Merge changes from topic "aidlize-audioflinger2"
* changes:
Break-up libaudioclient_aidl_conversion
Add missing header dependencies
Prepare openOutput() signature to AIDL conversion
Extract VALUE_OR_RETURN_STATUS to AidlConversion
Disambiguate types
Add some missing @hide annotations
Prepare openInput() signature to AIDL conversion
AIDLize createEffect params
Include the returned IAudio{Track,Record} in response parcelable
Make IAudioRecord pure (stable) AIDL
Convert IAudioTrack to AIDL
diff --git a/Android.bp b/Android.bp
index 87a8f41..327593f 100644
--- a/Android.bp
+++ b/Android.bp
@@ -8,6 +8,7 @@
srcs: [
"aidl/android/media/InterpolatorConfig.aidl",
"aidl/android/media/InterpolatorType.aidl",
+ "aidl/android/media/MicrophoneInfoData.aidl",
"aidl/android/media/VolumeShaperConfiguration.aidl",
"aidl/android/media/VolumeShaperConfigurationOptionFlag.aidl",
"aidl/android/media/VolumeShaperConfigurationType.aidl",
@@ -20,8 +21,37 @@
min_sdk_version: "29",
apex_available: [
"//apex_available:platform",
+ "com.android.bluetooth.updatable",
"com.android.media",
+ "com.android.media.swcodec",
],
},
},
}
+
+cc_library_headers {
+ name: "av-headers",
+ export_include_dirs: ["include"],
+ static_libs: [
+ "av-types-aidl-unstable-cpp",
+ ],
+ export_static_lib_headers: [
+ "av-types-aidl-unstable-cpp",
+ ],
+ header_libs: [
+ "libaudioclient_aidl_conversion_util",
+ ],
+ export_header_lib_headers: [
+ "libaudioclient_aidl_conversion_util",
+ ],
+ host_supported: true,
+ vendor_available: true,
+ double_loadable: true,
+ min_sdk_version: "29",
+ apex_available: [
+ "//apex_available:platform",
+ "com.android.bluetooth.updatable",
+ "com.android.media",
+ "com.android.media.swcodec",
+ ],
+}
diff --git a/aidl/android/media/MicrophoneInfoData.aidl b/aidl/android/media/MicrophoneInfoData.aidl
new file mode 100644
index 0000000..747bfa5
--- /dev/null
+++ b/aidl/android/media/MicrophoneInfoData.aidl
@@ -0,0 +1,39 @@
+/*
+ * Copyright 2020 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.
+ */
+
+package android.media;
+
+/**
+ * {@hide}
+ */
+parcelable MicrophoneInfoData {
+ @utf8InCpp String deviceId;
+ int portId;
+ int type;
+ @utf8InCpp String address;
+ int deviceLocation;
+ int deviceGroup;
+ int indexInTheGroup;
+ float[] geometricLocation;
+ float[] orientation;
+ float[] frequencies;
+ float[] frequencyResponses;
+ int[] channelMapping;
+ float sensitivity;
+ float maxSpl;
+ float minSpl;
+ int directionality;
+}
diff --git a/include/media/MicrophoneInfo.h b/include/media/MicrophoneInfo.h
index 0a24b02..e89401a 100644
--- a/include/media/MicrophoneInfo.h
+++ b/include/media/MicrophoneInfo.h
@@ -17,33 +17,24 @@
#ifndef ANDROID_MICROPHONE_INFO_H
#define ANDROID_MICROPHONE_INFO_H
+#include <android/media/MicrophoneInfoData.h>
#include <binder/Parcel.h>
#include <binder/Parcelable.h>
+#include <media/AidlConversionUtil.h>
#include <system/audio.h>
-#include <utils/String16.h>
-#include <utils/Vector.h>
namespace android {
namespace media {
-#define RETURN_IF_FAILED(calledOnce) \
- { \
- status_t returnStatus = calledOnce; \
- if (returnStatus) { \
- ALOGE("Failed at %s:%d (%s)", __FILE__, __LINE__, __func__); \
- return returnStatus; \
- } \
- }
-
class MicrophoneInfo : public Parcelable {
public:
MicrophoneInfo() = default;
MicrophoneInfo(const MicrophoneInfo& microphoneInfo) = default;
MicrophoneInfo(audio_microphone_characteristic_t& characteristic) {
- mDeviceId = String16(&characteristic.device_id[0]);
+ mDeviceId = std::string(&characteristic.device_id[0]);
mPortId = characteristic.id;
mType = characteristic.device;
- mAddress = String16(&characteristic.address[0]);
+ mAddress = std::string(&characteristic.address[0]);
mDeviceLocation = characteristic.location;
mDeviceGroup = characteristic.group;
mIndexInTheGroup = characteristic.index_in_the_group;
@@ -53,8 +44,8 @@
mOrientation.push_back(characteristic.orientation.x);
mOrientation.push_back(characteristic.orientation.y);
mOrientation.push_back(characteristic.orientation.z);
- Vector<float> frequencies;
- Vector<float> responses;
+ std::vector<float> frequencies;
+ std::vector<float> responses;
for (size_t i = 0; i < characteristic.num_frequency_responses; i++) {
frequencies.push_back(characteristic.frequency_responses[0][i]);
responses.push_back(characteristic.frequency_responses[1][i]);
@@ -73,76 +64,73 @@
virtual ~MicrophoneInfo() = default;
virtual status_t writeToParcel(Parcel* parcel) const {
- RETURN_IF_FAILED(parcel->writeString16(mDeviceId));
- RETURN_IF_FAILED(parcel->writeInt32(mPortId));
- RETURN_IF_FAILED(parcel->writeUint32(mType));
- RETURN_IF_FAILED(parcel->writeString16(mAddress));
- RETURN_IF_FAILED(parcel->writeInt32(mDeviceLocation));
- RETURN_IF_FAILED(parcel->writeInt32(mDeviceGroup));
- RETURN_IF_FAILED(parcel->writeInt32(mIndexInTheGroup));
- RETURN_IF_FAILED(writeFloatVector(parcel, mGeometricLocation));
- RETURN_IF_FAILED(writeFloatVector(parcel, mOrientation));
+ MicrophoneInfoData parcelable;
+ return writeToParcelable(&parcelable)
+ ?: parcelable.writeToParcel(parcel);
+ }
+
+ virtual status_t writeToParcelable(MicrophoneInfoData* parcelable) const {
+ parcelable->deviceId = mDeviceId;
+ parcelable->portId = mPortId;
+ parcelable->type = VALUE_OR_RETURN_STATUS(convertReinterpret<int32_t>(mType));
+ parcelable->address = mAddress;
+ parcelable->deviceGroup = mDeviceGroup;
+ parcelable->indexInTheGroup = mIndexInTheGroup;
+ parcelable->geometricLocation = mGeometricLocation;
+ parcelable->orientation = mOrientation;
if (mFrequencyResponses.size() != 2) {
return BAD_VALUE;
}
- for (size_t i = 0; i < mFrequencyResponses.size(); i++) {
- RETURN_IF_FAILED(parcel->writeInt32(mFrequencyResponses[i].size()));
- RETURN_IF_FAILED(writeFloatVector(parcel, mFrequencyResponses[i]));
- }
- std::vector<int> channelMapping;
- for (size_t i = 0; i < mChannelMapping.size(); ++i) {
- channelMapping.push_back(mChannelMapping[i]);
- }
- RETURN_IF_FAILED(parcel->writeInt32Vector(channelMapping));
- RETURN_IF_FAILED(parcel->writeFloat(mSensitivity));
- RETURN_IF_FAILED(parcel->writeFloat(mMaxSpl));
- RETURN_IF_FAILED(parcel->writeFloat(mMinSpl));
- RETURN_IF_FAILED(parcel->writeInt32(mDirectionality));
+ parcelable->frequencies = mFrequencyResponses[0];
+ parcelable->frequencyResponses = mFrequencyResponses[1];
+ parcelable->channelMapping = mChannelMapping;
+ parcelable->sensitivity = mSensitivity;
+ parcelable->maxSpl = mMaxSpl;
+ parcelable->minSpl = mMinSpl;
+ parcelable->directionality = mDirectionality;
return OK;
}
virtual status_t readFromParcel(const Parcel* parcel) {
- RETURN_IF_FAILED(parcel->readString16(&mDeviceId));
- RETURN_IF_FAILED(parcel->readInt32(&mPortId));
- RETURN_IF_FAILED(parcel->readUint32(&mType));
- RETURN_IF_FAILED(parcel->readString16(&mAddress));
- RETURN_IF_FAILED(parcel->readInt32(&mDeviceLocation));
- RETURN_IF_FAILED(parcel->readInt32(&mDeviceGroup));
- RETURN_IF_FAILED(parcel->readInt32(&mIndexInTheGroup));
- RETURN_IF_FAILED(readFloatVector(parcel, &mGeometricLocation, 3));
- RETURN_IF_FAILED(readFloatVector(parcel, &mOrientation, 3));
- int32_t frequenciesNum;
- RETURN_IF_FAILED(parcel->readInt32(&frequenciesNum));
- Vector<float> frequencies;
- RETURN_IF_FAILED(readFloatVector(parcel, &frequencies, frequenciesNum));
- int32_t responsesNum;
- RETURN_IF_FAILED(parcel->readInt32(&responsesNum));
- Vector<float> responses;
- RETURN_IF_FAILED(readFloatVector(parcel, &responses, responsesNum));
- if (frequencies.size() != responses.size()) {
+ MicrophoneInfoData data;
+ return data.readFromParcel(parcel)
+ ?: readFromParcelable(data);
+ }
+
+ virtual status_t readFromParcelable(const MicrophoneInfoData& parcelable) {
+ mDeviceId = parcelable.deviceId;
+ mPortId = parcelable.portId;
+ mType = VALUE_OR_RETURN_STATUS(convertReinterpret<uint32_t>(parcelable.type));
+ mAddress = parcelable.address;
+ mDeviceLocation = parcelable.deviceLocation;
+ mDeviceGroup = parcelable.deviceGroup;
+ mIndexInTheGroup = parcelable.indexInTheGroup;
+ if (parcelable.geometricLocation.size() != 3) {
return BAD_VALUE;
}
- mFrequencyResponses.push_back(frequencies);
- mFrequencyResponses.push_back(responses);
- std::vector<int> channelMapping;
- status_t result = parcel->readInt32Vector(&channelMapping);
- if (result != OK) {
- return result;
- }
- if (channelMapping.size() != AUDIO_CHANNEL_COUNT_MAX) {
+ mGeometricLocation = parcelable.geometricLocation;
+ if (parcelable.orientation.size() != 3) {
return BAD_VALUE;
}
- for (size_t i = 0; i < channelMapping.size(); i++) {
- mChannelMapping.push_back(channelMapping[i]);
+ mOrientation = parcelable.orientation;
+ if (parcelable.frequencies.size() != parcelable.frequencyResponses.size()) {
+ return BAD_VALUE;
}
- RETURN_IF_FAILED(parcel->readFloat(&mSensitivity));
- RETURN_IF_FAILED(parcel->readFloat(&mMaxSpl));
- RETURN_IF_FAILED(parcel->readFloat(&mMinSpl));
- RETURN_IF_FAILED(parcel->readInt32(&mDirectionality));
+
+ mFrequencyResponses.push_back(parcelable.frequencies);
+ mFrequencyResponses.push_back(parcelable.frequencyResponses);
+ if (parcelable.channelMapping.size() != AUDIO_CHANNEL_COUNT_MAX) {
+ return BAD_VALUE;
+ }
+ mChannelMapping = parcelable.channelMapping;
+ mSensitivity = parcelable.sensitivity;
+ mMaxSpl = parcelable.maxSpl;
+ mMinSpl = parcelable.minSpl;
+ mDirectionality = parcelable.directionality;
return OK;
}
- String16 getDeviceId() const {
+ std::string getDeviceId() const {
return mDeviceId;
}
@@ -154,7 +142,7 @@
return mType;
}
- String16 getAddress() const {
+ std::string getAddress() const {
return mAddress;
}
@@ -170,19 +158,19 @@
return mIndexInTheGroup;
}
- const Vector<float>& getGeometricLocation() const {
+ const std::vector<float>& getGeometricLocation() const {
return mGeometricLocation;
}
- const Vector<float>& getOrientation() const {
+ const std::vector<float>& getOrientation() const {
return mOrientation;
}
- const Vector<Vector<float>>& getFrequencyResponses() const {
+ const std::vector<std::vector<float>>& getFrequencyResponses() const {
return mFrequencyResponses;
}
- const Vector<int>& getChannelMapping() const {
+ const std::vector<int>& getChannelMapping() const {
return mChannelMapping;
}
@@ -203,40 +191,17 @@
}
private:
- status_t readFloatVector(
- const Parcel* parcel, Vector<float> *vectorPtr, size_t defaultLength) {
- std::optional<std::vector<float>> v;
- status_t result = parcel->readFloatVector(&v);
- if (result != OK) return result;
- vectorPtr->clear();
- if (v) {
- for (const auto& iter : *v) {
- vectorPtr->push_back(iter);
- }
- } else {
- vectorPtr->resize(defaultLength);
- }
- return OK;
- }
- status_t writeFloatVector(Parcel* parcel, const Vector<float>& vector) const {
- std::vector<float> v;
- for (size_t i = 0; i < vector.size(); i++) {
- v.push_back(vector[i]);
- }
- return parcel->writeFloatVector(v);
- }
-
- String16 mDeviceId;
+ std::string mDeviceId;
int32_t mPortId;
uint32_t mType;
- String16 mAddress;
+ std::string mAddress;
int32_t mDeviceLocation;
int32_t mDeviceGroup;
int32_t mIndexInTheGroup;
- Vector<float> mGeometricLocation;
- Vector<float> mOrientation;
- Vector<Vector<float>> mFrequencyResponses;
- Vector<int> mChannelMapping;
+ std::vector<float> mGeometricLocation;
+ std::vector<float> mOrientation;
+ std::vector<std::vector<float>> mFrequencyResponses;
+ std::vector<int> mChannelMapping;
float mSensitivity;
float mMaxSpl;
float mMinSpl;
diff --git a/media/libaudioclient/AidlConversion.cpp b/media/libaudioclient/AidlConversion.cpp
index 681d4d7..8fb836d 100644
--- a/media/libaudioclient/AidlConversion.cpp
+++ b/media/libaudioclient/AidlConversion.cpp
@@ -16,7 +16,6 @@
#define LOG_TAG "AidlConversion"
//#define LOG_NDEBUG 0
-#include <system/audio.h>
#include <utils/Log.h>
#include "media/AidlConversion.h"
@@ -135,12 +134,6 @@
////////////////////////////////////////////////////////////////////////////////////////////////////
-template<typename To, typename From>
-ConversionResult<To> convertReinterpret(From from) {
- static_assert(sizeof(From) == sizeof(To));
- return static_cast<To>(from);
-}
-
enum class Direction {
INPUT, OUTPUT
};
@@ -290,6 +283,14 @@
return std::string(String8(legacy).c_str());
}
+ConversionResult<String8> aidl2legacy_string_view_String8(std::string_view aidl) {
+ return String8(aidl.data(), aidl.size());
+}
+
+ConversionResult<std::string> legacy2aidl_String8_string(const String8& legacy) {
+ return std::string(legacy.c_str());
+}
+
// The legacy enum is unnamed. Thus, we use int.
