, force use... + // cached values are used by getDeviceForStrategy() if parameter fromCache is true. + // Must be called after checkOutputForAllStrategies() + + void updateDevicesAndOutputs(); + + virtual uint32_t getMaxEffectsCpuLoad(); + virtual uint32_t getMaxEffectsMemory(); +#ifdef AUDIO_POLICY_TEST + virtual bool threadLoop(); + void exit(); + int testOutputIndex(audio_io_handle_t output); +#endif //AUDIO_POLICY_TEST + + status_t setEffectEnabled(EffectDescriptor *pDesc, bool enabled); + + // returns the category the device belongs to with regard to volume curve management + static device_category getDeviceCategory(audio_devices_t device); + + // extract one device relevant for volume control from multiple device selection + static audio_devices_t getDeviceForVolume(audio_devices_t device); + + SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device, + DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs); + bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1, + SortedVector<audio_io_handle_t>& outputs2); + + // mute/unmute strategies using an incompatible device combination + // if muting, wait for the audio in pcm buffer to be drained before proceeding + // if unmuting, unmute only after the specified delay + // Returns the number of ms waited + uint32_t checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc, + audio_devices_t prevDevice, + uint32_t delayMs); + + audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs, + AudioSystem::output_flags flags); + IOProfile *getInputProfile(audio_devices_t device, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask); + IOProfile *getProfileForDirectOutput(audio_devices_t device, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags); + + audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs); + + bool isNonOffloadableEffectEnabled(); + + // + // Audio policy configuration file parsing (audio_policy.conf) + // + static uint32_t stringToEnum(const struct StringToEnum *table, + size_t size, + const char *name); + static bool stringToBool(const char *value); + static audio_output_flags_t parseFlagNames(char *name); + static audio_devices_t parseDeviceNames(char *name); + void loadSamplingRates(char *name, IOProfile *profile); + void loadFormats(char *name, IOProfile *profile); + void loadOutChannels(char *name, IOProfile *profile); + void loadInChannels(char *name, IOProfile *profile); + status_t loadOutput(cnode *root, HwModule *module); + status_t loadInput(cnode *root, HwModule *module); + void loadHwModule(cnode *root); + void loadHwModules(cnode *root); + void loadGlobalConfig(cnode *root); + status_t loadAudioPolicyConfig(const char *path); + void defaultAudioPolicyConfig(void); + + + AudioPolicyClientInterface *mpClientInterface; // audio policy client interface + audio_io_handle_t mPrimaryOutput; // primary output handle + // list of descriptors for outputs currently opened + DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs; + // copy of mOutputs before setDeviceConnectionState() opens new outputs + // reset to mOutputs when updateDevicesAndOutputs() is called. + DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mPreviousOutputs; + DefaultKeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs; // list of input descriptors + audio_devices_t mAvailableOutputDevices; // bit field of all available output devices + audio_devices_t mAvailableInputDevices; // bit field of all available input devices + // without AUDIO_DEVICE_BIT_IN to allow direct bit + // field comparisons + int mPhoneState; // current phone state + AudioSystem::forced_config mForceUse[AudioSystem::NUM_FORCE_USE]; // current forced use configuration + + StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES]; // stream descriptors for volume control + String8 mA2dpDeviceAddress; // A2DP device MAC address + String8 mScoDeviceAddress; // SCO device MAC address + String8 mUsbCardAndDevice; // USB audio ALSA card and device numbers: + // card=<card_number>;device=<><device_number> + bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected + audio_devices_t mDeviceForStrategy[NUM_STRATEGIES]; + float mLastVoiceVolume; // last voice volume value sent to audio HAL + + // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units + static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000; + // Maximum memory allocated to audio effects in KB + static const uint32_t MAX_EFFECTS_MEMORY = 512; + uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects + uint32_t mTotalEffectsMemory; // current memory used by effects + KeyedVector<int, EffectDescriptor *> mEffects; // list of registered audio effects + bool mA2dpSuspended; // true if A2DP output is suspended + bool mHasA2dp; // true on platforms with support for bluetooth A2DP + bool mHasUsb; // true on platforms with support for USB audio + bool mHasRemoteSubmix; // true on platforms with support for remote presentation of a submix + audio_devices_t mAttachedOutputDevices; // output devices always available on the platform + audio_devices_t mDefaultOutputDevice; // output device selected by default at boot time + // (must be in mAttachedOutputDevices) + bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path + // to boost soft sounds, used to adjust volume curves accordingly + + Vector <HwModule *> mHwModules; + +#ifdef AUDIO_POLICY_TEST + Mutex mLock; + Condition mWaitWorkCV; + + int mCurOutput; + bool mDirectOutput; + audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS]; + int mTestInput; + uint32_t mTestDevice; + uint32_t mTestSamplingRate; + uint32_t mTestFormat; + uint32_t mTestChannels; + uint32_t mTestLatencyMs; +#endif //AUDIO_POLICY_TEST + +private: + static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, + int indexInUi); + // updates device caching and output for streams that can influence the + // routing of notifications + void handleNotificationRoutingForStream(AudioSystem::stream_type stream); + static bool isVirtualInputDevice(audio_devices_t device); +}; + +}; diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audiopolicy/AudioPolicyService.cpp index 41bd990..49145a5 100644 --- a/services/audioflinger/AudioPolicyService.cpp +++ b/services/audiopolicy/AudioPolicyService.cpp @@ -138,445 +138,6 @@ AudioPolicyService::~AudioPolicyService() } } -status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device, - audio_policy_dev_state_t state, - const char *device_address) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (!audio_is_output_device(device) && !audio_is_input_device(device)) { - return BAD_VALUE; - } - if (state != AUDIO_POLICY_DEVICE_STATE_AVAILABLE && - state != AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) { - return BAD_VALUE; - } - - ALOGV("setDeviceConnectionState()"); - Mutex::Autolock _l(mLock); - return mpAudioPolicy->set_device_connection_state(mpAudioPolicy, device, - state, device_address); -} - -audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState( - audio_devices_t device, - const char *device_address) -{ - if (mpAudioPolicy == NULL) { - return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; - } - return mpAudioPolicy->get_device_connection_state(mpAudioPolicy, device, - device_address); -} - -status_t AudioPolicyService::setPhoneState(audio_mode_t state) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (uint32_t(state) >= AUDIO_MODE_CNT) { - return BAD_VALUE; - } - - ALOGV("setPhoneState()"); - - // TODO: check if it is more appropriate to do it in platform specific policy manager - AudioSystem::setMode(state); - - Mutex::Autolock _l(mLock); - mpAudioPolicy->set_phone_state(mpAudioPolicy, state); - return NO_ERROR; -} - -status_t AudioPolicyService::setForceUse(audio_policy_force_use_t usage, - audio_policy_forced_cfg_t config) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) { - return BAD_VALUE; - } - if (config < 0 || config >= AUDIO_POLICY_FORCE_CFG_CNT) { - return BAD_VALUE; - } - ALOGV("setForceUse()"); - Mutex::Autolock _l(mLock); - mpAudioPolicy->set_force_use(mpAudioPolicy, usage, config); - return NO_ERROR; -} - -audio_policy_forced_cfg_t AudioPolicyService::getForceUse(audio_policy_force_use_t usage) -{ - if (mpAudioPolicy == NULL) { - return AUDIO_POLICY_FORCE_NONE; - } - if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) { - return AUDIO_POLICY_FORCE_NONE; - } - return mpAudioPolicy->get_force_use(mpAudioPolicy, usage); -} - -audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo) -{ - if (mpAudioPolicy == NULL) { - return 0; - } - ALOGV("getOutput()"); - Mutex::Autolock _l(mLock); - return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, - format, channelMask, flags, offloadInfo); -} - -status_t AudioPolicyService::startOutput(audio_io_handle_t output, - audio_stream_type_t stream, - int session) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - ALOGV("startOutput()"); - Mutex::Autolock _l(mLock); - return mpAudioPolicy->start_output(mpAudioPolicy, output, stream, session); -} - -status_t AudioPolicyService::stopOutput(audio_io_handle_t output, - audio_stream_type_t stream, - int session) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - ALOGV("stopOutput()"); - mOutputCommandThread->stopOutputCommand(output, stream, session); - return NO_ERROR; -} - -status_t AudioPolicyService::doStopOutput(audio_io_handle_t output, - audio_stream_type_t stream, - int session) -{ - ALOGV("doStopOutput from tid %d", gettid()); - Mutex::Autolock _l(mLock); - return mpAudioPolicy->stop_output(mpAudioPolicy, output, stream, session); -} - -void AudioPolicyService::releaseOutput(audio_io_handle_t output) -{ - if (mpAudioPolicy == NULL) { - return; - } - ALOGV("releaseOutput()"); - mOutputCommandThread->releaseOutputCommand(output); -} - -void AudioPolicyService::doReleaseOutput(audio_io_handle_t output) -{ - ALOGV("doReleaseOutput from tid %d", gettid()); - Mutex::Autolock _l(mLock); - mpAudioPolicy->release_output(mpAudioPolicy, output); -} - -audio_io_handle_t AudioPolicyService::getInput(audio_source_t inputSource, - uint32_t samplingRate, - audio_format_t format, - audio_channel_mask_t channelMask, - int audioSession) -{ - if (mpAudioPolicy == NULL) { - return 0; - } - // already checked by client, but double-check in case the client wrapper is bypassed - if (inputSource >= AUDIO_SOURCE_CNT && inputSource != AUDIO_SOURCE_HOTWORD) { - return 0; - } - - if ((inputSource == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) { - return 0; - } - - Mutex::Autolock _l(mLock); - // the audio_in_acoustics_t parameter is ignored by get_input() - audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate, - format, channelMask, (audio_in_acoustics_t) 0); - - if (input == 0) { - return input; - } - // create audio pre processors according to input source - audio_source_t aliasSource = (inputSource == AUDIO_SOURCE_HOTWORD) ? - AUDIO_SOURCE_VOICE_RECOGNITION : inputSource; - - ssize_t index = mInputSources.indexOfKey(aliasSource); - if (index < 0) { - return input; - } - ssize_t idx = mInputs.indexOfKey(input); - InputDesc *inputDesc; - if (idx < 0) { - inputDesc = new InputDesc(audioSession); - mInputs.add(input, inputDesc); - } else { - inputDesc = mInputs.valueAt(idx); - } - - Vector <EffectDesc *> effects = mInputSources.valueAt(index)->mEffects; - for (size_t i = 0; i < effects.size(); i++) { - EffectDesc *effect = effects[i]; - sp<AudioEffect> fx = new AudioEffect(NULL, &effect->mUuid, -1, 0, 0, audioSession, input); - status_t status = fx->initCheck(); - if (status != NO_ERROR && status != ALREADY_EXISTS) { - ALOGW("Failed to create Fx %s on input %d", effect->mName, input); - // fx goes out of scope and strong ref on AudioEffect is released - continue; - } - for (size_t j = 0; j < effect->mParams.size(); j++) { - fx->setParameter(effect->mParams[j]); - } - inputDesc->mEffects.add(fx); - } - setPreProcessorEnabled(inputDesc, true); - return input; -} - -status_t AudioPolicyService::startInput(audio_io_handle_t input) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - Mutex::Autolock _l(mLock); - - return mpAudioPolicy->start_input(mpAudioPolicy, input); -} - -status_t AudioPolicyService::stopInput(audio_io_handle_t input) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - Mutex::Autolock _l(mLock); - - return mpAudioPolicy->stop_input(mpAudioPolicy, input); -} - -void AudioPolicyService::releaseInput(audio_io_handle_t input) -{ - if (mpAudioPolicy == NULL) { - return; - } - Mutex::Autolock _l(mLock); - mpAudioPolicy->release_input(mpAudioPolicy, input); - - ssize_t index = mInputs.indexOfKey(input); - if (index < 0) { - return; - } - InputDesc *inputDesc = mInputs.valueAt(index); - setPreProcessorEnabled(inputDesc, false); - delete inputDesc; - mInputs.removeItemsAt(index); -} - -status_t AudioPolicyService::initStreamVolume(audio_stream_type_t stream, - int indexMin, - int