ConversionResult<int> aidl2legacy_AudioPortConfigType(media::AudioPortConfigType aidl) {
switch (aidl) {
@@ -1513,7 +1514,7 @@
}
ConversionResult<audio_encapsulation_mode_t>
-aidl2legacy_audio_encapsulation_mode_t_AudioEncapsulationMode(media::AudioEncapsulationMode aidl) {
+aidl2legacy_AudioEncapsulationMode_audio_encapsulation_mode_t(media::AudioEncapsulationMode aidl) {
switch (aidl) {
case media::AudioEncapsulationMode::NONE:
return AUDIO_ENCAPSULATION_MODE_NONE;
@@ -1526,7 +1527,7 @@
}
ConversionResult<media::AudioEncapsulationMode>
-legacy2aidl_AudioEncapsulationMode_audio_encapsulation_mode_t(audio_encapsulation_mode_t legacy) {
+legacy2aidl_audio_encapsulation_mode_t_AudioEncapsulationMode(audio_encapsulation_mode_t legacy) {
switch (legacy) {
case AUDIO_ENCAPSULATION_MODE_NONE:
return media::AudioEncapsulationMode::NONE;
@@ -1558,7 +1559,7 @@
legacy.offload_buffer_size = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.offloadBufferSize));
legacy.usage = VALUE_OR_RETURN(aidl2legacy_AudioUsage_audio_usage_t(aidl.usage));
legacy.encapsulation_mode = VALUE_OR_RETURN(
- aidl2legacy_audio_encapsulation_mode_t_AudioEncapsulationMode(aidl.encapsulationMode));
+ aidl2legacy_AudioEncapsulationMode_audio_encapsulation_mode_t(aidl.encapsulationMode));
legacy.content_id = VALUE_OR_RETURN(convertReinterpret<int32_t>(aidl.contentId));
legacy.sync_id = VALUE_OR_RETURN(convertReinterpret<int32_t>(aidl.syncId));
return legacy;
@@ -1593,7 +1594,7 @@
return unexpected(BAD_VALUE);
}
aidl.encapsulationMode = VALUE_OR_RETURN(
- legacy2aidl_AudioEncapsulationMode_audio_encapsulation_mode_t(
+ legacy2aidl_audio_encapsulation_mode_t_AudioEncapsulationMode(
legacy.encapsulation_mode));
aidl.contentId = VALUE_OR_RETURN(convertReinterpret<int32_t>(legacy.content_id));
aidl.syncId = VALUE_OR_RETURN(convertReinterpret<int32_t>(legacy.sync_id));
@@ -1683,4 +1684,149 @@
return aidl;
}
+ConversionResult<AudioTimestamp>
+aidl2legacy_AudioTimestamp(const media::AudioTimestampInternal& aidl) {
+ AudioTimestamp legacy;
+ legacy.mPosition = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.position));
+ legacy.mTime.tv_sec = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.sec));
+ legacy.mTime.tv_nsec = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.nsec));
+ return legacy;
+}
+
+ConversionResult<media::AudioTimestampInternal>
+legacy2aidl_AudioTimestamp(const AudioTimestamp& legacy) {
+ media::AudioTimestampInternal aidl;
+ aidl.position = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.mPosition));
+ aidl.sec = VALUE_OR_RETURN(convertIntegral<int64_t>(legacy.mTime.tv_sec));
+ aidl.nsec = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.mTime.tv_nsec));
+ return aidl;
+}
+
+ConversionResult<audio_uuid_t>
+aidl2legacy_AudioUuid_audio_uuid_t(const media::AudioUuid& aidl) {
+ audio_uuid_t legacy;
+ legacy.timeLow = VALUE_OR_RETURN(convertReinterpret<uint32_t>(aidl.timeLow));
+ legacy.timeMid = VALUE_OR_RETURN(convertIntegral<uint16_t>(aidl.timeMid));
+ legacy.timeHiAndVersion = VALUE_OR_RETURN(convertIntegral<uint16_t>(aidl.timeHiAndVersion));
+ legacy.clockSeq = VALUE_OR_RETURN(convertIntegral<uint16_t>(aidl.clockSeq));
+ if (aidl.node.size() != std::size(legacy.node)) {
+ return unexpected(BAD_VALUE);
+ }
+ std::copy(aidl.node.begin(), aidl.node.end(), legacy.node);
+ return legacy;
+}
+
+ConversionResult<media::AudioUuid>
+legacy2aidl_audio_uuid_t_AudioUuid(const audio_uuid_t& legacy) {
+ media::AudioUuid aidl;
+ aidl.timeLow = VALUE_OR_RETURN(convertReinterpret<int32_t>(legacy.timeLow));
+ aidl.timeMid = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.timeMid));
+ aidl.timeHiAndVersion = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.timeHiAndVersion));
+ aidl.clockSeq = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.clockSeq));
+ std::copy(legacy.node, legacy.node + std::size(legacy.node), std::back_inserter(aidl.node));
+ return aidl;
+}
+
+ConversionResult<effect_descriptor_t>
+aidl2legacy_EffectDescriptor_effect_descriptor_t(const media::EffectDescriptor& aidl) {
+ effect_descriptor_t legacy;
+ legacy.type = VALUE_OR_RETURN(aidl2legacy_AudioUuid_audio_uuid_t(aidl.type));
+ legacy.uuid = VALUE_OR_RETURN(aidl2legacy_AudioUuid_audio_uuid_t(aidl.uuid));
+ legacy.apiVersion = VALUE_OR_RETURN(convertReinterpret<uint32_t>(aidl.apiVersion));
+ legacy.flags = VALUE_OR_RETURN(convertReinterpret<uint32_t>(aidl.flags));
+ legacy.cpuLoad = VALUE_OR_RETURN(convertIntegral<uint16_t>(aidl.cpuLoad));
+ legacy.memoryUsage = VALUE_OR_RETURN(convertIntegral<uint16_t>(aidl.memoryUsage));
+ RETURN_IF_ERROR(aidl2legacy_string(aidl.name, legacy.name, sizeof(legacy.name)));
+ RETURN_IF_ERROR(
+ aidl2legacy_string(aidl.implementor, legacy.implementor, sizeof(legacy.implementor)));
+ return legacy;
+}
+
+ConversionResult<media::EffectDescriptor>
+legacy2aidl_effect_descriptor_t_EffectDescriptor(const effect_descriptor_t& legacy) {
+ media::EffectDescriptor aidl;
+ aidl.type = VALUE_OR_RETURN(legacy2aidl_audio_uuid_t_AudioUuid(legacy.type));
+ aidl.uuid = VALUE_OR_RETURN(legacy2aidl_audio_uuid_t_AudioUuid(legacy.uuid));
+ aidl.apiVersion = VALUE_OR_RETURN(convertReinterpret<int32_t>(legacy.apiVersion));
+ aidl.flags = VALUE_OR_RETURN(convertReinterpret<int32_t>(legacy.flags));
+ aidl.cpuLoad = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.cpuLoad));
+ aidl.memoryUsage = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.memoryUsage));
+ aidl.name = VALUE_OR_RETURN(legacy2aidl_string(legacy.name, sizeof(legacy.name)));
+ aidl.implementor = VALUE_OR_RETURN(
+ legacy2aidl_string(legacy.implementor, sizeof(legacy.implementor)));
+ return aidl;
+}
+
+ConversionResult<audio_encapsulation_metadata_type_t>
+aidl2legacy_AudioEncapsulationMetadataType_audio_encapsulation_metadata_type_t(
+ media::AudioEncapsulationMetadataType aidl) {
+ switch (aidl) {
+ case media::AudioEncapsulationMetadataType::NONE:
+ return AUDIO_ENCAPSULATION_METADATA_TYPE_NONE;
+ case media::AudioEncapsulationMetadataType::FRAMEWORK_TUNER:
+ return AUDIO_ENCAPSULATION_METADATA_TYPE_FRAMEWORK_TUNER;
+ case media::AudioEncapsulationMetadataType::DVB_AD_DESCRIPTOR:
+ return AUDIO_ENCAPSULATION_METADATA_TYPE_DVB_AD_DESCRIPTOR;
+ }
+ return unexpected(BAD_VALUE);
+}
+
+ConversionResult<media::AudioEncapsulationMetadataType>
+legacy2aidl_audio_encapsulation_metadata_type_t_AudioEncapsulationMetadataType(
+ audio_encapsulation_metadata_type_t legacy) {
+ switch (legacy) {
+ case AUDIO_ENCAPSULATION_METADATA_TYPE_NONE:
+ return media::AudioEncapsulationMetadataType::NONE;
+ case AUDIO_ENCAPSULATION_METADATA_TYPE_FRAMEWORK_TUNER:
+ return media::AudioEncapsulationMetadataType::FRAMEWORK_TUNER;
+ case AUDIO_ENCAPSULATION_METADATA_TYPE_DVB_AD_DESCRIPTOR:
+ return media::AudioEncapsulationMetadataType::DVB_AD_DESCRIPTOR;
+ }
+ return unexpected(BAD_VALUE);
+}
+
+ConversionResult<uint32_t>
+aidl2legacy_AudioEncapsulationMode_mask(int32_t aidl) {
+ return convertBitmask<uint32_t,
+ int32_t,
+ audio_encapsulation_mode_t,
+ media::AudioEncapsulationMode>(
+ aidl, aidl2legacy_AudioEncapsulationMode_audio_encapsulation_mode_t,
+ index2enum_index<media::AudioEncapsulationMode>,
+ enumToMask_index<uint32_t, audio_encapsulation_mode_t>);
+}
+
+ConversionResult<int32_t>
+legacy2aidl_AudioEncapsulationMode_mask(uint32_t legacy) {
+ return convertBitmask<int32_t,
+ uint32_t,
+ media::AudioEncapsulationMode,
+ audio_encapsulation_mode_t>(
+ legacy, legacy2aidl_audio_encapsulation_mode_t_AudioEncapsulationMode,
+ index2enum_index<audio_encapsulation_mode_t>,
+ enumToMask_index<int32_t, media::AudioEncapsulationMode>);
+}
+
+ConversionResult<uint32_t>
+aidl2legacy_AudioEncapsulationMetadataType_mask(int32_t aidl) {
+ return convertBitmask<uint32_t,
+ int32_t,
+ audio_encapsulation_metadata_type_t,
+ media::AudioEncapsulationMetadataType>(
+ aidl, aidl2legacy_AudioEncapsulationMetadataType_audio_encapsulation_metadata_type_t,
+ index2enum_index<media::AudioEncapsulationMetadataType>,
+ enumToMask_index<uint32_t, audio_encapsulation_metadata_type_t>);
+}
+
+ConversionResult<int32_t>
+legacy2aidl_AudioEncapsulationMetadataType_mask(uint32_t legacy) {
+ return convertBitmask<int32_t,
+ uint32_t,
+ media::AudioEncapsulationMetadataType,
+ audio_encapsulation_metadata_type_t>(
+ legacy, legacy2aidl_audio_encapsulation_metadata_type_t_AudioEncapsulationMetadataType,
+ index2enum_index<audio_encapsulation_metadata_type_t>,
+ enumToMask_index<int32_t, media::AudioEncapsulationMetadataType>);
+}
+
} // namespace android
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index c23c38c..299df96 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -73,7 +73,6 @@
// AIDL files for audioclient interfaces
// The headers for these interfaces will be available to any modules that
// include libaudioclient, at the path "aidl/package/path/BnFoo.h"
- ":libaudioclient_aidl_private",
":libaudioclient_aidl",
"AudioEffect.cpp",
@@ -84,7 +83,6 @@
"IAudioFlinger.cpp",
"IAudioPolicyService.cpp",
"IAudioPolicyServiceClient.cpp",
- "IAudioTrack.cpp",
"ToneGenerator.cpp",
"PlayerBase.cpp",
"RecordingActivityTracker.cpp",
@@ -132,12 +130,12 @@
],
export_header_lib_headers: ["libaudioclient_headers"],
export_static_lib_headers: [
- "effect-aidl-cpp",
+ "effect-aidl-unstable-cpp",
"shared-file-region-aidl-unstable-cpp",
],
static_libs: [
- "effect-aidl-cpp",
+ "effect-aidl-unstable-cpp",
// for memory heap analysis
"libc_malloc_debug_backtrace",
"shared-file-region-aidl-unstable-cpp",
@@ -155,10 +153,46 @@
},
}
-cc_library_shared {
+// This is intended for clients needing to include AidlConversionUtil.h, without dragging in a lot of extra
+// dependencies.
+cc_library_headers {
+ name: "libaudioclient_aidl_conversion_util",
+ host_supported: true,
+ vendor_available: true,
+ double_loadable: true,
+ min_sdk_version: "29",
+ export_include_dirs: [
+ "include",
+ ],
+ header_libs: [
+ "libbase_headers",
+ ],
+ export_header_lib_headers: [
+ "libbase_headers",
+ ],
+ apex_available: [
+ "//apex_available:platform",
+ "com.android.bluetooth.updatable",
+ "com.android.media",
+ "com.android.media.swcodec",
+ ],
+}
+
+cc_library {
name: "libaudioclient_aidl_conversion",
srcs: ["AidlConversion.cpp"],
- local_include_dirs: ["include"],
+ export_include_dirs: ["include"],
+ host_supported: true,
+ vendor_available: true,
+ double_loadable: true,
+ min_sdk_version: "29",
+ header_libs: [
+ "libaudioclient_aidl_conversion_util",
+ "libaudio_system_headers",
+ ],
+ export_header_lib_headers: [
+ "libaudioclient_aidl_conversion_util",
+ ],
shared_libs: [
"audioclient-types-aidl-unstable-cpp",
"libbase",
@@ -195,16 +229,6 @@
path: "aidl",
}
-// Used to strip the "aidl/" from the path, so the build system can predict the
-// output filename.
-filegroup {
- name: "libaudioclient_aidl_private",
- srcs: [
- "aidl/android/media/IAudioRecord.aidl",
- ],
- path: "aidl",
-}
-
aidl_interface {
name: "capture_state_listener-aidl",
unstable: true,
@@ -218,6 +242,9 @@
name: "effect-aidl",
unstable: true,
local_include_dir: "aidl",
+ host_supported: true,
+ double_loadable: true,
+ vendor_available: true,
srcs: [
"aidl/android/media/IEffect.aidl",
"aidl/android/media/IEffectClient.aidl",
@@ -240,17 +267,22 @@
"aidl/android/media/AudioConfig.aidl",
"aidl/android/media/AudioConfigBase.aidl",
"aidl/android/media/AudioContentType.aidl",
+ "aidl/android/media/AudioDevice.aidl",
"aidl/android/media/AudioEncapsulationMode.aidl",
+ "aidl/android/media/AudioEncapsulationMetadataType.aidl",
"aidl/android/media/AudioFlag.aidl",
+ "aidl/android/media/AudioGain.aidl",
"aidl/android/media/AudioGainConfig.aidl",
"aidl/android/media/AudioGainMode.aidl",
"aidl/android/media/AudioInputFlags.aidl",
"aidl/android/media/AudioIoConfigEvent.aidl",
"aidl/android/media/AudioIoDescriptor.aidl",
"aidl/android/media/AudioIoFlags.aidl",
+ "aidl/android/media/AudioMode.aidl",
"aidl/android/media/AudioOffloadInfo.aidl",
"aidl/android/media/AudioOutputFlags.aidl",
"aidl/android/media/AudioPatch.aidl",
+ "aidl/android/media/AudioPort.aidl",
"aidl/android/media/AudioPortConfig.aidl",
"aidl/android/media/AudioPortConfigType.aidl",
"aidl/android/media/AudioPortConfigDeviceExt.aidl",
@@ -260,10 +292,16 @@
"aidl/android/media/AudioPortConfigSessionExt.aidl",
"aidl/android/media/AudioPortRole.aidl",
"aidl/android/media/AudioPortType.aidl",
+ "aidl/android/media/AudioProfile.aidl",
"aidl/android/media/AudioSourceType.aidl",
"aidl/android/media/AudioStreamType.aidl",
+ "aidl/android/media/AudioTimestampInternal.aidl",
+ "aidl/android/media/AudioUniqueIdUse.aidl",
"aidl/android/media/AudioUsage.aidl",
- ],
+ "aidl/android/media/AudioUuid.aidl",
+ "aidl/android/media/DeviceDescriptorBase.aidl",
+ "aidl/android/media/EffectDescriptor.aidl",
+ ],
imports: [
"audio_common-aidl",
],
@@ -285,16 +323,27 @@
host_supported: true,
vendor_available: true,
srcs: [
+ "aidl/android/media/CreateEffectRequest.aidl",
+ "aidl/android/media/CreateEffectResponse.aidl",
"aidl/android/media/CreateRecordRequest.aidl",
"aidl/android/media/CreateRecordResponse.aidl",
"aidl/android/media/CreateTrackRequest.aidl",
"aidl/android/media/CreateTrackResponse.aidl",
+ "aidl/android/media/OpenInputRequest.aidl",
+ "aidl/android/media/OpenInputResponse.aidl",
+ "aidl/android/media/OpenOutputRequest.aidl",
+ "aidl/android/media/OpenOutputResponse.aidl",
"aidl/android/media/IAudioFlingerClient.aidl",
+ "aidl/android/media/IAudioRecord.aidl",
+ "aidl/android/media/IAudioTrack.aidl",
"aidl/android/media/IAudioTrackCallback.aidl",
],
imports: [
+ "audio_common-aidl",
"audioclient-types-aidl",
+ "av-types-aidl",
+ "effect-aidl",
"shared-file-region-aidl",
],
double_loadable: true,
diff --git a/media/libaudioclient/AudioEffect.cpp b/media/libaudioclient/AudioEffect.cpp
index 1282474..ae899c0 100644
--- a/media/libaudioclient/AudioEffect.cpp
+++ b/media/libaudioclient/AudioEffect.cpp
@@ -101,9 +101,29 @@
mClientPid = IPCThreadState::self()->getCallingPid();
mClientUid = IPCThreadState::self()->getCallingUid();
- iEffect = audioFlinger->createEffect((effect_descriptor_t *)&mDescriptor,
- mIEffectClient, priority, io, mSessionId, device, mOpPackageName, mClientPid,
- probe, &mStatus, &mId, &enabled);
+ media::CreateEffectRequest request;
+ request.desc = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_effect_descriptor_t_EffectDescriptor(mDescriptor));
+ request.client = mIEffectClient;
+ request.priority = priority;
+ request.output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(io));
+ request.sessionId = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_session_t_int32_t(mSessionId));
+ request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(device));
+ request.opPackageName = VALUE_OR_RETURN_STATUS(legacy2aidl_String16_string(mOpPackageName));
+ request.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(mClientPid));
+ request.probe = probe;
+
+ media::CreateEffectResponse response;
+
+ mStatus = audioFlinger->createEffect(request, &response);
+
+ if (mStatus == OK) {
+ mId = response.id;
+ enabled = response.enabled;
+ iEffect = response.effect;
+ mDescriptor = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_EffectDescriptor_effect_descriptor_t(response.desc));
+ }
// In probe mode, we stop here and return the status: the IEffect interface to
// audio flinger will not be retained. initCheck() will return the creation status
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index 4d9fbb0..f01b1d0 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -748,7 +748,6 @@
IAudioFlinger::CreateRecordInput input;
IAudioFlinger::CreateRecordOutput output;
audio_session_t originalSessionId;
- sp<media::IAudioRecord> record;
void *iMemPointer;
audio_track_cblk_t* cblk;
status_t status;
@@ -817,7 +816,7 @@
do {
media::CreateRecordResponse response;
- record = audioFlinger->createRecord(VALUE_OR_FATAL(input.toAidl()), response, &status);
+ status = audioFlinger->createRecord(VALUE_OR_FATAL(input.toAidl()), response);
output = VALUE_OR_FATAL(IAudioFlinger::CreateRecordOutput::fromAidl(response));
if (status == NO_ERROR) {
break;
@@ -893,7 +892,7 @@
IInterface::asBinder(mAudioRecord)->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
- mAudioRecord = record;
+ mAudioRecord = output.audioRecord;
mCblkMemory = output.cblk;
mBufferMemory = output.buffers;
IPCThreadState::self()->flushCommands();
@@ -1531,7 +1530,13 @@
status_t AudioRecord::getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones)
{
AutoMutex lock(mLock);
- return mAudioRecord->getActiveMicrophones(activeMicrophones).transactionError();
+ std::vector<media::MicrophoneInfoData> mics;
+ status_t status = mAudioRecord->getActiveMicrophones(&mics).transactionError();
+ activeMicrophones->resize(mics.size());
+ for (size_t i = 0; status == OK && i < mics.size(); ++i) {
+ status = activeMicrophones->at(i).readFromParcelable(mics[i]);
+ }
+ return status;
}
status_t AudioRecord::setPreferredMicrophoneDirection(audio_microphone_direction_t direction)
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index cfe5f3a..d163feb 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -33,7 +33,7 @@
#include <system/audio.h>
-#define VALUE_OR_RETURN_STATUS(x) \
+#define VALUE_OR_RETURN_BINDER_STATUS(x) \
({ auto _tmp = (x); \
if (!_tmp.ok()) return Status::fromStatusT(_tmp.error()); \
std::move(_tmp.value()); })
@@ -532,10 +532,10 @@
Status AudioSystem::AudioFlingerClient::ioConfigChanged(
media::AudioIoConfigEvent _event,
const media::AudioIoDescriptor& _ioDesc) {
- audio_io_config_event event = VALUE_OR_RETURN_STATUS(
+ audio_io_config_event event = VALUE_OR_RETURN_BINDER_STATUS(
aidl2legacy_AudioIoConfigEvent_audio_io_config_event(_event));
sp<AudioIoDescriptor> ioDesc(
- VALUE_OR_RETURN_STATUS(aidl2legacy_AudioIoDescriptor_AudioIoDescriptor(_ioDesc)));
+ VALUE_OR_RETURN_BINDER_STATUS(aidl2legacy_AudioIoDescriptor_AudioIoDescriptor(_ioDesc)));
ALOGV("ioConfigChanged() event %d", event);
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 14950a8..8c53c5b 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -784,7 +784,7 @@
int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags);
if (!(flags & CBLK_INVALID)) {
- status = mAudioTrack->start();
+ mAudioTrack->start(&status);
if (status == DEAD_OBJECT) {
flags |= CBLK_INVALID;
}
@@ -1477,7 +1477,8 @@
status_t AudioTrack::attachAuxEffect(int effectId)
{
AutoMutex lock(mLock);
- status_t status = mAudioTrack->attachAuxEffect(effectId);
+ status_t status;
+ mAudioTrack->attachAuxEffect(effectId, &status);
if (status == NO_ERROR) {
mAuxEffectId = effectId;
}
@@ -1607,9 +1608,7 @@
input.opPackageName = mOpPackageName;
media::CreateTrackResponse response;
- sp<IAudioTrack> track = audioFlinger->createTrack(VALUE_OR_FATAL(input.toAidl()),
- response,
- &status);
+ status = audioFlinger->createTrack(VALUE_OR_FATAL(input.toAidl()), response);
IAudioFlinger::CreateTrackOutput output = VALUE_OR_FATAL(
IAudioFlinger::CreateTrackOutput::fromAidl(
response));
@@ -1622,7 +1621,7 @@
}
goto exit;
}
- ALOG_ASSERT(track != 0);
+ ALOG_ASSERT(output.audioTrack != 0);
mFrameCount = output.frameCount;
mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
@@ -1644,7 +1643,9 @@
// so we are no longer responsible for releasing it.
// FIXME compare to AudioRecord
- sp<IMemory> iMem = track->getCblk();
+ std::optional<media::SharedFileRegion> sfr;
+ output.audioTrack->getCblk(&sfr);
+ sp<IMemory> iMem = VALUE_OR_FATAL(aidl2legacy_NullableSharedFileRegion_IMemory(sfr));
if (iMem == 0) {
ALOGE("%s(%d): Could not get control block", __func__, mPortId);
status = NO_INIT;
@@ -1665,7 +1666,7 @@
IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
- mAudioTrack = track;
+ mAudioTrack = output.audioTrack;
mCblkMemory = iMem;
IPCThreadState::self()->flushCommands();
@@ -1721,7 +1722,7 @@
}
}
- mAudioTrack->attachAuxEffect(mAuxEffectId);
+ mAudioTrack->attachAuxEffect(mAuxEffectId, &status);
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
@@ -1965,7 +1966,8 @@
ALOGW("%s(%d): releaseBuffer() track %p disabled due to previous underrun, restarting",
__func__, mPortId, this);
// FIXME ignoring status
- mAudioTrack->start();
+ status_t status;
+ mAudioTrack->start(&status);
}
}
@@ -2573,11 +2575,17 @@
if (shaper.isStarted()) {
operationToEnd->setNormalizedTime(1.f);
}
- return mAudioTrack->applyVolumeShaper(shaper.mConfiguration, operationToEnd);
+ media::VolumeShaperConfiguration config;
+ shaper.mConfiguration->writeToParcelable(&config);
+ media::VolumeShaperOperation operation;
+ operationToEnd->writeToParcelable(&operation);
+ status_t status;
+ mAudioTrack->applyVolumeShaper(config, operation, &status);
+ return status;
});
if (mState == STATE_ACTIVE) {
- result = mAudioTrack->start();
+ mAudioTrack->start(&result);
}
// server resets to zero so we offset
mFramesWrittenServerOffset =
@@ -2647,7 +2655,9 @@
status_t AudioTrack::setParameters(const String8& keyValuePairs)
{
AutoMutex lock(mLock);
- return mAudioTrack->setParameters(keyValuePairs);
+ status_t status;
+ mAudioTrack->setParameters(keyValuePairs.c_str(), &status);
+ return status;
}
status_t AudioTrack::selectPresentation(int presentationId, int programId)
@@ -2659,7 +2669,9 @@
ALOGV("%s(%d): PresentationId/ProgramId[%s]",
__func__, mPortId, param.toString().string());
- return mAudioTrack->setParameters(param.toString());
+ status_t status;
+ mAudioTrack->setParameters(param.toString().c_str(), &status);
+ return status;
}
VolumeShaper::Status AudioTrack::applyVolumeShaper(
@@ -2668,11 +2680,16 @@
{
AutoMutex lock(mLock);
mVolumeHandler->setIdIfNecessary(configuration);
- VolumeShaper::Status status = mAudioTrack->applyVolumeShaper(configuration, operation);
+ media::VolumeShaperConfiguration config;
+ configuration->writeToParcelable(&config);
+ media::VolumeShaperOperation op;
+ operation->writeToParcelable(&op);
+ VolumeShaper::Status status;
+ mAudioTrack->applyVolumeShaper(config, op, &status);
if (status == DEAD_OBJECT) {
if (restoreTrack_l("applyVolumeShaper") == OK) {
- status = mAudioTrack->applyVolumeShaper(configuration, operation);
+ mAudioTrack->applyVolumeShaper(config, op, &status);
}
}
if (status >= 0) {
@@ -2692,10 +2709,20 @@
sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id)
{
AutoMutex lock(mLock);
- sp<VolumeShaper::State> state = mAudioTrack->getVolumeShaperState(id);
+ std::optional<media::VolumeShaperState> vss;
+ mAudioTrack->getVolumeShaperState(id, &vss);
+ sp<VolumeShaper::State> state;
+ if (vss.has_value()) {
+ state = new VolumeShaper::State();
+ state->readFromParcelable(vss.value());
+ }
if (state.get() == nullptr && (mCblk->mFlags & CBLK_INVALID) != 0) {
if (restoreTrack_l("getVolumeShaperState") == OK) {
- state = mAudioTrack->getVolumeShaperState(id);
+ mAudioTrack->getVolumeShaperState(id, &vss);
+ if (vss.has_value()) {
+ state = new VolumeShaper::State();
+ state->readFromParcelable(vss.value());
+ }
}
}
return state;
@@ -2789,7 +2816,11 @@
status_t status;
if (isOffloadedOrDirect_l()) {
// use Binder to get timestamp
- status = mAudioTrack->getTimestamp(timestamp);
+ media::AudioTimestampInternal ts;
+ mAudioTrack->getTimestamp(&ts, &status);
+ if (status == OK) {
+ timestamp = VALUE_OR_FATAL(aidl2legacy_AudioTimestamp(ts));
+ }
} else {
// read timestamp from shared memory
ExtendedTimestamp ets;
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 771cfd9..5dfda09 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -153,6 +153,7 @@
aidl.afLatencyMs = VALUE_OR_RETURN(convertIntegral<int32_t>(afLatencyMs));
aidl.outputId = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(outputId));
aidl.portId = VALUE_OR_RETURN(legacy2aidl_audio_port_handle_t_int32_t(portId));
+ aidl.audioTrack = audioTrack;
return aidl;
}
@@ -173,6 +174,7 @@
legacy.afLatencyMs = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.afLatencyMs));
legacy.outputId = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_io_handle_t(aidl.outputId));
legacy.portId = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_port_handle_t(aidl.portId));
+ legacy.audioTrack = aidl.audioTrack;
return legacy;
}
@@ -226,6 +228,7 @@
aidl.cblk = VALUE_OR_RETURN(legacy2aidl_NullableIMemory_SharedFileRegion(cblk));
aidl.buffers = VALUE_OR_RETURN(legacy2aidl_NullableIMemory_SharedFileRegion(buffers));
aidl.portId = VALUE_OR_RETURN(legacy2aidl_audio_port_handle_t_int32_t(portId));
+ aidl.audioRecord = audioRecord;
return aidl;
}
@@ -245,6 +248,7 @@
legacy.cblk = VALUE_OR_RETURN(aidl2legacy_NullableSharedFileRegion_IMemory(aidl.cblk));
legacy.buffers = VALUE_OR_RETURN(aidl2legacy_NullableSharedFileRegion_IMemory(aidl.buffers));
legacy.portId = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_port_handle_t(aidl.portId));
+ legacy.audioRecord = aidl.audioRecord;
return legacy;
}
@@ -256,75 +260,58 @@
{
}
- virtual sp<IAudioTrack> createTrack(const media::CreateTrackRequest& input,
- media::CreateTrackResponse& output,
- status_t* status)
+ virtual status_t createTrack(const media::CreateTrackRequest& input,
+ media::CreateTrackResponse& output)
{
Parcel data, reply;
- sp<IAudioTrack> track;
+ status_t status;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-
- if (status == nullptr) {
- return track;
- }
-
data.writeParcelable(input);
status_t lStatus = remote()->transact(CREATE_TRACK, data, &reply);
if (lStatus != NO_ERROR) {
ALOGE("createTrack transaction error %d", lStatus);
- *status = DEAD_OBJECT;
- return track;
+ return DEAD_OBJECT;
}
- *status = reply.readInt32();
- if (*status != NO_ERROR) {
- ALOGE("createTrack returned error %d", *status);
- return track;
- }
- track = interface_cast<IAudioTrack>(reply.readStrongBinder());
- if (track == 0) {
- ALOGE("createTrack returned an NULL IAudioTrack with status OK");
- *status = DEAD_OBJECT;
- return track;
+ status = reply.readInt32();
+ if (status != NO_ERROR) {
+ ALOGE("createTrack returned error %d", status);
+ return status;
}
output.readFromParcel(&reply);
- return track;
+ if (output.audioTrack == 0) {
+ ALOGE("createTrack returned an NULL IAudioTrack with status OK");
+ return DEAD_OBJECT;
+ }
+ return OK;
}
- virtual sp<media::IAudioRecord> createRecord(const media::CreateRecordRequest& input,
- media::CreateRecordResponse& output,
- status_t* status)
+ virtual status_t createRecord(const media::CreateRecordRequest& input,
+ media::CreateRecordResponse& output)
{
Parcel data, reply;
- sp<media::IAudioRecord> record;
+ status_t status;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- if (status == nullptr) {
- return record;
- }
-
data.writeParcelable(input);
status_t lStatus = remote()->transact(CREATE_RECORD, data, &reply);
if (lStatus != NO_ERROR) {
ALOGE("createRecord transaction error %d", lStatus);
- *status = DEAD_OBJECT;
- return record;
+ return DEAD_OBJECT;
}
- *status = reply.readInt32();
- if (*status != NO_ERROR) {
- ALOGE("createRecord returned error %d", *status);
- return record;
+ status = reply.readInt32();
+ if (status != NO_ERROR) {
+ ALOGE("createRecord returned error %d", status);
+ return status;
}
- record = interface_cast<media::IAudioRecord>(reply.readStrongBinder());
- if (record == 0) {
- ALOGE("createRecord returned a NULL IAudioRecord with status OK");
- *status = DEAD_OBJECT;
- return record;
- }
output.readFromParcel(&reply);
- return record;
+ if (output.audioRecord == 0) {
+ ALOGE("createRecord returned a NULL IAudioRecord with status OK");
+ return DEAD_OBJECT;
+ }
+ return OK;
}
virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const
@@ -543,37 +530,16 @@
return reply.readInt64();
}
- virtual status_t openOutput(audio_module_handle_t module,
- audio_io_handle_t *output,
- audio_config_t *config,
- const sp<DeviceDescriptorBase>& device,
- uint32_t *latencyMs,
- audio_output_flags_t flags)
+ virtual status_t openOutput(const media::OpenOutputRequest& request,
+ media::OpenOutputResponse* response)
{
- if (output == nullptr || config == nullptr || device == nullptr || latencyMs == nullptr) {
- return BAD_VALUE;
- }
+ status_t status;
Parcel data, reply;
- data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(module);
- data.write(config, sizeof(audio_config_t));
- data.writeParcelable(*device);
- data.writeInt32((int32_t) flags);
- status_t status = remote()->transact(OPEN_OUTPUT, data, &reply);
- if (status != NO_ERROR) {
- *output = AUDIO_IO_HANDLE_NONE;
- return status;
- }
- status = (status_t)reply.readInt32();
- if (status != NO_ERROR) {
- *output = AUDIO_IO_HANDLE_NONE;
- return status;
- }
- *output = (audio_io_handle_t)reply.readInt32();
- ALOGV("openOutput() returned output, %d", *output);
- reply.read(config, sizeof(audio_config_t));
- *latencyMs = reply.readInt32();
- return NO_ERROR;
+ return data.writeParcelable(request)
+ ?: remote()->transact(OPEN_OUTPUT, data, &reply)
+ ?: data.readInt32(&status)
+ ?: status
+ ?: data.readParcelable(response);
}
virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
@@ -614,40 +580,17 @@
return reply.readInt32();
}
- virtual status_t openInput(audio_module_handle_t module,
- audio_io_handle_t *input,
- audio_config_t *config,
- audio_devices_t *device,
- const String8& address,
- audio_source_t source,
- audio_input_flags_t flags)
+ virtual status_t openInput(const media::OpenInputRequest& request,
+ media::OpenInputResponse* response)
{
- if (input == NULL || config == NULL || device == NULL) {
- return BAD_VALUE;
- }
Parcel data, reply;
- data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(module);
- data.writeInt32(*input);
- data.write(config, sizeof(audio_config_t));
- data.writeInt32(*device);
- data.writeString8(address);
- data.writeInt32(source);
- data.writeInt32(flags);
- status_t status = remote()->transact(OPEN_INPUT, data, &reply);
- if (status != NO_ERROR) {
- *input = AUDIO_IO_HANDLE_NONE;
- return status;
- }
- status = (status_t)reply.readInt32();
- if (status != NO_ERROR) {
- *input = AUDIO_IO_HANDLE_NONE;
- return status;
- }
- *input = (audio_io_handle_t)reply.readInt32();
- reply.read(config, sizeof(audio_config_t));
- *device = (audio_devices_t)reply.readInt32();
- return NO_ERROR;
+ status_t status;
+ return data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor())
+ ?: data.writeParcelable(request)
+ ?: remote()->transact(OPEN_INPUT, data, &reply)
+ ?: reply.readInt32(&status)
+ ?: status
+ ?: reply.readParcelable(response);
}
virtual status_t closeInput(int input)
@@ -805,66 +748,25 @@
return NO_ERROR;
}
- virtual sp<media::IEffect> createEffect(
- effect_descriptor_t *pDesc,
- const sp<media::IEffectClient>& client,
- int32_t priority,
- audio_io_handle_t output,
- audio_session_t sessionId,
- const AudioDeviceTypeAddr& device,
- const String16& opPackageName,
- pid_t pid,
- bool probe,
- status_t *status,
- int *id,
- int *enabled)
+ virtual status_t createEffect(const media::CreateEffectRequest& request,
+ media::CreateEffectResponse* response)
{
Parcel data, reply;
sp<media::IEffect> effect;
- if (pDesc == NULL) {
- if (status != NULL) {
- *status = BAD_VALUE;
- }
- return nullptr;
+ if (response == nullptr) {
+ return BAD_VALUE;
}
-
- data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.write(pDesc, sizeof(effect_descriptor_t));
- data.writeStrongBinder(IInterface::asBinder(client));
- data.writeInt32(priority);
- data.writeInt32((int32_t) output);
- data.writeInt32(sessionId);
- if (data.writeParcelable(device) != NO_ERROR) {
- if (status != NULL) {
- *status = NO_INIT;
- }
- return nullptr;
- }
- data.writeString16(opPackageName);
- data.writeInt32((int32_t) pid);
- data.writeInt32(probe ? 1 : 0);
-
- status_t lStatus = remote()->transact(CREATE_EFFECT, data, &reply);
+ status_t status;
+ status_t lStatus = data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor())
+ ?: data.writeParcelable(request)
+ ?: remote()->transact(CREATE_EFFECT, data, &reply)
+ ?: reply.readInt32(&status)
+ ?: reply.readParcelable(response)
+ ?: status;
if (lStatus != NO_ERROR) {
ALOGE("createEffect error: %s", strerror(-lStatus));
- } else {
- lStatus = reply.readInt32();
- int tmp = reply.readInt32();
- if (id != NULL) {
- *id = tmp;
- }
- tmp = reply.readInt32();
- if (enabled != NULL) {
- *enabled = tmp;
- }
- effect = interface_cast<media::IEffect>(reply.readStrongBinder());
- reply.read(pDesc, sizeof(effect_descriptor_t));
}
- if (status != NULL) {
- *status = lStatus;
- }
-
- return effect;
+ return lStatus;
}
virtual status_t moveEffects(audio_session_t session, audio_io_handle_t srcOutput,
@@ -1199,16 +1101,13 @@
status_t status;
media::CreateTrackResponse output;
- sp<IAudioTrack> track= createTrack(input,
- output,
- &status);
+ status = createTrack(input, output);
- LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR));
+ LOG_ALWAYS_FATAL_IF((output.audioTrack != 0) != (status == NO_ERROR));
reply->writeInt32(status);
if (status != NO_ERROR) {
return NO_ERROR;
}
- reply->writeStrongBinder(IInterface::asBinder(track));
output.writeToParcel(reply);
return NO_ERROR;
} break;
@@ -1224,16 +1123,13 @@
status_t status;
media::CreateRecordResponse output;
- sp<media::IAudioRecord> record = createRecord(input,
- output,
- &status);
+ status = createRecord(input, output);
- LOG_ALWAYS_FATAL_IF((record != 0) != (status == NO_ERROR));
+ LOG_ALWAYS_FATAL_IF((output.audioRecord != 0) != (status == NO_ERROR));
reply->writeInt32(status);
if (status != NO_ERROR) {
return NO_ERROR;
}
- reply->writeStrongBinder(IInterface::asBinder(record));
output.writeToParcel(reply);
return NO_ERROR;
} break;
@@ -1378,29 +1274,13 @@
} break;
case OPEN_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- audio_module_handle_t module = (audio_module_handle_t)data.readInt32();
- audio_config_t config = {};
- if (data.read(&config, sizeof(audio_config_t)) != NO_ERROR) {
- ALOGE("b/23905951");
- }
- sp<DeviceDescriptorBase> device = new DeviceDescriptorBase(AUDIO_DEVICE_NONE);
- status_t status = NO_ERROR;
- if ((status = data.readParcelable(device.get())) != NO_ERROR) {
- reply->writeInt32((int32_t)status);
- return NO_ERROR;
- }
- audio_output_flags_t flags = (audio_output_flags_t) data.readInt32();
- uint32_t latencyMs = 0;
- audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- status = openOutput(module, &output, &config, device, &latencyMs, flags);
- ALOGV("OPEN_OUTPUT output, %d", output);
- reply->writeInt32((int32_t)status);
- if (status == NO_ERROR) {
- reply->writeInt32((int32_t)output);
- reply->write(&config, sizeof(audio_config_t));
- reply->writeInt32(latencyMs);
- }
- return NO_ERROR;
+ status_t status;
+ media::OpenOutputRequest request;
+ media::OpenOutputResponse response;
+ return data.readParcelable(&request)
+ ?: (status = openOutput(request, &response), OK)
+ ?: reply->writeInt32(status)