indexMax) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (uint32_t(stream) >= AUDIO_STREAM_CNT) { - return BAD_VALUE; - } - Mutex::Autolock _l(mLock); - mpAudioPolicy->init_stream_volume(mpAudioPolicy, stream, indexMin, indexMax); - return NO_ERROR; -} - -status_t AudioPolicyService::setStreamVolumeIndex(audio_stream_type_t stream, - int index, - audio_devices_t device) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - if (!settingsAllowed()) { - return PERMISSION_DENIED; - } - if (uint32_t(stream) >= AUDIO_STREAM_CNT) { - return BAD_VALUE; - } - Mutex::Autolock _l(mLock); - if (mpAudioPolicy->set_stream_volume_index_for_device) { - return mpAudioPolicy->set_stream_volume_index_for_device(mpAudioPolicy, - stream, - index, - device); - } else { - return mpAudioPolicy->set_stream_volume_index(mpAudioPolicy, stream, index); - } -} - -status_t AudioPolicyService::getStreamVolumeIndex(audio_stream_type_t stream, - int *index, - audio_devices_t device) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - if (uint32_t(stream) >= AUDIO_STREAM_CNT) { - return BAD_VALUE; - } - Mutex::Autolock _l(mLock); - if (mpAudioPolicy->get_stream_volume_index_for_device) { - return mpAudioPolicy->get_stream_volume_index_for_device(mpAudioPolicy, - stream, - index, - device); - } else { - return mpAudioPolicy->get_stream_volume_index(mpAudioPolicy, stream, index); - } -} - -uint32_t AudioPolicyService::getStrategyForStream(audio_stream_type_t stream) -{ - if (mpAudioPolicy == NULL) { - return 0; - } - return mpAudioPolicy->get_strategy_for_stream(mpAudioPolicy, stream); -} - -//audio policy: use audio_device_t appropriately - -audio_devices_t AudioPolicyService::getDevicesForStream(audio_stream_type_t stream) -{ - if (mpAudioPolicy == NULL) { - return (audio_devices_t)0; - } - return mpAudioPolicy->get_devices_for_stream(mpAudioPolicy, stream); -} - -audio_io_handle_t AudioPolicyService::getOutputForEffect(const effect_descriptor_t *desc) -{ - // FIXME change return type to status_t, and return NO_INIT here - if (mpAudioPolicy == NULL) { - return 0; - } - Mutex::Autolock _l(mLock); - return mpAudioPolicy->get_output_for_effect(mpAudioPolicy, desc); -} - -status_t AudioPolicyService::registerEffect(const effect_descriptor_t *desc, - audio_io_handle_t io, - uint32_t strategy, - int session, - int id) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - return mpAudioPolicy->register_effect(mpAudioPolicy, desc, io, strategy, session, id); -} - -status_t AudioPolicyService::unregisterEffect(int id) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - return mpAudioPolicy->unregister_effect(mpAudioPolicy, id); -} - -status_t AudioPolicyService::setEffectEnabled(int id, bool enabled) -{ - if (mpAudioPolicy == NULL) { - return NO_INIT; - } - return mpAudioPolicy->set_effect_enabled(mpAudioPolicy, id, enabled); -} - -bool AudioPolicyService::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const -{ - if (mpAudioPolicy == NULL) { - return 0; - } - Mutex::Autolock _l(mLock); - return mpAudioPolicy->is_stream_active(mpAudioPolicy, stream, inPastMs); -} - -bool AudioPolicyService::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const -{ - if (mpAudioPolicy == NULL) { - return 0; - } - Mutex::Autolock _l(mLock); - return mpAudioPolicy->is_stream_active_remotely(mpAudioPolicy, stream, inPastMs); -} - -bool AudioPolicyService::isSourceActive(audio_source_t source) const -{ - if (mpAudioPolicy == NULL) { - return false; - } - if (mpAudioPolicy->is_source_active == 0) { - return false; - } - Mutex::Autolock _l(mLock); - return mpAudioPolicy->is_source_active(mpAudioPolicy, source); -} - -status_t AudioPolicyService::queryDefaultPreProcessing(int audioSession, - effect_descriptor_t *descriptors, - uint32_t *count) -{ - - if (mpAudioPolicy == NULL) { - *count = 0; - return NO_INIT; - } - Mutex::Autolock _l(mLock); - status_t status = NO_ERROR; - - size_t index; - for (index = 0; index < mInputs.size(); index++) { - if (mInputs.valueAt(index)->mSessionId == audioSession) { - break; - } - } - if (index == mInputs.size()) { - *count = 0; - return BAD_VALUE; - } - Vector< sp<AudioEffect> > effects = mInputs.valueAt(index)->mEffects; - - for (size_t i = 0; i < effects.size(); i++) { - effect_descriptor_t desc = effects[i]->descriptor(); - if (i < *count) { - descriptors[i] = desc; - } - } - if (effects.size() > *count) { - status = NO_MEMORY; - } - *count = effects.size(); - return status; -} void AudioPolicyService::binderDied(const wp<IBinder>& who) { ALOGW("binderDied() %p, calling pid %d", who.unsafe_get(), @@ -1144,21 +705,6 @@ int AudioPolicyService::setVoiceVolume(float volume, int delayMs) return (int)mAudioCommandThread->voiceVolumeCommand(volume, delayMs); } -bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info) -{ - if (mpAudioPolicy == NULL) { - ALOGV("mpAudioPolicy == NULL"); - return false; - } - - if (mpAudioPolicy->is_offload_supported == NULL) { - ALOGV("HAL does not implement is_offload_supported"); - return false; - } - - return mpAudioPolicy->is_offload_supported(mpAudioPolicy, &info); -} - // ---------------------------------------------------------------------------- // Audio pre-processing configuration // ---------------------------------------------------------------------------- @@ -1457,42 +1003,18 @@ status_t AudioPolicyService::loadPreProcessorConfig(const char *path) return NO_ERROR; } -/* implementation of the interface to the policy manager */ extern "C" { - - -static audio_module_handle_t aps_load_hw_module(void *service __unused, - const char *name) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return 0; - } - - return af->loadHwModule(name); -} - -// deprecated: replaced by aps_open_output_on_module() -static audio_io_handle_t aps_open_output(void *service __unused, +audio_module_handle_t aps_load_hw_module(void *service __unused, + const char *name); +audio_io_handle_t aps_open_output(void *service __unused, audio_devices_t *pDevices, uint32_t *pSamplingRate, audio_format_t *pFormat, audio_channel_mask_t *pChannelMask, uint32_t *pLatencyMs, - audio_output_flags_t flags) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return 0; - } + audio_output_flags_t flags); - return af->openOutput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask, - pLatencyMs, flags); -} - -static audio_io_handle_t aps_open_output_on_module(void *service __unused, +audio_io_handle_t aps_open_output_on_module(void *service __unused, audio_module_handle_t module, audio_devices_t *pDevices, uint32_t *pSamplingRate, @@ -1500,174 +1022,42 @@ static audio_io_handle_t aps_open_output_on_module(void *service __unused, audio_channel_mask_t *pChannelMask, uint32_t *pLatencyMs, audio_output_flags_t flags, - const audio_offload_info_t *offloadInfo) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return 0; - } - return af->openOutput(module, pDevices, pSamplingRate, pFormat, pChannelMask, - pLatencyMs, flags, offloadInfo); -} - -static audio_io_handle_t aps_open_dup_output(void *service __unused, + const audio_offload_info_t *offloadInfo); +audio_io_handle_t aps_open_dup_output(void *service __unused, audio_io_handle_t output1, - audio_io_handle_t output2) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return 0; - } - return af->openDuplicateOutput(output1, output2); -} - -static int aps_close_output(void *service __unused, audio_io_handle_t output) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - return PERMISSION_DENIED; - } - - return af->closeOutput(output); -} - -static int aps_suspend_output(void *service __unused, audio_io_handle_t output) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return PERMISSION_DENIED; - } - - return af->suspendOutput(output); -} - -static int aps_restore_output(void *service __unused, audio_io_handle_t output) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return PERMISSION_DENIED; - } - - return af->restoreOutput(output); -} - -// deprecated: replaced by aps_open_input_on_module(), and acoustics parameter is ignored -static audio_io_handle_t aps_open_input(void *service __unused, + audio_io_handle_t output2); +int aps_close_output(void *service __unused, audio_io_handle_t output); +int aps_suspend_output(void *service __unused, audio_io_handle_t output); +int aps_restore_output(void *service __unused, audio_io_handle_t output); +audio_io_handle_t aps_open_input(void *service __unused, audio_devices_t *pDevices, uint32_t *pSamplingRate, audio_format_t *pFormat, audio_channel_mask_t *pChannelMask, - audio_in_acoustics_t acoustics __unused) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return 