+ ?: reply->writeParcelable(response);
} break;
case OPEN_DUPLICATE_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
@@ -1426,26 +1306,13 @@
} break;
case OPEN_INPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- audio_module_handle_t module = (audio_module_handle_t)data.readInt32();
- audio_io_handle_t input = (audio_io_handle_t)data.readInt32();
- audio_config_t config = {};
- if (data.read(&config, sizeof(audio_config_t)) != NO_ERROR) {
- ALOGE("b/23905951");
- }
- audio_devices_t device = (audio_devices_t)data.readInt32();
- String8 address(data.readString8());
- audio_source_t source = (audio_source_t)data.readInt32();
- audio_input_flags_t flags = (audio_input_flags_t) data.readInt32();
-
- status_t status = openInput(module, &input, &config,
- &device, address, source, flags);
- reply->writeInt32((int32_t) status);
- if (status == NO_ERROR) {
- reply->writeInt32((int32_t) input);
- reply->write(&config, sizeof(audio_config_t));
- reply->writeInt32(device);
- }
- return NO_ERROR;
+ media::OpenInputRequest request;
+ media::OpenInputResponse response;
+ status_t status;
+ return data.readParcelable(&request)
+ ?: (status = openInput(request, &response), OK)
+ ?: reply->writeInt32(status)
+ ?: reply->writeParcelable(response);
} break;
case CLOSE_INPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
@@ -1544,35 +1411,13 @@
}
case CREATE_EFFECT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- effect_descriptor_t desc = {};
- if (data.read(&desc, sizeof(effect_descriptor_t)) != NO_ERROR) {
- ALOGE("b/23905951");
- }
- sp<media::IEffectClient> client =
- interface_cast<media::IEffectClient>(data.readStrongBinder());
- int32_t priority = data.readInt32();
- audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
- audio_session_t sessionId = (audio_session_t) data.readInt32();
- AudioDeviceTypeAddr device;
- status_t status = NO_ERROR;
- if ((status = data.readParcelable(&device)) != NO_ERROR) {
- return status;
- }
- const String16 opPackageName = data.readString16();
- pid_t pid = (pid_t)data.readInt32();
- bool probe = data.readInt32() == 1;
- int id = 0;
- int enabled = 0;
+ media::CreateEffectRequest request;
+ media::CreateEffectResponse response;
- sp<media::IEffect> effect = createEffect(&desc, client, priority, output, sessionId,
- device, opPackageName, pid, probe, &status, &id, &enabled);
- reply->writeInt32(status);
- reply->writeInt32(id);
- reply->writeInt32(enabled);
- reply->writeStrongBinder(IInterface::asBinder(effect));
- reply->write(&desc, sizeof(effect_descriptor_t));
- return NO_ERROR;
+ return data.readParcelable(&request)
+ ?: reply->writeInt32(createEffect(request, &response))
+ ?: reply->writeParcelable(response);
} break;
case MOVE_EFFECTS: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
diff --git a/media/libaudioclient/IAudioTrack.cpp b/media/libaudioclient/IAudioTrack.cpp
deleted file mode 100644
index 6219e7a..0000000
--- a/media/libaudioclient/IAudioTrack.cpp
+++ /dev/null
@@ -1,317 +0,0 @@
-/*
-**
-** Copyright 2007, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-** http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-#define LOG_TAG "IAudioTrack"
-//#define LOG_NDEBUG 0
-#include <utils/Log.h>
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <binder/Parcel.h>
-
-#include <media/IAudioTrack.h>
-
-namespace android {
-
-using media::VolumeShaper;
-
-enum {
- GET_CBLK = IBinder::FIRST_CALL_TRANSACTION,
- START,
- STOP,
- FLUSH,
- RESERVED, // was MUTE
- PAUSE,
- ATTACH_AUX_EFFECT,
- SET_PARAMETERS,
- SELECT_PRESENTATION,
- GET_TIMESTAMP,
- SIGNAL,
- APPLY_VOLUME_SHAPER,
- GET_VOLUME_SHAPER_STATE,
-};
-
-class BpAudioTrack : public BpInterface<IAudioTrack>
-{
-public:
- explicit BpAudioTrack(const sp<IBinder>& impl)
- : BpInterface<IAudioTrack>(impl)
- {
- }
-
- virtual sp<IMemory> getCblk() const
- {
- Parcel data, reply;
- sp<IMemory> cblk;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- status_t status = remote()->transact(GET_CBLK, data, &reply);
- if (status == NO_ERROR) {
- cblk = interface_cast<IMemory>(reply.readStrongBinder());
- if (cblk != 0 && cblk->unsecurePointer() == NULL) {
- cblk.clear();
- }
- }
- return cblk;
- }
-
- virtual status_t start()
- {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- status_t status = remote()->transact(START, data, &reply);
- if (status == NO_ERROR) {
- status = reply.readInt32();
- } else {
- ALOGW("start() error: %s", strerror(-status));
- }
- return status;
- }
-
- virtual void stop()
- {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- remote()->transact(STOP, data, &reply);
- }
-
- virtual void flush()
- {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- remote()->transact(FLUSH, data, &reply);
- }
-
- virtual void pause()
- {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- remote()->transact(PAUSE, data, &reply);
- }
-
- virtual status_t attachAuxEffect(int effectId)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- data.writeInt32(effectId);
- status_t status = remote()->transact(ATTACH_AUX_EFFECT, data, &reply);
- if (status == NO_ERROR) {
- status = reply.readInt32();
- } else {
- ALOGW("attachAuxEffect() error: %s", strerror(-status));
- }
- return status;
- }
-
- virtual status_t setParameters(const String8& keyValuePairs) {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- data.writeString8(keyValuePairs);
- status_t status = remote()->transact(SET_PARAMETERS, data, &reply);
- if (status == NO_ERROR) {
- status = reply.readInt32();
- }
- return status;
- }
-
- /* Selects the presentation (if available) */
- virtual status_t selectPresentation(int presentationId, int programId) {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- data.writeInt32(presentationId);
- data.writeInt32(programId);
- status_t status = remote()->transact(SELECT_PRESENTATION, data, &reply);
- if (status == NO_ERROR) {
- status = reply.readInt32();
- }
- return status;
- }
-
- virtual status_t getTimestamp(AudioTimestamp& timestamp) {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- status_t status = remote()->transact(GET_TIMESTAMP, data, &reply);
- if (status == NO_ERROR) {
- status = reply.readInt32();
- if (status == NO_ERROR) {
- timestamp.mPosition = reply.readInt32();
- timestamp.mTime.tv_sec = reply.readInt32();
- timestamp.mTime.tv_nsec = reply.readInt32();
- }
- }
- return status;
- }
-
- virtual void signal() {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- remote()->transact(SIGNAL, data, &reply);
- }
-
- virtual VolumeShaper::Status applyVolumeShaper(
- const sp<VolumeShaper::Configuration>& configuration,
- const sp<VolumeShaper::Operation>& operation) {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
-
- status_t status = configuration.get() == nullptr
- ? data.writeInt32(0)
- : data.writeInt32(1)
- ?: configuration->writeToParcel(&data);
- if (status != NO_ERROR) {
- return VolumeShaper::Status(status);
- }
-
- status = operation.get() == nullptr
- ? status = data.writeInt32(0)
- : data.writeInt32(1)
- ?: operation->writeToParcel(&data);
- if (status != NO_ERROR) {
- return VolumeShaper::Status(status);
- }
-
- int32_t remoteVolumeShaperStatus;
- status = remote()->transact(APPLY_VOLUME_SHAPER, data, &reply)
- ?: reply.readInt32(&remoteVolumeShaperStatus);
-
- return VolumeShaper::Status(status ?: remoteVolumeShaperStatus);
- }
-
- virtual sp<VolumeShaper::State> getVolumeShaperState(int id) {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
-
- data.writeInt32(id);
- status_t status = remote()->transact(GET_VOLUME_SHAPER_STATE, data, &reply);
- if (status != NO_ERROR) {
- return nullptr;
- }
- sp<VolumeShaper::State> state = new VolumeShaper::State;
- status = state->readFromParcel(&reply);
- if (status != NO_ERROR) {
- return nullptr;
- }
- return state;
- }
-};
-
-IMPLEMENT_META_INTERFACE(AudioTrack, "android.media.IAudioTrack");
-
-// ----------------------------------------------------------------------
-
-status_t BnAudioTrack::onTransact(
- uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
- switch (code) {
- case GET_CBLK: {
- CHECK_INTERFACE(IAudioTrack, data, reply);
- reply->writeStrongBinder(IInterface::asBinder(getCblk()));
- return NO_ERROR;
- } break;
- case START: {
- CHECK_INTERFACE(IAudioTrack, data, reply);
- reply->writeInt32(start());
- return NO_ERROR;
- } break;
- case STOP: {
- CHECK_INTERFACE(IAudioTrack, data, reply);
- stop();
- return NO_ERROR;
- } break;
- case FLUSH: {
- CHECK_INTERFACE(IAudioTrack, data, reply);
- flush();
- return NO_ERROR;
- } break;
- case PAUSE: {
- CHECK_INTERFACE(IAudioTrack, data, reply);
- pause();
- return NO_ERROR;
- }
- case ATTACH_AUX_EFFECT: {
- CHECK_INTERFACE(IAudioTrack, data, reply);
- reply->writeInt32(attachAuxEffect(data.readInt32()));
- return NO_ERROR;
- } break;
- case SET_PARAMETERS: {
- CHECK_INTERFACE(IAudioTrack, data, reply);
- String8 keyValuePairs(data.readString8());
- reply->writeInt32(setParameters(keyValuePairs));
- return NO_ERROR;
- } break;
- case SELECT_PRESENTATION: {
- CHECK_INTERFACE(IAudioTrack, data, reply);
- reply->writeInt32(selectPresentation(data.readInt32(), data.readInt32()));
- return NO_ERROR;
- } break;
- case GET_TIMESTAMP: {
- CHECK_INTERFACE(IAudioTrack, data, reply);
- AudioTimestamp timestamp;
- status_t status = getTimestamp(timestamp);
- reply->writeInt32(status);
- if (status == NO_ERROR) {
- reply->writeInt32(timestamp.mPosition);
- reply->writeInt32(timestamp.mTime.tv_sec);
- reply->writeInt32(timestamp.mTime.tv_nsec);
- }
- return NO_ERROR;
- } break;
- case SIGNAL: {
- CHECK_INTERFACE(IAudioTrack, data, reply);
- signal();
- return NO_ERROR;
- } break;
- case APPLY_VOLUME_SHAPER: {
- CHECK_INTERFACE(IAudioTrack, data, reply);
- sp<VolumeShaper::Configuration> configuration;
- sp<VolumeShaper::Operation> operation;
-
- int32_t present;
- status_t status = data.readInt32(&present);
- if (status == NO_ERROR && present != 0) {
- configuration = new VolumeShaper::Configuration();
- status = configuration->readFromParcel(&data);
- }
- status = status ?: data.readInt32(&present);
- if (status == NO_ERROR && present != 0) {
- operation = new VolumeShaper::Operation();
- status = operation->readFromParcel(&data);
- }
- if (status == NO_ERROR) {
- status = (status_t)applyVolumeShaper(configuration, operation);
- }
- reply->writeInt32(status);
- return NO_ERROR;
- } break;
- case GET_VOLUME_SHAPER_STATE: {
- CHECK_INTERFACE(IAudioTrack, data, reply);
- int id;
- status_t status = data.readInt32(&id);
- if (status == NO_ERROR) {
- sp<VolumeShaper::State> state = getVolumeShaperState(id);
- if (state.get() != nullptr) {
- status = state->writeToParcel(reply);
- }
- }
- return NO_ERROR;
- } break;
- default:
- return BBinder::onTransact(code, data, reply, flags);
- }
-}
-
-} // namespace android
diff --git a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl b/media/libaudioclient/aidl/android/media/AudioDevice.aidl
similarity index 75%
rename from media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
rename to media/libaudioclient/aidl/android/media/AudioDevice.aidl
index d6e46cb..b200697 100644
--- a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioDevice.aidl
@@ -1,5 +1,5 @@
/*
- * Copyright 2018 The Android Open Source Project
+ * Copyright (C) 2020 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.
@@ -16,4 +16,11 @@
package android.media;
-parcelable MicrophoneInfo cpp_header "media/MicrophoneInfo.h";
+/**
+ * {@hide}
+ */
+parcelable AudioDevice {
+ /** Interpreted as audio_devices_t. */
+ int type;
+ @utf8InCpp String address;
+}
diff --git a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl b/media/libaudioclient/aidl/android/media/AudioEncapsulationMetadataType.aidl
similarity index 74%
copy from media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
copy to media/libaudioclient/aidl/android/media/AudioEncapsulationMetadataType.aidl
index d6e46cb..b03adfe 100644
--- a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioEncapsulationMetadataType.aidl
@@ -1,5 +1,5 @@
/*
- * Copyright 2018 The Android Open Source Project
+ * Copyright (C) 2020 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.
@@ -13,7 +13,14 @@
* See the License for the specific language governing permissions and
* limitations under the License.
*/
-
package android.media;
-parcelable MicrophoneInfo cpp_header "media/MicrophoneInfo.h";
+/**
+ * {@hide}
+ */
+@Backing(type="int")
+enum AudioEncapsulationMetadataType {
+ NONE = 0,
+ FRAMEWORK_TUNER = 1,
+ DVB_AD_DESCRIPTOR = 2,
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioEncapsulationMode.aidl b/media/libaudioclient/aidl/android/media/AudioEncapsulationMode.aidl
index 74a6141..9e04e82 100644
--- a/media/libaudioclient/aidl/android/media/AudioEncapsulationMode.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioEncapsulationMode.aidl
@@ -15,6 +15,9 @@
*/
package android.media;
+/**
+ * {@hide}
+ */
@Backing(type="int")
enum AudioEncapsulationMode {
NONE = 0,
diff --git a/media/libaudioclient/aidl/android/media/AudioFlag.aidl b/media/libaudioclient/aidl/android/media/AudioFlag.aidl
index 2602fe5..58b493b 100644
--- a/media/libaudioclient/aidl/android/media/AudioFlag.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioFlag.aidl
@@ -15,6 +15,9 @@
*/
package android.media;
+/**
+ * {@hide}
+ */
@Backing(type="int")
enum AudioFlag {
AUDIBILITY_ENFORCED = 0,
diff --git a/media/libaudioclient/aidl/android/media/AudioGain.aidl b/media/libaudioclient/aidl/android/media/AudioGain.aidl
new file mode 100644
index 0000000..048b295
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioGain.aidl
@@ -0,0 +1,36 @@
+/*
+ * Copyright (C) 2020 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.
+ */
+
+package android.media;
+
+/**
+ * {@hide}
+ */
+parcelable AudioGain {
+ int index;
+ boolean useInChannelMask;
+ boolean useForVolume;
+ /** Bitmask, indexed by AudioGainMode. */
+ int mode;
+ /** Interpreted as audio_channel_mask_t. */
+ int channelMask;
+ int minValue;
+ int maxValue;
+ int defaultValue;
+ int stepValue;
+ int minRampMs;
+ int maxRampMs;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioGainMode.aidl b/media/libaudioclient/aidl/android/media/AudioGainMode.aidl
index 39395e5..e1b9f0b 100644
--- a/media/libaudioclient/aidl/android/media/AudioGainMode.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioGainMode.aidl
@@ -15,6 +15,9 @@
*/
package android.media;
+/**
+ * {@hide}
+ */
@Backing(type="int")
enum AudioGainMode {
JOINT = 0,
diff --git a/media/libaudioclient/aidl/android/media/AudioInputFlags.aidl b/media/libaudioclient/aidl/android/media/AudioInputFlags.aidl
index 8f517e7..bfc0eb0 100644
--- a/media/libaudioclient/aidl/android/media/AudioInputFlags.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioInputFlags.aidl
@@ -15,6 +15,9 @@
*/
package android.media;
+/**
+ * {@hide}
+ */
@Backing(type="int")
enum AudioInputFlags {
FAST = 0,
diff --git a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl b/media/libaudioclient/aidl/android/media/AudioMode.aidl
similarity index 70%
copy from media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
copy to media/libaudioclient/aidl/android/media/AudioMode.aidl
index d6e46cb..7067dd3 100644
--- a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioMode.aidl
@@ -1,5 +1,5 @@
/*
- * Copyright 2018 The Android Open Source Project
+ * Copyright (C) 2020 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.
@@ -13,7 +13,18 @@
* See the License for the specific language governing permissions and
* limitations under the License.
*/
-
package android.media;
-parcelable MicrophoneInfo cpp_header "media/MicrophoneInfo.h";
+/**
+ * {@hide}
+ */
+@Backing(type="int")
+enum AudioMode {
+ INVALID = -2,
+ CURRENT = -1,
+ NORMAL = 0,
+ RINGTONE = 1,
+ IN_CALL = 2,
+ IN_COMMUNICATION = 3,
+ CALL_SCREEN = 4,
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioOutputFlags.aidl b/media/libaudioclient/aidl/android/media/AudioOutputFlags.aidl
index aebf871..cf11528 100644
--- a/media/libaudioclient/aidl/android/media/AudioOutputFlags.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioOutputFlags.aidl
@@ -15,6 +15,9 @@
*/
package android.media;
+/**
+ * {@hide}
+ */
@Backing(type="int")
enum AudioOutputFlags {
DIRECT = 0,
diff --git a/media/libaudioclient/aidl/android/media/AudioPort.aidl b/media/libaudioclient/aidl/android/media/AudioPort.aidl
new file mode 100644
index 0000000..1aa532b
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioPort.aidl
@@ -0,0 +1,35 @@
+/*
+ * Copyright (C) 2020 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.
+ */
+
+package android.media;
+
+import android.media.AudioGain;
+import android.media.AudioPortRole;
+import android.media.AudioPortType;
+import android.media.AudioProfile;
+
+/**
+ * {@hide}
+ */
+parcelable AudioPort {
+ /** Gain controllers. */
+ AudioGain[] gains;
+ @utf8InCpp String name;
+ AudioPortType type;
+ AudioPortRole role;
+ /** AudioProfiles supported by this port (format, Rates, Channels). */
+ AudioProfile[] profiles;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigType.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigType.aidl
index c7bb4d8..6e22b8d 100644
--- a/media/libaudioclient/aidl/android/media/AudioPortConfigType.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfigType.aidl
@@ -15,6 +15,9 @@
*/
package android.media;
+/**
+ * {@hide}
+ */
@Backing(type="int")
enum AudioPortConfigType {
SAMPLE_RATE = 0,
diff --git a/media/libaudioclient/aidl/android/media/AudioPortRole.aidl b/media/libaudioclient/aidl/android/media/AudioPortRole.aidl
index 3212325..ea2ef3a 100644
--- a/media/libaudioclient/aidl/android/media/AudioPortRole.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortRole.aidl
@@ -15,6 +15,9 @@
*/
package android.media;
+/**
+ * {@hide}
+ */
@Backing(type="int")
enum AudioPortRole {
NONE = 0,
diff --git a/media/libaudioclient/aidl/android/media/AudioPortType.aidl b/media/libaudioclient/aidl/android/media/AudioPortType.aidl
index 90eea9a..9e6af49 100644
--- a/media/libaudioclient/aidl/android/media/AudioPortType.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortType.aidl
@@ -15,6 +15,9 @@
*/
package android.media;
+/**
+ * {@hide}
+ */
@Backing(type="int")
enum AudioPortType {
NONE = 0,
diff --git a/media/libaudioclient/aidl/android/media/AudioProfile.aidl b/media/libaudioclient/aidl/android/media/AudioProfile.aidl
new file mode 100644
index 0000000..e5e8812
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioProfile.aidl
@@ -0,0 +1,34 @@
+/*
+ * Copyright (C) 2020 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.
+ */
+
+package android.media;
+
+import android.media.audio.common.AudioFormat;
+
+/**
+ * {@hide}
+ */
+parcelable AudioProfile {
+ @utf8InCpp String name;
+ /** The format for an audio profile should only be set when initialized. */
+ AudioFormat format;
+ /** Interpreted as audio_channel_mask_t. */
+ int[] channelMasks;
+ int[] samplingRates;
+ boolean isDynamicFormat;
+ boolean isDynamicChannels;
+ boolean isDynamicRate;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioSourceType.aidl b/media/libaudioclient/aidl/android/media/AudioSourceType.aidl
index 35320f8..8673b92 100644
--- a/media/libaudioclient/aidl/android/media/AudioSourceType.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioSourceType.aidl
@@ -15,6 +15,9 @@
*/
package android.media;
+/**
+ * {@hide}
+ */
@Backing(type="int")
enum AudioSourceType {
INVALID = -1,
diff --git a/media/libaudioclient/aidl/android/media/AudioStreamType.aidl b/media/libaudioclient/aidl/android/media/AudioStreamType.aidl
index 803b87b..d777882 100644
--- a/media/libaudioclient/aidl/android/media/AudioStreamType.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioStreamType.aidl
@@ -15,6 +15,9 @@
*/
package android.media;
+/**
+ * {@hide}
+ */
@Backing(type="int")
enum AudioStreamType {
DEFAULT = -1,
diff --git a/media/libaudioclient/aidl/android/media/AudioTimestampInternal.aidl b/media/libaudioclient/aidl/android/media/AudioTimestampInternal.aidl
new file mode 100644
index 0000000..8bbfb57
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioTimestampInternal.aidl
@@ -0,0 +1,30 @@
+/*
+ * Copyright (C) 2020 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.
+ */
+
+package android.media;
+
+/**
+ * The "Internal" timestamp is intended to disambiguate from the android.media.AudioTimestamp type.
+ *
+ * {@hide}
+ */
+parcelable AudioTimestampInternal {
+ /** A frame position in AudioTrack::getPosition() units. */
+ int position;
+ /** corresponding CLOCK_MONOTONIC when frame is expected to present. */
+ long sec;
+ int nsec;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioUniqueIdUse.aidl b/media/libaudioclient/aidl/android/media/AudioUniqueIdUse.aidl
new file mode 100644
index 0000000..fdb6d2d
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioUniqueIdUse.aidl
@@ -0,0 +1,35 @@
+/*
+ * Copyright (C) 2020 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.
+ */
+package android.media;
+
+/**
+ * {@hide}
+ */
+@Backing(type="int")
+enum AudioUniqueIdUse {
+ UNSPECIFIED = 0,
+ SESSION = 1, // audio_session_t
+ // for allocated sessions, not special AUDIO_SESSION_*
+ MODULE = 2, // audio_module_handle_t
+ EFFECT = 3, // audio_effect_handle_t
+ PATCH = 4, // audio_patch_handle_t
+ OUTPUT = 5, // audio_io_handle_t
+ INPUT = 6, // audio_io_handle_t
+ CLIENT = 7, // client-side players and recorders
+ // FIXME should move to a separate namespace;
+ // these IDs are allocated by AudioFlinger on client request,
+ // but are never used by AudioFlinger
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioUsage.aidl b/media/libaudioclient/aidl/android/media/AudioUsage.aidl
index 137e7ff..66c5c30 100644
--- a/media/libaudioclient/aidl/android/media/AudioUsage.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioUsage.aidl
@@ -15,6 +15,9 @@
*/
package android.media;
+/**
+ * {@hide}
+ */
@Backing(type="int")
enum AudioUsage {
UNKNOWN = 0,
diff --git a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl b/media/libaudioclient/aidl/android/media/AudioUuid.aidl
similarity index 73%
copy from media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
copy to media/libaudioclient/aidl/android/media/AudioUuid.aidl
index d6e46cb..bba9039 100644
--- a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioUuid.aidl
@@ -1,5 +1,5 @@
/*
- * Copyright 2018 The Android Open Source Project
+ * Copyright (C) 2020 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.
@@ -13,7 +13,15 @@
* See the License for the specific language governing permissions and
* limitations under the License.
*/
-
package android.media;
-parcelable MicrophoneInfo cpp_header "media/MicrophoneInfo.h";
+/**
+ * {@hide}
+ */
+parcelable AudioUuid {
+ int timeLow;
+ int timeMid;
+ int timeHiAndVersion;
+ int clockSeq;
+ byte[] node; // Length = 6
+}
diff --git a/media/libaudioclient/aidl/android/media/CreateEffectRequest.aidl b/media/libaudioclient/aidl/android/media/CreateEffectRequest.aidl
new file mode 100644
index 0000000..8368854
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/CreateEffectRequest.aidl
@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2020 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.
+ */
+
+package android.media;
+
+import android.media.AudioDevice;
+import android.media.EffectDescriptor;
+import android.media.IEffectClient;
+
+/**
+ * Input arguments of the createEffect() method.
+ *
+ * {@hide}
+ */
+parcelable CreateEffectRequest {
+ EffectDescriptor desc;
+ @nullable IEffectClient client;
+ int priority;
+ /** Interpreted as audio_io_handle_t. */
+ int output;
+ /** Interpreted as audio_session_t. */
+ int sessionId;
+ AudioDevice device;
+ @utf8InCpp String opPackageName;
+ /** Interpreted as pid_t. */
+ int pid;
+ boolean probe;
+}
diff --git a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl b/media/libaudioclient/aidl/android/media/CreateEffectResponse.aidl
similarity index 64%
copy from media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
copy to media/libaudioclient/aidl/android/media/CreateEffectResponse.aidl
index d6e46cb..0aa640a 100644
--- a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
+++ b/media/libaudioclient/aidl/android/media/CreateEffectResponse.aidl
@@ -1,5 +1,5 @@
/*
- * Copyright 2018 The Android Open Source Project
+ * Copyright (C) 2020 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.
@@ -16,4 +16,17 @@
package android.media;
-parcelable MicrophoneInfo cpp_header "media/MicrophoneInfo.h";
+import android.media.EffectDescriptor;
+import android.media.IEffect;
+
+/**
+ * Output arguments of the createEffect() method.
+ *
+ * {@hide}
+ */
+parcelable CreateEffectResponse {
+ int id;
+ boolean enabled;
+ @nullable IEffect effect;
+ EffectDescriptor desc;
+}
diff --git a/media/libaudioclient/aidl/android/media/CreateRecordResponse.aidl b/media/libaudioclient/aidl/android/media/CreateRecordResponse.aidl
index 0c9d7c3..d78b3fc 100644
--- a/media/libaudioclient/aidl/android/media/CreateRecordResponse.aidl
+++ b/media/libaudioclient/aidl/android/media/CreateRecordResponse.aidl
@@ -16,6 +16,7 @@
package android.media;
+import android.media.IAudioRecord;
import android.media.SharedFileRegion;
/**
@@ -40,4 +41,6 @@
@nullable SharedFileRegion buffers;
/** Interpreted as audio_port_handle_t. */
int portId;
+ /** The newly created record. */
+ @nullable IAudioRecord audioRecord;
}
diff --git a/media/libaudioclient/aidl/android/media/CreateTrackResponse.aidl b/media/libaudioclient/aidl/android/media/CreateTrackResponse.aidl
index 494e63f..6bdd8e4 100644
--- a/media/libaudioclient/aidl/android/media/CreateTrackResponse.aidl
+++ b/media/libaudioclient/aidl/android/media/CreateTrackResponse.aidl
@@ -16,6 +16,8 @@
package android.media;
+import android.media.IAudioTrack;
+
/**
* CreateTrackOutput contains all output arguments returned by AudioFlinger to AudioTrack
* when calling createTrack() including arguments that were passed as I/O for update by
@@ -39,4 +41,6 @@
int outputId;
/** Interpreted as audio_port_handle_t. */
int portId;
+ /** The newly created track. */
+ @nullable IAudioTrack audioTrack;
}
diff --git a/media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl b/media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl
new file mode 100644
index 0000000..aa0f149
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/DeviceDescriptorBase.aidl
@@ -0,0 +1,34 @@
+/*
+ * Copyright (C) 2020 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.
+ */
+
+package android.media;
+
+import android.media.AudioPort;
+import android.media.AudioPortConfig;
+import android.media.AudioDevice;
+
+/**
+ * {@hide}
+ */
+parcelable DeviceDescriptorBase {
+ AudioPort port;
+ AudioPortConfig portConfig;
+ AudioDevice device;
+ /** Bitmask, indexed by AudioEncapsulationMode. */
+ int encapsulationModes;
+ /** Bitmask, indexed by AudioEncapsulationMetadataType. */
+ int encapsulationMetadataTypes;
+}
diff --git a/media/libaudioclient/aidl/android/media/EffectDescriptor.aidl b/media/libaudioclient/aidl/android/media/EffectDescriptor.aidl
new file mode 100644
index 0000000..35a3d74
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/EffectDescriptor.aidl
@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2020 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.
+ */
+
+package android.media;
+
+import android.media.AudioUuid;
+
+/**
+ * {@hide}
+ */
+parcelable EffectDescriptor {
+ /** UUID of to the OpenSL ES interface implemented by this effect. */
+ AudioUuid type;
+ /** UUID for this particular implementation. */
+ AudioUuid uuid;
+ /** Version of the effect control API implemented. */
+ int apiVersion;
+ /** Effect engine capabilities/requirements flags. */
+ int flags;
+ /** CPU load indication.. */
+ int cpuLoad;
+ /** Data Memory usage.. */
+ int memoryUsage;
+ /** Human readable effect name. */
+ @utf8InCpp String name;
+ /** Human readable effect implementor name. */
+ @utf8InCpp String implementor;
+}
diff --git a/media/libaudioclient/aidl/android/media/IAudioRecord.aidl b/media/libaudioclient/aidl/android/media/IAudioRecord.aidl
index ecf58b6..1772653 100644
--- a/media/libaudioclient/aidl/android/media/IAudioRecord.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioRecord.aidl
@@ -16,9 +16,13 @@
package android.media;
-import android.media.MicrophoneInfo;
+import android.media.MicrophoneInfoData;
-/* Native code must specify namespace media (media::IAudioRecord) when referring to this class */
+/**
+ * Native code must specify namespace media (media::IAudioRecord) when referring to this class.
+ *
+ * {@hide}
+ */
interface IAudioRecord {
/* After it's created the track is not active. Call start() to
@@ -35,7 +39,7 @@
/* Get a list of current active microphones.
*/
- void getActiveMicrophones(out MicrophoneInfo[] activeMicrophones);
+ void getActiveMicrophones(out MicrophoneInfoData[] activeMicrophones);
/* Set the microphone direction (for processing).
*/
diff --git a/media/libaudioclient/aidl/android/media/IAudioTrack.aidl b/media/libaudioclient/aidl/android/media/IAudioTrack.aidl
new file mode 100644
index 0000000..2b6c362
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/IAudioTrack.aidl
@@ -0,0 +1,84 @@
+/*
+ * Copyright (C) 2020 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.
+ */
+
+package android.media;
+
+import android.media.AudioTimestampInternal;
+import android.media.SharedFileRegion;
+import android.media.VolumeShaperConfiguration;
+import android.media.VolumeShaperOperation;
+import android.media.VolumeShaperState;
+
+/**
+ * Unless otherwise noted, methods returning int expect it to be interpreted as a status_t.
+ *
+ * {@hide}
+ */
+interface IAudioTrack {
+ /** Get this track's control block */
+ @nullable SharedFileRegion getCblk();
+
+ /**
+ * After it's created the track is not active. Call start() to
+ * make it active.
+ */
+ int start();
+
+ /**
+ * Stop a track. If set, the callback will cease being called and
+ * obtainBuffer will return an error. Buffers that are already released
+ * will continue to be processed, unless/until flush() is called.
+ */
+ void stop();
+
+ /**
+ * Flush a stopped or paused track. All pending/released buffers are discarded.
+ * This function has no effect if the track is not stopped or paused.
+ */
+ void flush();
+
+ /**
+ * Pause a track. If set, the callback will cease being called and
+ * obtainBuffer will return an error. Buffers that are already released
+ * will continue to be processed, unless/until flush() is called.
+ */
+ void pause();
+
+ /**
+ * Attach track auxiliary output to specified effect. Use effectId = 0
+ * to detach track from effect.
+ */
+ int attachAuxEffect(int effectId);
+
+ /** Send parameters to the audio hardware. */
+ int setParameters(@utf8InCpp String keyValuePairs);
+
+ /** Selects the presentation (if available). */
+ int selectPresentation(int presentationId, int programId);
+
+ /** Return NO_ERROR if timestamp is valid. */
+ int getTimestamp(out AudioTimestampInternal timestamp);
+
+ /** Signal the playback thread for a change in control block. */
+ void signal();
+
+ /** Sets the volume shaper. Returns the volume shaper status. */
+ int applyVolumeShaper(in VolumeShaperConfiguration configuration,
+ in VolumeShaperOperation operation);
+
+ /** Gets the volume shaper state. */
+ @nullable VolumeShaperState getVolumeShaperState(int id);
+}
diff --git a/media/libaudioclient/aidl/android/media/IAudioTrackCallback.aidl b/media/libaudioclient/aidl/android/media/IAudioTrackCallback.aidl
index 21553b5..f593e22 100644
--- a/media/libaudioclient/aidl/android/media/IAudioTrackCallback.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioTrackCallback.aidl
@@ -17,7 +17,7 @@
package android.media;
/**
- * @hide
+ * {@hide}
*/
interface IAudioTrackCallback {
oneway void onCodecFormatChanged(in byte[] audioMetadata);
diff --git a/media/libaudioclient/aidl/android/media/ICaptureStateListener.aidl b/media/libaudioclient/aidl/android/media/ICaptureStateListener.aidl
index 8502282..3b2206a 100644
--- a/media/libaudioclient/aidl/android/media/ICaptureStateListener.aidl
+++ b/media/libaudioclient/aidl/android/media/ICaptureStateListener.aidl
@@ -16,6 +16,9 @@
package android.media;
+/**
+ * {@hide}
+ */
interface ICaptureStateListener {
void setCaptureState(boolean active);
}
diff --git a/media/libaudioclient/aidl/android/media/IEffect.aidl b/media/libaudioclient/aidl/android/media/IEffect.aidl
index 9548e46..813cd5c 100644
--- a/media/libaudioclient/aidl/android/media/IEffect.aidl
+++ b/media/libaudioclient/aidl/android/media/IEffect.aidl
@@ -21,7 +21,7 @@
/**
* The IEffect interface enables control of the effect module activity and parameters.