0; - } - - return af->openInput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask); -} - -static audio_io_handle_t aps_open_input_on_module(void *service __unused, + audio_in_acoustics_t acoustics __unused); +audio_io_handle_t aps_open_input_on_module(void *service __unused, audio_module_handle_t module, audio_devices_t *pDevices, uint32_t *pSamplingRate, audio_format_t *pFormat, - audio_channel_mask_t *pChannelMask) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - ALOGW("%s: could not get AudioFlinger", __func__); - return 0; - } - - return af->openInput(module, pDevices, pSamplingRate, pFormat, pChannelMask); -} - -static int aps_close_input(void *service __unused, audio_io_handle_t input) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - return PERMISSION_DENIED; - } - - return af->closeInput(input); -} - -static int aps_invalidate_stream(void *service __unused, audio_stream_type_t stream) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - return PERMISSION_DENIED; - } - - return af->invalidateStream(stream); -} - -static int aps_move_effects(void *service __unused, int session, + audio_channel_mask_t *pChannelMask); +int aps_close_input(void *service __unused, audio_io_handle_t input); +int aps_invalidate_stream(void *service __unused, audio_stream_type_t stream); +int aps_move_effects(void *service __unused, int session, audio_io_handle_t src_output, - audio_io_handle_t dst_output) -{ - sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) { - return PERMISSION_DENIED; - } - - return af->moveEffects(session, src_output, dst_output); -} - -static char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle, - const char *keys) -{ - String8 result = AudioSystem::getParameters(io_handle, String8(keys)); - return strdup(result.string()); -} - -static void aps_set_parameters(void *service, audio_io_handle_t io_handle, - const char *kv_pairs, int delay_ms) -{ - AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; - - audioPolicyService->setParameters(io_handle, kv_pairs, delay_ms); -} - -static int aps_set_stream_volume(void *service, audio_stream_type_t stream, + audio_io_handle_t dst_output); +char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle, + const char *keys); +void aps_set_parameters(void *service, audio_io_handle_t io_handle, + const char *kv_pairs, int delay_ms); +int aps_set_stream_volume(void *service, audio_stream_type_t stream, float volume, audio_io_handle_t output, - int delay_ms) -{ - AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; - - return audioPolicyService->setStreamVolume(stream, volume, output, - delay_ms); -} - -static int aps_start_tone(void *service, audio_policy_tone_t tone, - audio_stream_type_t stream) -{ - AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; - - return audioPolicyService->startTone(tone, stream); -} - -static int aps_stop_tone(void *service) -{ - AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; - - return audioPolicyService->stopTone(); -} - -static int aps_set_voice_volume(void *service, float volume, int delay_ms) -{ - AudioPolicyService *audioPolicyService = (AudioPolicyService *)service; - - return audioPolicyService->setVoiceVolume(volume, delay_ms); -} - -}; // extern "C" + int delay_ms); +int aps_start_tone(void *service, audio_policy_tone_t tone, + audio_stream_type_t stream); +int aps_stop_tone(void *service); +int aps_set_voice_volume(void *service, float volume, int delay_ms); +}; namespace { struct audio_policy_service_ops aps_ops = { diff --git a/services/audioflinger/AudioPolicyService.h b/services/audiopolicy/AudioPolicyService.h index ae053a9..ae053a9 100644 --- a/services/audioflinger/AudioPolicyService.h +++ b/services/audiopolicy/AudioPolicyService.h diff --git a/services/camera/libcameraservice/Android.mk b/services/camera/libcameraservice/Android.mk index 51ba698..4e2272d 100644 --- a/services/camera/libcameraservice/Android.mk +++ b/services/camera/libcameraservice/Android.mk @@ -1,3 +1,17 @@ +# Copyright 2010 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. + LOCAL_PATH:= $(call my-dir) # @@ -53,11 +67,13 @@ LOCAL_SHARED_LIBRARIES:= \ LOCAL_C_INCLUDES += \ system/media/camera/include \ + system/media/private/camera/include \ external/jpeg LOCAL_CFLAGS += -Wall -Wextra LOCAL_MODULE:= libcameraservice +LOCAL_32_BIT_ONLY := true include $(BUILD_SHARED_LIBRARY) diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp index 79fbf76..b83c315 100644 --- a/services/camera/libcameraservice/CameraService.cpp +++ b/services/camera/libcameraservice/CameraService.cpp @@ -1,24 +1,24 @@ /* -** -** 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. -*/ + * 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. + */ #define LOG_TAG "CameraService" //#define LOG_NDEBUG 0 #include <stdio.h> +#include <string.h> #include <sys/types.h> #include <pthread.h> @@ -37,6 +37,8 @@ #include <utils/Errors.h> #include <utils/Log.h> #include <utils/String16.h> +#include <utils/Trace.h> +#include <system/camera_vendor_tags.h> #include "CameraService.h" #include "api1/CameraClient.h" @@ -131,6 +133,12 @@ void CameraService::onFirstRef() mModule->set_callbacks(this); } + VendorTagDescriptor::clearGlobalVendorTagDescriptor(); + + if (mModule->common.module_api_version >= CAMERA_MODULE_API_VERSION_2_2) { + setUpVendorTags(); + } + CameraDeviceFactory::registerService(this); } } @@ -142,6 +150,7 @@ CameraService::~CameraService() { } } + VendorTagDescriptor::clearGlobalVendorTagDescriptor(); gCameraService = NULL; } @@ -270,6 +279,22 @@ status_t CameraService::getCameraCharacteristics(int cameraId, return ret; } +status_t CameraService::getCameraVendorTagDescriptor(/*out*/sp<VendorTagDescriptor>& desc) { + if (!mModule) { + ALOGE("%s: camera hardware module doesn't exist", __FUNCTION__); + return -ENODEV; + } + + if (mModule->common.module_api_version < CAMERA_MODULE_API_VERSION_2_2) { + // TODO: Remove this check once HAL1 shim is in place. + ALOGW("%s: Only HAL module version V2.2 or higher supports vendor tags", __FUNCTION__); + return -EOPNOTSUPP; + } + + desc = VendorTagDescriptor::getGlobalVendorTagDescriptor(); + return OK; +} + int CameraService::getDeviceVersion(int cameraId, int* facing) { struct camera_info info; if (mModule->get_camera_info(cameraId, &info) != OK) { @@ -307,6 +332,44 @@ bool CameraService::isValidCameraId(int cameraId) { return false; } +bool CameraService::setUpVendorTags() { + vendor_tag_ops_t vOps = vendor_tag_ops_t(); + + // Check if vendor operations have been implemented + if (mModule->get_vendor_tag_ops == NULL) { + ALOGI("%s: No vendor tags defined for this device.", __FUNCTION__); + return false; + } + + ATRACE_BEGIN("camera3->get_metadata_vendor_tag_ops"); + mModule->get_vendor_tag_ops(&vOps); + ATRACE_END(); + + // Ensure all vendor operations are present + if (vOps.get_tag_count == NULL || vOps.get_all_tags == NULL || + vOps.get_section_name == NULL || vOps.get_tag_name == NULL || + vOps.get_tag_type == NULL) { + ALOGE("%s: Vendor tag operations not fully defined. Ignoring definitions." + , __FUNCTION__); + return false; + } + + // Read all vendor tag definitions into a descriptor + sp<VendorTagDescriptor> desc; + status_t res; + if ((res = VendorTagDescriptor::createDescriptorFromOps(&vOps, /*out*/desc)) + != OK) { + ALOGE("%s: Could not generate descriptor from vendor tag operations," + "received error %s (%d). Camera clients will not be able to use" + "vendor tags", __FUNCTION__, strerror(res), res); + return false; + } + + // Set the global descriptor to use with camera metadata + VendorTagDescriptor::setAsGlobalVendorTagDescriptor(desc); + return true; +} + status_t CameraService::validateConnect(int cameraId, /*inout*/ int& clientUid) const { @@ -656,6 +719,11 @@ status_t CameraService::addListener( const sp<ICameraServiceListener>& listener) { ALOGV("%s: Add listener %p", __FUNCTION__, listener.get()); + if (listener == 0) { + ALOGE("%s: Listener must not be null", __FUNCTION__); + return BAD_VALUE; + } + Mutex::Autolock lock(mServiceLock); Vector<sp<ICameraServiceListener> >::iterator it, end; @@ -684,6 +752,11 @@ status_t CameraService::removeListener( const sp<ICameraServiceListener>& listener) { ALOGV("%s: Remove listener %p", __FUNCTION__, listener.get()); + if (listener == 0) { + ALOGE("%s: Listener must not be null", __FUNCTION__); + return BAD_VALUE; + } + Mutex::Autolock lock(mServiceLock); Vector<sp<ICameraServiceListener> >::iterator it; diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h index ad6a582..8853e48 100644 --- a/services/camera/libcameraservice/CameraService.h +++ b/services/camera/libcameraservice/CameraService.h @@ -1,19 +1,18 @@ /* -** -** 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. -*/ + * 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 ANDROID_SERVERS_CAMERA_CAMERASERVICE_H #define ANDROID_SERVERS_CAMERA_CAMERASERVICE_H @@ -31,6 +30,7 @@ #include <camera/IProCameraCallbacks.h> #include <camera/camera2/ICameraDeviceUser.h> #include <camera/camera2/ICameraDeviceCallbacks.h> +#include <camera/VendorTagDescriptor.h> #include <camera/ICameraServiceListener.h> @@ -73,6 +73,7 @@ public: struct CameraInfo* cameraInfo); virtual status_t getCameraCharacteristics(int cameraId, CameraMetadata* cameraInfo); + virtual status_t getCameraVendorTagDescriptor(/*out*/ sp<VendorTagDescriptor>& desc); virtual status_t connect(const sp<ICameraClient>& cameraClient, int cameraId, const String16& clientPackageName, int clientUid, @@ -387,6 +388,8 @@ private: // Helpers bool isValidCameraId(int cameraId); + + bool setUpVendorTags(); }; } // namespace android diff --git a/services/camera/libcameraservice/device2/Camera2Device.cpp b/services/camera/libcameraservice/device2/Camera2Device.cpp index dc97c47..f60ca98 100644 --- a/services/camera/libcameraservice/device2/Camera2Device.cpp +++ b/services/camera/libcameraservice/device2/Camera2Device.cpp @@ -112,20 +112,6 @@ status_t Camera2Device::initialize(camera_module_t *module) return res; } - res = device->ops->get_metadata_vendor_tag_ops(device, &mVendorTagOps); - if (res != OK ) { - ALOGE("%s: Camera %d: Unable to retrieve tag ops from device: %s (%d)", - __FUNCTION__, mId, strerror(-res), res); - device->common.close(&device->common); - return res; - } - res = set_camera_metadata_vendor_tag_ops(mVendorTagOps); - if (res != OK) { - ALOGE("%s: Camera %d: Unable to set tag ops: %s (%d)", - __FUNCTION__, mId, strerror(-res), res); - device->common.close(&device->common); - return res; - } res = device->ops->set_notify_callback(device, notificationCallback, NULL); if (res != OK) { diff --git a/services/camera/libcameraservice/device2/Camera2Device.h b/services/camera/libcameraservice/device2/Camera2Device.h index 1f53c56..5b91f88 100644 --- a/services/camera/libcameraservice/device2/Camera2Device.h +++ b/services/camera/libcameraservice/device2/Camera2Device.h @@ -78,7 +78,6 @@ class Camera2Device: public CameraDeviceBase { camera2_device_t *mHal2Device; CameraMetadata mDeviceInfo; - vendor_tag_query_ops_t *mVendorTagOps; /** * Queue class for both sending requests to a camera2 device, and for diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp index da3e121..08e03ce 100644 --- a/services/camera/libcameraservice/device3/Camera3Device.cpp +++ b/services/camera/libcameraservice/device3/Camera3Device.cpp @@ -146,24 +146,6 @@ status_t Camera3Device::initialize(camera_module_t *module) return BAD_VALUE; } - /** Get vendor metadata tags */ - - mVendorTagOps.get_camera_vendor_section_name = NULL; - - ATRACE_BEGIN("camera3->get_metadata_vendor_tag_ops"); - device->ops->get_metadata_vendor_tag_ops(device, &mVendorTagOps); - ATRACE_END(); - - if (mVendorTagOps.get_camera_vendor_section_name != NULL) { - res = set_camera_metadata_vendor_tag_ops(&mVendorTagOps); - if (res != OK) { - SET_ERR_L("Unable to set tag ops: %s (%d)", - strerror(-res), res); - device->common.close(&device->common); - return res; - } - } - /** Start up status tracker thread */ mStatusTracker = new StatusTracker(this); res = mStatusTracker->run(String8::format("C3Dev-%d-Status", mId).string()); diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h index 468f641..9007a9b 100644 --- a/services/camera/libcameraservice/device3/Camera3Device.h +++ b/services/camera/libcameraservice/device3/Camera3Device.h @@ -157,7 +157,6 @@ class Camera3Device : camera3_device_t *mHal3Device; CameraMetadata mDeviceInfo; - vendor_tag_query_ops_t mVendorTagOps; enum Status { STATUS_ERROR, diff --git a/services/medialog/Android.mk b/services/medialog/Android.mk index 08006c8..95f2fef 100644 --- a/services/medialog/Android.mk +++ b/services/medialog/Android.mk @@ -8,4 +8,6 @@ LOCAL_SHARED_LIBRARIES := libmedia libbinder libutils liblog libnbaio LOCAL_MODULE:= libmedialogservice +LOCAL_32_BIT_ONLY := true + include $(BUILD_SHARED_LIBRARY) |