*
- * @hide
+ * {@hide}
*/
interface IEffect {
/**
diff --git a/media/libaudioclient/aidl/android/media/IEffectClient.aidl b/media/libaudioclient/aidl/android/media/IEffectClient.aidl
index d1e331c..3b6bcf1 100644
--- a/media/libaudioclient/aidl/android/media/IEffectClient.aidl
+++ b/media/libaudioclient/aidl/android/media/IEffectClient.aidl
@@ -19,7 +19,7 @@
/**
* A callback interface for getting effect-related notifications.
*
- * @hide
+ * {@hide}
*/
interface IEffectClient {
/**
diff --git a/media/libaudioclient/aidl/android/media/IPlayer.aidl b/media/libaudioclient/aidl/android/media/IPlayer.aidl
index 8c2c471..43bb7f3 100644
--- a/media/libaudioclient/aidl/android/media/IPlayer.aidl
+++ b/media/libaudioclient/aidl/android/media/IPlayer.aidl
@@ -20,7 +20,7 @@
import android.media.VolumeShaperOperation;
/**
- * @hide
+ * {@hide}
*/
interface IPlayer {
oneway void start();
diff --git a/media/libaudioclient/aidl/android/media/OpenInputRequest.aidl b/media/libaudioclient/aidl/android/media/OpenInputRequest.aidl
new file mode 100644
index 0000000..2e55526
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/OpenInputRequest.aidl
@@ -0,0 +1,36 @@
+/*
+ * Copyright (C) 2020 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.
+ */
+
+package android.media;
+
+import android.media.AudioConfig;
+import android.media.AudioDevice;
+import android.media.AudioSourceType;
+
+/**
+ * {@hide}
+ */
+parcelable OpenInputRequest {
+ /** Interpreted as audio_module_handle_t. */
+ int module;
+ /** Interpreted as audio_io_handle_t. */
+ int input;
+ AudioConfig config;
+ AudioDevice device;
+ AudioSourceType source;
+ /** Bitmask, indexed by AudioInputFlag. */
+ int flags;
+}
diff --git a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl b/media/libaudioclient/aidl/android/media/OpenInputResponse.aidl
similarity index 67%
copy from media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
copy to media/libaudioclient/aidl/android/media/OpenInputResponse.aidl
index d6e46cb..b613ba5 100644
--- a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
+++ b/media/libaudioclient/aidl/android/media/OpenInputResponse.aidl
@@ -1,5 +1,5 @@
/*
- * Copyright 2018 The Android Open Source Project
+ * Copyright (C) 2020 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.
@@ -16,4 +16,15 @@
package android.media;
-parcelable MicrophoneInfo cpp_header "media/MicrophoneInfo.h";
+import android.media.AudioConfig;
+import android.media.AudioDevice;
+
+/**
+ * {@hide}
+ */
+parcelable OpenInputResponse {
+ /** Interpreted as audio_io_handle_t. */
+ int input;
+ AudioConfig config;
+ AudioDevice device;
+}
diff --git a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
similarity index 61%
copy from media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
copy to media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
index d6e46cb..4518adb 100644
--- a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
+++ b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
@@ -1,5 +1,5 @@
/*
- * Copyright 2018 The Android Open Source Project
+ * Copyright (C) 2020 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.
@@ -16,4 +16,17 @@
package android.media;
-parcelable MicrophoneInfo cpp_header "media/MicrophoneInfo.h";
+import android.media.AudioConfig;
+import android.media.DeviceDescriptorBase;
+
+/**
+ * {@hide}
+ */
+parcelable OpenOutputRequest {
+ /** Interpreted as audio_module_handle_t. */
+ int module;
+ AudioConfig config;
+ DeviceDescriptorBase device;
+ /** Bitmask, indexed by AudioOutputFlag. */
+ int flags;
+}
diff --git a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl b/media/libaudioclient/aidl/android/media/OpenOutputResponse.aidl
similarity index 65%
copy from media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
copy to media/libaudioclient/aidl/android/media/OpenOutputResponse.aidl
index d6e46cb..a051969 100644
--- a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
+++ b/media/libaudioclient/aidl/android/media/OpenOutputResponse.aidl
@@ -1,5 +1,5 @@
/*
- * Copyright 2018 The Android Open Source Project
+ * Copyright (C) 2020 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.
@@ -16,4 +16,16 @@
package android.media;
-parcelable MicrophoneInfo cpp_header "media/MicrophoneInfo.h";
+import android.media.AudioConfig;
+
+/**
+ * {@hide}
+ */
+parcelable OpenOutputResponse {
+ /** Interpreted as audio_io_handle_t. */
+ int output;
+ AudioConfig config;
+ int latencyMs;
+ /** Bitmask, indexed by AudioOutputFlag. */
+ int flags;
+}
diff --git a/media/libaudioclient/include/media/AidlConversion.h b/media/libaudioclient/include/media/AidlConversion.h
index 4df8083..894e56e 100644
--- a/media/libaudioclient/include/media/AidlConversion.h
+++ b/media/libaudioclient/include/media/AidlConversion.h
@@ -21,12 +21,12 @@
#include <system/audio.h>
-#include <android-base/expected.h>
-
#include <android/media/AudioAttributesInternal.h>
#include <android/media/AudioClient.h>
#include <android/media/AudioConfig.h>
#include <android/media/AudioConfigBase.h>
+#include <android/media/AudioEncapsulationMode.h>
+#include <android/media/AudioEncapsulationMetadataType.h>
#include <android/media/AudioFlag.h>
#include <android/media/AudioGainMode.h>
#include <android/media/AudioInputFlags.h>
@@ -34,55 +34,19 @@
#include <android/media/AudioIoDescriptor.h>
#include <android/media/AudioOutputFlags.h>
#include <android/media/AudioPortConfigType.h>
+#include <android/media/AudioTimestampInternal.h>
+#include <android/media/EffectDescriptor.h>
#include <android/media/SharedFileRegion.h>
-
#include <binder/IMemory.h>
+#include <media/AidlConversionUtil.h>
#include <media/AudioClient.h>
#include <media/AudioIoDescriptor.h>
+#include <media/AudioTimestamp.h>
+#include <system/audio_effect.h>
namespace android {
-template <typename T>
-using ConversionResult = base::expected<T, status_t>;
-
-// Convenience macros for working with ConversionResult, useful for writing converted for aggregate
-// types.
-
-#define VALUE_OR_RETURN(result) \
- ({ \
- auto _tmp = (result); \
- if (!_tmp.ok()) return base::unexpected(_tmp.error()); \
- std::move(_tmp.value()); \
- })
-
-#define RETURN_IF_ERROR(result) \
- if (status_t _tmp = (result); _tmp != OK) return base::unexpected(_tmp);
-
-/**
- * A generic template to safely cast between integral types, respecting limits of the destination
- * type.
- */
-template<typename To, typename From>
-ConversionResult<To> convertIntegral(From from) {
- // Special handling is required for signed / vs. unsigned comparisons, since otherwise we may
- // have the signed converted to unsigned and produce wrong results.
- if (std::is_signed_v<From> && !std::is_signed_v<To>) {
- if (from < 0 || from > std::numeric_limits<To>::max()) {
- return base::unexpected(BAD_VALUE);
- }
- } else if (std::is_signed_v<To> && !std::is_signed_v<From>) {
- if (from > std::numeric_limits<To>::max()) {
- return base::unexpected(BAD_VALUE);
- }
- } else {
- if (from < std::numeric_limits<To>::min() || from > std::numeric_limits<To>::max()) {
- return base::unexpected(BAD_VALUE);
- }
- }
- return static_cast<To>(from);
-}
-
// maxSize is the size of the C-string buffer (including the 0-terminator), NOT the max length of
// the string.
status_t aidl2legacy_string(std::string_view aidl, char* dest, size_t maxSize);
@@ -120,6 +84,9 @@
ConversionResult<uid_t> aidl2legacy_int32_t_uid_t(int32_t aidl);
ConversionResult<int32_t> legacy2aidl_uid_t_int32_t(uid_t legacy);
+ConversionResult<String8> aidl2legacy_string_view_String8(std::string_view aidl);
+ConversionResult<std::string> legacy2aidl_String8_string(const String8& legacy);
+
ConversionResult<String16> aidl2legacy_string_view_String16(std::string_view aidl);
ConversionResult<std::string> legacy2aidl_String16_string(const String16& legacy);
@@ -251,9 +218,9 @@
legacy2aidl_audio_attributes_t_AudioAttributesInternal(const audio_attributes_t& legacy);
ConversionResult<audio_encapsulation_mode_t>
-aidl2legacy_audio_encapsulation_mode_t_AudioEncapsulationMode(media::AudioEncapsulationMode aidl);
+aidl2legacy_AudioEncapsulationMode_audio_encapsulation_mode_t(media::AudioEncapsulationMode aidl);
ConversionResult<media::AudioEncapsulationMode>
-legacy2aidl_AudioEncapsulationMode_audio_encapsulation_mode_t(audio_encapsulation_mode_t legacy);
+legacy2aidl_audio_encapsulation_mode_t_AudioEncapsulationMode(audio_encapsulation_mode_t legacy);
ConversionResult<audio_offload_info_t>
aidl2legacy_AudioOffloadInfo_audio_offload_info_t(const media::AudioOffloadInfo& aidl);
@@ -280,4 +247,36 @@
ConversionResult<std::optional<media::SharedFileRegion>>
legacy2aidl_NullableIMemory_SharedFileRegion(const sp<IMemory>& legacy);
+ConversionResult<AudioTimestamp>
+aidl2legacy_AudioTimestamp(const media::AudioTimestampInternal& aidl);
+ConversionResult<media::AudioTimestampInternal>
+legacy2aidl_AudioTimestamp(const AudioTimestamp& legacy);
+
+ConversionResult<audio_uuid_t>
+aidl2legacy_AudioUuid_audio_uuid_t(const media::AudioUuid& aidl);
+ConversionResult<media::AudioUuid>
+legacy2aidl_audio_uuid_t_AudioUuid(const audio_uuid_t& legacy);
+
+ConversionResult<effect_descriptor_t>
+aidl2legacy_EffectDescriptor_effect_descriptor_t(const media::EffectDescriptor& aidl);
+ConversionResult<media::EffectDescriptor>
+legacy2aidl_effect_descriptor_t_EffectDescriptor(const effect_descriptor_t& legacy);
+
+ConversionResult<audio_encapsulation_metadata_type_t>
+aidl2legacy_AudioEncapsulationMetadataType_audio_encapsulation_metadata_type_t(
+ media::AudioEncapsulationMetadataType aidl);
+ConversionResult<media::AudioEncapsulationMetadataType>
+legacy2aidl_audio_encapsulation_metadata_type_t_AudioEncapsulationMetadataType(
+ audio_encapsulation_metadata_type_t legacy);
+
+ConversionResult<uint32_t>
+aidl2legacy_AudioEncapsulationMode_mask(int32_t aidl);
+ConversionResult<int32_t>
+legacy2aidl_AudioEncapsulationMode_mask(uint32_t legacy);
+
+ConversionResult<uint32_t>
+aidl2legacy_AudioEncapsulationMetadataType_mask(int32_t aidl);
+ConversionResult<int32_t>
+legacy2aidl_AudioEncapsulationMetadataType_mask(uint32_t legacy);
+
} // namespace android
diff --git a/media/libaudioclient/include/media/AidlConversionUtil.h b/media/libaudioclient/include/media/AidlConversionUtil.h
new file mode 100644
index 0000000..00e5ff2
--- /dev/null
+++ b/media/libaudioclient/include/media/AidlConversionUtil.h
@@ -0,0 +1,98 @@
+/*
+ * Copyright (C) 2020 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.
+ */
+
+#pragma once
+
+#include <limits>
+#include <type_traits>
+#include <utility>
+
+#include <android-base/expected.h>
+
+namespace android {
+
+template <typename T>
+using ConversionResult = base::expected<T, status_t>;
+
+// Convenience macros for working with ConversionResult, useful for writing converted for aggregate
+// types.
+
+#define VALUE_OR_RETURN(result) \
+ ({ \
+ auto _tmp = (result); \
+ if (!_tmp.ok()) return base::unexpected(_tmp.error()); \
+ std::move(_tmp.value()); \
+ })
+
+#define RETURN_IF_ERROR(result) \
+ if (status_t _tmp = (result); _tmp != OK) return base::unexpected(_tmp);
+
+#define VALUE_OR_RETURN_STATUS(x) \
+ ({ \
+ auto _tmp = (x); \
+ if (!_tmp.ok()) return _tmp.error(); \
+ std::move(_tmp.value()); \
+ })
+
+/**
+ * A generic template to safely cast between integral types, respecting limits of the destination
+ * type.
+ */
+template<typename To, typename From>
+ConversionResult<To> convertIntegral(From from) {
+ // Special handling is required for signed / vs. unsigned comparisons, since otherwise we may
+ // have the signed converted to unsigned and produce wrong results.
+ if (std::is_signed_v<From> && !std::is_signed_v<To>) {
+ if (from < 0 || from > std::numeric_limits<To>::max()) {
+ return base::unexpected(BAD_VALUE);
+ }
+ } else if (std::is_signed_v<To> && !std::is_signed_v<From>) {
+ if (from > std::numeric_limits<To>::max()) {
+ return base::unexpected(BAD_VALUE);
+ }
+ } else {
+ if (from < std::numeric_limits<To>::min() || from > std::numeric_limits<To>::max()) {
+ return base::unexpected(BAD_VALUE);
+ }
+ }
+ return static_cast<To>(from);
+}
+
+/**
+ * A generic template to safely cast between types, that are intended to be the same size, but
+ * interpreted differently.
+ */
+template<typename To, typename From>
+ConversionResult<To> convertReinterpret(From from) {
+ static_assert(sizeof(From) == sizeof(To));
+ return static_cast<To>(from);
+}
+
+/**
+ * A generic template that helps convert containers of convertible types.
+ */
+template<typename OutputContainer, typename InputContainer, typename Func>
+ConversionResult<OutputContainer>
+convertContainer(const InputContainer& input, const Func& itemConversion) {
+ OutputContainer output;
+ auto ins = std::inserter(output, output.begin());
+ for (const auto& item : input) {
+ *ins = VALUE_OR_RETURN(itemConversion(item));
+ }
+ return output;
+}
+
+} // namespace android
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index de183d8..3728a16 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -17,18 +17,20 @@
#ifndef ANDROID_AUDIOTRACK_H
#define ANDROID_AUDIOTRACK_H
+#include <binder/IMemory.h>
#include <cutils/sched_policy.h>
#include <media/AudioSystem.h>
#include <media/AudioTimestamp.h>
-#include <media/IAudioTrack.h>
#include <media/AudioResamplerPublic.h>
#include <media/MediaMetricsItem.h>
#include <media/Modulo.h>
+#include <media/VolumeShaper.h>
#include <utils/threads.h>
#include <string>
#include "android/media/BnAudioTrackCallback.h"
+#include "android/media/IAudioTrack.h"
#include "android/media/IAudioTrackCallback.h"
namespace android {
@@ -1071,7 +1073,7 @@
void updateRoutedDeviceId_l();
// Next 4 fields may be changed if IAudioTrack is re-created, but always != 0
- sp<IAudioTrack> mAudioTrack;
+ sp<media::IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
audio_io_handle_t mOutput = AUDIO_IO_HANDLE_NONE; // from AudioSystem::getOutputForAttr()
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index db7e4f6..11d341e 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -29,7 +29,6 @@
#include <media/AidlConversion.h>
#include <media/AudioClient.h>
#include <media/DeviceDescriptorBase.h>
-#include <media/IAudioTrack.h>
#include <system/audio.h>
#include <system/audio_effect.h>
#include <system/audio_policy.h>
@@ -38,15 +37,22 @@
#include <string>
#include <vector>
+#include "android/media/CreateEffectRequest.h"
+#include "android/media/CreateEffectResponse.h"
#include "android/media/CreateRecordRequest.h"
#include "android/media/CreateRecordResponse.h"
#include "android/media/CreateTrackRequest.h"
#include "android/media/CreateTrackResponse.h"
#include "android/media/IAudioRecord.h"
#include "android/media/IAudioFlingerClient.h"
+#include "android/media/IAudioTrack.h"
#include "android/media/IAudioTrackCallback.h"
#include "android/media/IEffect.h"
#include "android/media/IEffectClient.h"
+#include "android/media/OpenInputRequest.h"
+#include "android/media/OpenInputResponse.h"
+#include "android/media/OpenOutputRequest.h"
+#include "android/media/OpenOutputResponse.h"
namespace android {
@@ -104,6 +110,7 @@
uint32_t afLatencyMs;
audio_io_handle_t outputId;
audio_port_handle_t portId;
+ sp<media::IAudioTrack> audioTrack;
ConversionResult<media::CreateTrackResponse> toAidl() const;
static ConversionResult<CreateTrackOutput> fromAidl(const media::CreateTrackResponse& aidl);
@@ -152,24 +159,25 @@
sp<IMemory> cblk;
sp<IMemory> buffers;
audio_port_handle_t portId;
+ sp<media::IAudioRecord> audioRecord;
ConversionResult<media::CreateRecordResponse> toAidl() const;
static ConversionResult<CreateRecordOutput> fromAidl(const media::CreateRecordResponse& aidl);
};
- // invariant on exit for all APIs that return an sp<>:
- // (return value != 0) == (*status == NO_ERROR)
-
/* create an audio track and registers it with AudioFlinger.
- * return null if the track cannot be created.
+ * The audioTrack field will be null if the track cannot be created and the status will reflect
+ * failure.
*/
- virtual sp<IAudioTrack> createTrack(const media::CreateTrackRequest& input,
- media::CreateTrackResponse& output,
- status_t* status) = 0;
+ virtual status_t createTrack(const media::CreateTrackRequest& input,
+ media::CreateTrackResponse& output) = 0;
- virtual sp<media::IAudioRecord> createRecord(const media::CreateRecordRequest& input,
- media::CreateRecordResponse& output,
- status_t* status) = 0;
+ /* create an audio record and registers it with AudioFlinger.
+ * The audioRecord field will be null if the track cannot be created and the status will reflect
+ * failure.
+ */
+ virtual status_t createRecord(const media::CreateRecordRequest& input,
+ media::CreateRecordResponse& output) = 0;
// FIXME Surprisingly, format/latency don't work for input handles
@@ -232,25 +240,17 @@
virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
audio_channel_mask_t channelMask) const = 0;
- virtual status_t openOutput(audio_module_handle_t module,
- audio_io_handle_t *output,
- audio_config_t *config,
- const sp<DeviceDescriptorBase>& device,
- uint32_t *latencyMs,
- audio_output_flags_t flags) = 0;
+ virtual status_t openOutput(const media::OpenOutputRequest& request,
+ media::OpenOutputResponse* response) = 0;
virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
audio_io_handle_t output2) = 0;
virtual status_t closeOutput(audio_io_handle_t output) = 0;
virtual status_t suspendOutput(audio_io_handle_t output) = 0;
virtual status_t restoreOutput(audio_io_handle_t output) = 0;
- virtual status_t openInput(audio_module_handle_t module,
- audio_io_handle_t *input,
- audio_config_t *config,
- audio_devices_t *device,
- const String8& address,
- audio_source_t source,
- audio_input_flags_t flags) = 0;
+ virtual status_t openInput(const media::OpenInputRequest& request,
+ media::OpenInputResponse* response) = 0;
+
virtual status_t closeInput(audio_io_handle_t input) = 0;
virtual status_t invalidateStream(audio_stream_type_t stream) = 0;
@@ -276,20 +276,8 @@
uint32_t preferredTypeFlag,
effect_descriptor_t *pDescriptor) const = 0;
- virtual sp<media::IEffect> createEffect(
- effect_descriptor_t *pDesc,
- const sp<media::IEffectClient>& client,
- int32_t priority,
- // AudioFlinger doesn't take over handle reference from client
- audio_io_handle_t output,
- audio_session_t sessionId,
- const AudioDeviceTypeAddr& device,
- const String16& callingPackage,
- pid_t pid,
- bool probe,
- status_t *status,
- int *id,
- int *enabled) = 0;
+ virtual status_t createEffect(const media::CreateEffectRequest& request,
+ media::CreateEffectResponse* response) = 0;
virtual status_t moveEffects(audio_session_t session, audio_io_handle_t srcOutput,
audio_io_handle_t dstOutput) = 0;
diff --git a/media/libaudioclient/include/media/IAudioTrack.h b/media/libaudioclient/include/media/IAudioTrack.h
deleted file mode 100644
index 06e786d..0000000
--- a/media/libaudioclient/include/media/IAudioTrack.h
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- * Copyright (C) 2007 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_IAUDIOTRACK_H
-#define ANDROID_IAUDIOTRACK_H
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <utils/RefBase.h>
-#include <utils/Errors.h>
-#include <binder/IInterface.h>
-#include <binder/IMemory.h>
-#include <utils/String8.h>
-#include <media/AudioTimestamp.h>
-#include <media/VolumeShaper.h>
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-class IAudioTrack : public IInterface
-{
-public:
- DECLARE_META_INTERFACE(AudioTrack);
-
- /* Get this track's control block */
- virtual sp<IMemory> getCblk() const = 0;
-
- /* After it's created the track is not active. Call start() to
- * make it active.
- */
- virtual status_t start() = 0;
-
- /* Stop a track. If set, the callback will cease being called and
- * obtainBuffer will return an error. Buffers that are already released
- * will continue to be processed, unless/until flush() is called.
- */
- virtual void stop() = 0;
-
- /* Flush a stopped or paused track. All pending/released buffers are discarded.
- * This function has no effect if the track is not stopped or paused.
- */
- virtual void flush() = 0;
-
- /* Pause a track. If set, the callback will cease being called and
- * obtainBuffer will return an error. Buffers that are already released
- * will continue to be processed, unless/until flush() is called.
- */
- virtual void pause() = 0;
-
- /* Attach track auxiliary output to specified effect. Use effectId = 0
- * to detach track from effect.
- */
- virtual status_t attachAuxEffect(int effectId) = 0;
-
- /* Send parameters to the audio hardware */
- virtual status_t setParameters(const String8& keyValuePairs) = 0;
-
- /* Selects the presentation (if available) */
- virtual status_t selectPresentation(int presentationId, int programId) = 0;
-
- /* Return NO_ERROR if timestamp is valid. timestamp is undefined otherwise. */
- virtual status_t getTimestamp(AudioTimestamp& timestamp) = 0;
-
- /* Signal the playback thread for a change in control block */
- virtual void signal() = 0;
-
- /* Sets the volume shaper */
- virtual media::VolumeShaper::Status applyVolumeShaper(
- const sp<media::VolumeShaper::Configuration>& configuration,
- const sp<media::VolumeShaper::Operation>& operation) = 0;
-
- /* gets the volume shaper state */
- virtual sp<media::VolumeShaper::State> getVolumeShaperState(int id) = 0;
-};
-
-// ----------------------------------------------------------------------------
-
-class BnAudioTrack : public BnInterface<IAudioTrack>
-{
-public:
- virtual status_t onTransact( uint32_t code,
- const Parcel& data,
- Parcel* reply,
- uint32_t flags = 0);
-};
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
-
-#endif // ANDROID_IAUDIOTRACK_H
diff --git a/media/libaudiofoundation/Android.bp b/media/libaudiofoundation/Android.bp
index a8e6c31..9296d0e 100644
--- a/media/libaudiofoundation/Android.bp
+++ b/media/libaudiofoundation/Android.bp
@@ -5,13 +5,21 @@
export_include_dirs: ["include"],
header_libs: [
+ "libaudioclient_aidl_conversion_util",
"libaudio_system_headers",
"libmedia_helper_headers",
],
export_header_lib_headers: [
+ "libaudioclient_aidl_conversion_util",
"libaudio_system_headers",
"libmedia_helper_headers",
],
+ static_libs: [
+ "audioclient-types-aidl-unstable-cpp",
+ ],
+ export_static_lib_headers: [
+ "audioclient-types-aidl-unstable-cpp",
+ ],
host_supported: true,
target: {
darwin: {
@@ -35,6 +43,8 @@
],
shared_libs: [
+ "audioclient-types-aidl-unstable-cpp",
+ "libaudioclient_aidl_conversion",
"libaudioutils",
"libbase",
"libbinder",
@@ -43,6 +53,11 @@
"libutils",
],
+ export_shared_lib_headers: [
+ "audioclient-types-aidl-unstable-cpp",
+ "libaudioclient_aidl_conversion",
+ ],
+
header_libs: [
"libaudiofoundation_headers",
],
diff --git a/media/libaudiofoundation/AudioDeviceTypeAddr.cpp b/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
index a47337b..8f1e113 100644
--- a/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
+++ b/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
@@ -155,4 +155,18 @@
return stream.str();
}
+ConversionResult<AudioDeviceTypeAddr>
+aidl2legacy_AudioDeviceTypeAddress(const media::AudioDevice& aidl) {
+ audio_devices_t type = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_devices_t(aidl.type));
+ return AudioDeviceTypeAddr(type, aidl.address);
+}
+
+ConversionResult<media::AudioDevice>
+legacy2aidl_AudioDeviceTypeAddress(const AudioDeviceTypeAddr& legacy) {
+ media::AudioDevice aidl;
+ aidl.type = VALUE_OR_RETURN(legacy2aidl_audio_devices_t_int32_t(legacy.mType));
+ aidl.address = legacy.getAddress();
+ return aidl;
+}
+
} // namespace android
diff --git a/media/libaudiofoundation/AudioGain.cpp b/media/libaudiofoundation/AudioGain.cpp
index 759140e..c59e966 100644
--- a/media/libaudiofoundation/AudioGain.cpp
+++ b/media/libaudiofoundation/AudioGain.cpp
@@ -129,42 +129,49 @@
mGain.max_ramp_ms == other->mGain.max_ramp_ms;
}
-status_t AudioGain::writeToParcel(android::Parcel *parcel) const
-{
- status_t status = NO_ERROR;
- if ((status = parcel->writeInt32(mIndex)) != NO_ERROR) return status;
- if ((status = parcel->writeBool(mUseInChannelMask)) != NO_ERROR) return status;
- if ((status = parcel->writeBool(mUseForVolume)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mGain.mode)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mGain.channel_mask)) != NO_ERROR) return status;
- if ((status = parcel->writeInt32(mGain.min_value)) != NO_ERROR) return status;
- if ((status = parcel->writeInt32(mGain.max_value)) != NO_ERROR) return status;
- if ((status = parcel->writeInt32(mGain.default_value)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mGain.step_value)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mGain.min_ramp_ms)) != NO_ERROR) return status;
- status = parcel->writeUint32(mGain.max_ramp_ms);
- return status;
+status_t AudioGain::writeToParcel(android::Parcel *parcel) const {
+ media::AudioGain parcelable;
+ return writeToParcelable(&parcelable)
+ ?: parcelable.writeToParcel(parcel);
}
-status_t AudioGain::readFromParcel(const android::Parcel *parcel)
-{
- status_t status = NO_ERROR;
- if ((status = parcel->readInt32(&mIndex)) != NO_ERROR) return status;
- if ((status = parcel->readBool(&mUseInChannelMask)) != NO_ERROR) return status;
- if ((status = parcel->readBool(&mUseForVolume)) != NO_ERROR) return status;
- uint32_t rawGainMode;
- if ((status = parcel->readUint32(&rawGainMode)) != NO_ERROR) return status;
- mGain.mode = static_cast<audio_gain_mode_t>(rawGainMode);
- uint32_t rawChannelMask;
- if ((status = parcel->readUint32(&rawChannelMask)) != NO_ERROR) return status;
- mGain.channel_mask = static_cast<audio_channel_mask_t>(rawChannelMask);
- if ((status = parcel->readInt32(&mGain.min_value)) != NO_ERROR) return status;
- if ((status = parcel->readInt32(&mGain.max_value)) != NO_ERROR) return status;
- if ((status = parcel->readInt32(&mGain.default_value)) != NO_ERROR) return status;
- if ((status = parcel->readUint32(&mGain.step_value)) != NO_ERROR) return status;
- if ((status = parcel->readUint32(&mGain.min_ramp_ms)) != NO_ERROR) return status;
- status = parcel->readUint32(&mGain.max_ramp_ms);
- return status;
+status_t AudioGain::writeToParcelable(media::AudioGain* parcelable) const {
+ parcelable->index = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mIndex));
+ parcelable->useInChannelMask = mUseInChannelMask;
+ parcelable->useForVolume = mUseForVolume;
+ parcelable->mode = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_gain_mode_t_int32_t(mGain.mode));
+ parcelable->channelMask = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_channel_mask_t_int32_t(mGain.channel_mask));
+ parcelable->minValue = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.min_value));
+ parcelable->maxValue = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.max_value));
+ parcelable->defaultValue = VALUE_OR_RETURN_STATUS(
+ convertIntegral<int32_t>(mGain.default_value));
+ parcelable->stepValue = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.step_value));
+ parcelable->minRampMs = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.min_ramp_ms));
+ parcelable->maxRampMs = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.max_ramp_ms));
+ return OK;
+}
+
+status_t AudioGain::readFromParcel(const android::Parcel *parcel) {
+ media::AudioGain parcelable;
+ return parcelable.readFromParcel(parcel)
+ ?: readFromParcelable(parcelable);
+}
+
+status_t AudioGain::readFromParcelable(const media::AudioGain& parcelable) {
+ mIndex = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.index));
+ mUseInChannelMask = parcelable.useInChannelMask;
+ mUseForVolume = parcelable.useForVolume;
+ mGain.mode = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_gain_mode_t(parcelable.mode));
+ mGain.channel_mask = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_int32_t_audio_channel_mask_t(parcelable.channelMask));
+ mGain.min_value = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.minValue));
+ mGain.max_value = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.maxValue));
+ mGain.default_value = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.defaultValue));
+ mGain.step_value = VALUE_OR_RETURN_STATUS(convertIntegral<unsigned int>(parcelable.stepValue));
+ mGain.min_ramp_ms = VALUE_OR_RETURN_STATUS(convertIntegral<unsigned int>(parcelable.minRampMs));
+ mGain.max_ramp_ms = VALUE_OR_RETURN_STATUS(convertIntegral<unsigned int>(parcelable.maxRampMs));
+ return OK;
}
bool AudioGains::equals(const AudioGains &other) const
@@ -200,4 +207,34 @@
return status;
}
+ConversionResult<sp<AudioGain>>
+aidl2legacy_AudioGain(const media::AudioGain& aidl) {
+ sp<AudioGain> legacy = new AudioGain(0, false);
+ status_t status = legacy->readFromParcelable(aidl);
+ if (status != OK) {
+ return base::unexpected(status);
+ }
+ return legacy;
+}
+
+ConversionResult<media::AudioGain>
+legacy2aidl_AudioGain(const sp<AudioGain>& legacy) {
+ media::AudioGain aidl;
+ status_t status = legacy->writeToParcelable(&aidl);
+ if (status != OK) {
+ return base::unexpected(status);
+ }
+ return aidl;
+}
+
+ConversionResult<AudioGains>
+aidl2legacy_AudioGains(const std::vector<media::AudioGain>& aidl) {
+ return convertContainer<AudioGains>(aidl, aidl2legacy_AudioGain);
+}
+
+ConversionResult<std::vector<media::AudioGain>>
+legacy2aidl_AudioGains(const AudioGains& legacy) {
+ return convertContainer<std::vector<media::AudioGain>>(legacy, legacy2aidl_AudioGain);
+}
+
} // namespace android
diff --git a/media/libaudiofoundation/AudioPort.cpp b/media/libaudiofoundation/AudioPort.cpp
index d650c67..559c711 100644
--- a/media/libaudiofoundation/AudioPort.cpp
+++ b/media/libaudiofoundation/AudioPort.cpp
@@ -150,32 +150,33 @@
status_t AudioPort::writeToParcel(Parcel *parcel) const
{
- status_t status = NO_ERROR;
- if ((status = parcel->writeUtf8AsUtf16(mName)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mType)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mRole)) != NO_ERROR) return status;
- if ((status = parcel->writeParcelable(mProfiles)) != NO_ERROR) return status;
- if ((status = parcel->writeParcelable(mGains)) != NO_ERROR) return status;
- return status;
+ media::AudioPort parcelable;
+ return writeToParcelable(&parcelable)
+ ?: parcelable.writeToParcel(parcel);
}
-status_t AudioPort::readFromParcel(const Parcel *parcel)
-{
- status_t status = NO_ERROR;
- if ((status = parcel->readUtf8FromUtf16(&mName)) != NO_ERROR) return status;
- static_assert(sizeof(mType) == sizeof(uint32_t));
- if ((status = parcel->readUint32(reinterpret_cast<uint32_t*>(&mType))) != NO_ERROR) {
- return status;
- }
- static_assert(sizeof(mRole) == sizeof(uint32_t));
- if ((status = parcel->readUint32(reinterpret_cast<uint32_t*>(&mRole))) != NO_ERROR) {
- return status;
- }
- mProfiles.clear();
- if ((status = parcel->readParcelable(&mProfiles)) != NO_ERROR) return status;
- mGains.clear();
- if ((status = parcel->readParcelable(&mGains)) != NO_ERROR) return status;
- return status;
+status_t AudioPort::writeToParcelable(media::AudioPort* parcelable) const {
+ parcelable->name = mName;
+ parcelable->type = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_type_t_AudioPortType(mType));
+ parcelable->role = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_role_t_AudioPortRole(mRole));
+ parcelable->profiles = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioProfileVector(mProfiles));
+ parcelable->gains = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioGains(mGains));
+ return OK;
+}
+
+status_t AudioPort::readFromParcel(const Parcel *parcel) {
+ media::AudioPort parcelable;
+ return parcelable.readFromParcel(parcel)
+ ?: readFromParcelable(parcelable);
+}
+
+status_t AudioPort::readFromParcelable(const media::AudioPort& parcelable) {
+ mName = parcelable.name;
+ mType = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioPortType_audio_port_type_t(parcelable.type));
+ mRole = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioPortRole_audio_port_role_t(parcelable.role));
+ mProfiles = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioProfileVector(parcelable.profiles));
+ mGains = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioGains(parcelable.gains));
+ return OK;
}
// --- AudioPortConfig class implementation
@@ -276,50 +277,56 @@
mGain.ramp_duration_ms == other->mGain.ramp_duration_ms;
}
-status_t AudioPortConfig::writeToParcel(Parcel *parcel) const
-{
- status_t status = NO_ERROR;
- if ((status = parcel->writeUint32(mSamplingRate)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mFormat)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mChannelMask)) != NO_ERROR) return status;
- if ((status = parcel->writeInt32(mId)) != NO_ERROR) return status;
- // Write mGain to parcel.
- if ((status = parcel->writeInt32(mGain.index)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mGain.mode)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mGain.channel_mask)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mGain.ramp_duration_ms)) != NO_ERROR) return status;
- std::vector<int> values(std::begin(mGain.values), std::end(mGain.values));
- if ((status = parcel->writeInt32Vector(values)) != NO_ERROR) return status;
- return status;
+status_t AudioPortConfig::writeToParcel(Parcel *parcel) const {
+ media::AudioPortConfig parcelable;
+ return writeToParcelable(&parcelable)
+ ?: parcelable.writeToParcel(parcel);
}
-status_t AudioPortConfig::readFromParcel(const Parcel *parcel)
-{
- status_t status = NO_ERROR;
- if ((status = parcel->readUint32(&mSamplingRate)) != NO_ERROR) return status;
- static_assert(sizeof(mFormat) == sizeof(uint32_t));
- if ((status = parcel->readUint32(reinterpret_cast<uint32_t*>(&mFormat))) != NO_ERROR) {
- return status;
- }
- uint32_t rawChannelMask;
- if ((status = parcel->readUint32(&rawChannelMask)) != NO_ERROR) return status;
- mChannelMask = static_cast<audio_channel_mask_t>(rawChannelMask);
- if ((status = parcel->readInt32(&mId)) != NO_ERROR) return status;
- // Read mGain from parcel.
- if ((status = parcel->readInt32(&mGain.index)) != NO_ERROR) return status;
- uint32_t rawGainMode;
- if ((status = parcel->readUint32(&rawGainMode)) != NO_ERROR) return status;
- mGain.mode = static_cast<audio_gain_mode_t>(rawGainMode);
- if ((status = parcel->readUint32(&rawChannelMask)) != NO_ERROR) return status;
- mGain.channel_mask = static_cast<audio_channel_mask_t>(rawChannelMask);
- if ((status = parcel->readUint32(&mGain.ramp_duration_ms)) != NO_ERROR) return status;
- std::vector<int> values;
- if ((status = parcel->readInt32Vector(&values)) != NO_ERROR) return status;
- if (values.size() != std::size(mGain.values)) {
+status_t AudioPortConfig::writeToParcelable(media::AudioPortConfig* parcelable) const {
+ parcelable->sampleRate = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mSamplingRate));
+ parcelable->format = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_format_t_AudioFormat(mFormat));
+ parcelable->channelMask = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_channel_mask_t_int32_t(mChannelMask));
+ parcelable->id = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_handle_t_int32_t(mId));
+ parcelable->gain.index = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.index));
+ parcelable->gain.mode = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_gain_mode_t_int32_t(mGain.mode));
+ parcelable->gain.channelMask = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_channel_mask_t_int32_t(mGain.channel_mask));
+ parcelable->gain.rampDurationMs = VALUE_OR_RETURN_STATUS(
+ convertIntegral<int32_t>(mGain.ramp_duration_ms));
+ parcelable->gain.values = VALUE_OR_RETURN_STATUS(convertContainer<std::vector<int32_t>>(
+ mGain.values, convertIntegral<int32_t, int>));
+ return OK;
+}
+
+status_t AudioPortConfig::readFromParcel(const Parcel *parcel) {
+ media::AudioPortConfig parcelable;
+ return parcelable.readFromParcel(parcel)
+ ?: readFromParcelable(parcelable);
+}
+
+status_t AudioPortConfig::readFromParcelable(const media::AudioPortConfig& parcelable) {
+ mSamplingRate = VALUE_OR_RETURN_STATUS(convertIntegral<unsigned int>(parcelable.sampleRate));
+ mFormat = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioFormat_audio_format_t(parcelable.format));
+ mChannelMask = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_int32_t_audio_channel_mask_t(parcelable.channelMask));
+ mId = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_port_handle_t(parcelable.id));
+ mGain.index = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.gain.index));
+ mGain.mode = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_int32_t_audio_gain_mode_t(parcelable.gain.mode));
+ mGain.channel_mask = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_int32_t_audio_channel_mask_t(parcelable.gain.channelMask));
+ mGain.ramp_duration_ms = VALUE_OR_RETURN_STATUS(
+ convertIntegral<unsigned int>(parcelable.gain.rampDurationMs));
+ if (parcelable.gain.values.size() > std::size(mGain.values)) {
return BAD_VALUE;
}
- std::copy(values.begin(), values.end(), mGain.values);
- return status;
+ for (size_t i = 0; i < parcelable.gain.values.size(); ++i) {
+ mGain.values[i] = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.gain.values[i]));
+ }
+ return OK;
}
} // namespace android
diff --git a/media/libaudiofoundation/AudioProfile.cpp b/media/libaudiofoundation/AudioProfile.cpp
index ec52575..3b47fed 100644
--- a/media/libaudiofoundation/AudioProfile.cpp
+++ b/media/libaudiofoundation/AudioProfile.cpp
@@ -130,44 +130,73 @@
mIsDynamicRate == other->isDynamicRate();
}
-status_t AudioProfile::writeToParcel(Parcel *parcel) const
-{
- status_t status = NO_ERROR;
- if ((status = parcel->writeUtf8AsUtf16(mName)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mFormat)) != NO_ERROR) return status;
- std::vector<int> values(mChannelMasks.begin(), mChannelMasks.end());
- if ((status = parcel->writeInt32Vector(values)) != NO_ERROR) return status;
- values.clear();
- values.assign(mSamplingRates.begin(), mSamplingRates.end());
- if ((status = parcel->writeInt32Vector(values)) != NO_ERROR) return status;
- if ((status = parcel->writeBool(mIsDynamicFormat)) != NO_ERROR) return status;
- if ((status = parcel->writeBool(mIsDynamicChannels)) != NO_ERROR) return status;
- if ((status = parcel->writeBool(mIsDynamicRate)) != NO_ERROR) return status;
- return status;
+AudioProfile& AudioProfile::operator=(const AudioProfile& other) {
+ mName = other.mName;
+ mFormat = other.mFormat;
+ mChannelMasks = other.mChannelMasks;
+ mSamplingRates = other.mSamplingRates;
+ mIsDynamicFormat = other.mIsDynamicFormat;
+ mIsDynamicChannels = other.mIsDynamicChannels;
+ mIsDynamicRate = other.mIsDynamicRate;
+ return *this;
}
-status_t AudioProfile::readFromParcel(const Parcel *parcel)
-{
- status_t status = NO_ERROR;
- if ((status = parcel->readUtf8FromUtf16(&mName)) != NO_ERROR) return status;
- static_assert(sizeof(mFormat) == sizeof(uint32_t));
- if ((status = parcel->readUint32(reinterpret_cast<uint32_t*>(&mFormat))) != NO_ERROR) {
+status_t AudioProfile::writeToParcel(Parcel *parcel) const {
+ media::AudioProfile parcelable = VALUE_OR_RETURN_STATUS(toParcelable());
+ return parcelable.writeToParcel(parcel);
+ }
+
+ConversionResult<media::AudioProfile>
+AudioProfile::toParcelable() const {
+ media::AudioProfile parcelable;
+ parcelable.name = mName;
+ parcelable.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormat(mFormat));
+ parcelable.channelMasks = VALUE_OR_RETURN(
+ convertContainer<std::vector<int32_t>>(mChannelMasks,
+ legacy2aidl_audio_channel_mask_t_int32_t));
+ parcelable.samplingRates = VALUE_OR_RETURN(
+ convertContainer<std::vector<int32_t>>(mSamplingRates,
+ convertIntegral<int32_t, uint32_t>));
+ parcelable.isDynamicFormat = mIsDynamicFormat;
+ parcelable.isDynamicChannels = mIsDynamicChannels;
+ parcelable.isDynamicRate = mIsDynamicRate;
+ return parcelable;
+}
+
+status_t AudioProfile::readFromParcel(const Parcel *parcel) {
+ media::AudioProfile parcelable;
+ if (status_t status = parcelable.readFromParcel(parcel); status != OK) {
return status;
}
- std::vector<int> values;
- if ((status = parcel->readInt32Vector(&values)) != NO_ERROR) return status;
- mChannelMasks.clear();
- for (auto raw : values) {
- mChannelMasks.insert(static_cast<audio_channel_mask_t>(raw));
- }
- values.clear();
- if ((status = parcel->readInt32Vector(&values)) != NO_ERROR) return status;
- mSamplingRates.clear();
- mSamplingRates.insert(values.begin(), values.end());
- if ((status = parcel->readBool(&mIsDynamicFormat)) != NO_ERROR) return status;
- if ((status = parcel->readBool(&mIsDynamicChannels)) != NO_ERROR) return status;
- if ((status = parcel->readBool(&mIsDynamicRate)) != NO_ERROR) return status;
- return status;
+ *this = *VALUE_OR_RETURN_STATUS(fromParcelable(parcelable));
+ return OK;
+}
+
+ConversionResult<sp<AudioProfile>>
+AudioProfile::fromParcelable(const media::AudioProfile& parcelable) {
+ sp<AudioProfile> legacy = new AudioProfile();
+ legacy->mName = parcelable.name;
+ legacy->mFormat = VALUE_OR_RETURN(aidl2legacy_AudioFormat_audio_format_t(parcelable.format));
+ legacy->mChannelMasks = VALUE_OR_RETURN(
+ convertContainer<ChannelMaskSet>(parcelable.channelMasks,
+ aidl2legacy_int32_t_audio_channel_mask_t));
+ legacy->mSamplingRates = VALUE_OR_RETURN(
+ convertContainer<SampleRateSet>(parcelable.samplingRates,
+ convertIntegral<uint32_t, int32_t>));
+ legacy->mIsDynamicFormat = parcelable.isDynamicFormat;
+ legacy->mIsDynamicChannels = parcelable.isDynamicChannels;
+ legacy->mIsDynamicRate = parcelable.isDynamicRate;
+ return legacy;
+}
+
+ConversionResult<sp<AudioProfile>>
+aidl2legacy_AudioProfile(const media::AudioProfile& aidl) {
+ return AudioProfile::fromParcelable(aidl);
+}
+
+ConversionResult<media::AudioProfile>
+legacy2aidl_AudioProfile(const sp<AudioProfile>& legacy) {
+ return legacy->toParcelable();
}
ssize_t AudioProfileVector::add(const sp<AudioProfile> &profile)
@@ -316,4 +345,14 @@
});
}
+ConversionResult<AudioProfileVector>
+aidl2legacy_AudioProfileVector(const std::vector<media::AudioProfile>& aidl) {
+ return convertContainer<AudioProfileVector>(aidl, aidl2legacy_AudioProfile);
+}
+
+ConversionResult<std::vector<media::AudioProfile>>
+legacy2aidl_AudioProfileVector(const AudioProfileVector& legacy) {
+ return convertContainer<std::vector<media::AudioProfile>>(legacy, legacy2aidl_AudioProfile);
+}
+
} // namespace android
diff --git a/media/libaudiofoundation/DeviceDescriptorBase.cpp b/media/libaudiofoundation/DeviceDescriptorBase.cpp
index 50bda8b..6261559 100644
--- a/media/libaudiofoundation/DeviceDescriptorBase.cpp
+++ b/media/libaudiofoundation/DeviceDescriptorBase.cpp
@@ -19,6 +19,7 @@
#include <android-base/stringprintf.h>
#include <audio_utils/string.h>
+#include <media/AidlConversion.h>
#include <media/DeviceDescriptorBase.h>
#include <media/TypeConverter.h>
@@ -155,26 +156,45 @@
mDeviceTypeAddr.equals(other->mDeviceTypeAddr);
}
+
status_t DeviceDescriptorBase::writeToParcel(Parcel *parcel) const
{
- status_t status = NO_ERROR;
- if ((status = AudioPort::writeToParcel(parcel)) != NO_ERROR) return status;
- if ((status = AudioPortConfig::writeToParcel(parcel)) != NO_ERROR) return status;
- if ((status = parcel->writeParcelable(mDeviceTypeAddr)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mEncapsulationModes)) != NO_ERROR) return status;
- if ((status = parcel->writeUint32(mEncapsulationMetadataTypes)) != NO_ERROR) return status;
- return status;
+ media::DeviceDescriptorBase parcelable;
+ return writeToParcelable(&parcelable)
+ ?: parcelable.writeToParcel(parcel);
}
-status_t DeviceDescriptorBase::readFromParcel(const Parcel *parcel)
-{
- status_t status = NO_ERROR;
- if ((status = AudioPort::readFromParcel(parcel)) != NO_ERROR) return status;
- if ((status = AudioPortConfig::readFromParcel(parcel)) != NO_ERROR) return status;
- if ((status = parcel->readParcelable(&mDeviceTypeAddr)) != NO_ERROR) return status;
- if ((status = parcel->readUint32(&mEncapsulationModes)) != NO_ERROR) return status;
- if ((status = parcel->readUint32(&mEncapsulationMetadataTypes)) != NO_ERROR) return status;
- return status;
+status_t DeviceDescriptorBase::writeToParcelable(media::DeviceDescriptorBase* parcelable) const {
+ AudioPort::writeToParcelable(&parcelable->port);
+ AudioPortConfig::writeToParcelable(&parcelable->portConfig);
+ parcelable->device = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_AudioDeviceTypeAddress(mDeviceTypeAddr));
+ parcelable->encapsulationModes = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_AudioEncapsulationMode_mask(mEncapsulationModes));
+ parcelable->encapsulationMetadataTypes = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_AudioEncapsulationMetadataType_mask(mEncapsulationMetadataTypes));
+ return OK;
+}
+
+status_t DeviceDescriptorBase::readFromParcel(const Parcel *parcel) {
+ media::DeviceDescriptorBase parcelable;
+ return parcelable.readFromParcel(parcel)
+ ?: readFromParcelable(parcelable);
+}
+
+status_t DeviceDescriptorBase::readFromParcelable(const media::DeviceDescriptorBase& parcelable) {
+ status_t status = AudioPort::readFromParcelable(parcelable.port)
+ ?: AudioPortConfig::readFromParcelable(parcelable.portConfig);
+ if (status != OK) {
+ return status;
+ }
+ mDeviceTypeAddr = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioDeviceTypeAddress(parcelable.device));
+ mEncapsulationModes = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioEncapsulationMode_mask(parcelable.encapsulationModes));
+ mEncapsulationMetadataTypes = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioEncapsulationMetadataType_mask(parcelable.encapsulationMetadataTypes));
+ return OK;
}
std::string toString(const DeviceDescriptorBaseVector& devices)
@@ -198,4 +218,24 @@
return deviceTypeAddrs;
}
+ConversionResult<sp<DeviceDescriptorBase>>
+aidl2legacy_DeviceDescriptorBase(const media::DeviceDescriptorBase& aidl) {
+ sp<DeviceDescriptorBase> result = new DeviceDescriptorBase(AUDIO_DEVICE_NONE);
+ status_t status = result->readFromParcelable(aidl);
+ if (status != OK) {
+ return base::unexpected(status);
+ }
+ return result;
+}
+
+ConversionResult<media::DeviceDescriptorBase>
+legacy2aidl_DeviceDescriptorBase(const sp<DeviceDescriptorBase>& legacy) {
+ media::DeviceDescriptorBase aidl;
+ status_t status = legacy->writeToParcelable(&aidl);
+ if (status != OK) {
+ return base::unexpected(status);
+ }
+ return aidl;
+}
+
} // namespace android
diff --git a/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h b/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h
index 7497faf..34da233 100644
--- a/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h
+++ b/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h
@@ -19,9 +19,11 @@
#include <string>
#include <vector>
+#include <android/media/AudioDevice.h>
#include <binder/Parcelable.h>
#include <binder/Parcel.h>
#include <media/AudioContainers.h>
+#include <media/AidlConversion.h>
#include <system/audio.h>
#include <utils/Errors.h>
@@ -84,4 +86,10 @@
std::string dumpAudioDeviceTypeAddrVector(const AudioDeviceTypeAddrVector& deviceTypeAddrs,
bool includeSensitiveInfo=false);
+// Conversion routines, according to AidlConversion.h conventions.
+ConversionResult<AudioDeviceTypeAddr>
+aidl2legacy_AudioDeviceTypeAddress(const media::AudioDevice& aidl);
+ConversionResult<media::AudioDevice>
+legacy2aidl_AudioDeviceTypeAddress(const AudioDeviceTypeAddr& legacy);
+
} // namespace android
diff --git a/media/libaudiofoundation/include/media/AudioGain.h b/media/libaudiofoundation/include/media/AudioGain.h
index 859f1e7..a06b686 100644
--- a/media/libaudiofoundation/include/media/AudioGain.h
+++ b/media/libaudiofoundation/include/media/AudioGain.h
@@ -16,8 +16,10 @@
#pragma once
+#include <android/media/AudioGain.h>
#include <binder/Parcel.h>
#include <binder/Parcelable.h>
+#include <media/AidlConversion.h>
#include <utils/Errors.h>
#include <utils/RefBase.h>
#include <system/audio.h>
@@ -72,6 +74,9 @@
status_t writeToParcel(Parcel* parcel) const override;
status_t readFromParcel(const Parcel* parcel) override;
+ status_t writeToParcelable(media::AudioGain* parcelable) const;
+ status_t readFromParcelable(const media::AudioGain& parcelable);
+
private:
int mIndex;
struct audio_gain mGain;
@@ -79,6 +84,12 @@
bool mUseForVolume = false;
};
+// Conversion routines, according to AidlConversion.h conventions.
+ConversionResult<sp<AudioGain>>
+aidl2legacy_AudioGain(const media::AudioGain& aidl);
+ConversionResult<media::AudioGain>
+legacy2aidl_AudioGain(const sp<AudioGain>& legacy);
+
class AudioGains : public std::vector<sp<AudioGain> >, public Parcelable
{
public:
@@ -104,4 +115,10 @@
status_t readFromParcel(const Parcel* parcel) override;
};
+// Conversion routines, according to AidlConversion.h conventions.
+ConversionResult<AudioGains>
+aidl2legacy_AudioGains(const std::vector<media::AudioGain>& aidl);
+ConversionResult<std::vector<media::AudioGain>>
+legacy2aidl_AudioGains(const AudioGains& legacy);
+
} // namespace android
diff --git a/media/libaudiofoundation/include/media/AudioPort.h b/media/libaudiofoundation/include/media/AudioPort.h
index 622b97b..633e4e3 100644
--- a/media/libaudiofoundation/include/media/AudioPort.h
+++ b/media/libaudiofoundation/include/media/AudioPort.h
@@ -19,6 +19,8 @@
#include <string>
#include <type_traits>
+#include <android/media/AudioPort.h>
+#include <android/media/AudioPortConfig.h>
#include <binder/Parcel.h>
#include <binder/Parcelable.h>
#include <media/AudioGain.h>
@@ -91,6 +93,9 @@
status_t writeToParcel(Parcel* parcel) const override;
status_t readFromParcel(const Parcel* parcel) override;
+ status_t writeToParcelable(media::AudioPort* parcelable) const;
+ status_t readFromParcelable(const media::AudioPort& parcelable);
+
AudioGains mGains; // gain controllers
protected:
std::string mName;
@@ -136,6 +141,8 @@
status_t writeToParcel(Parcel* parcel) const override;
status_t readFromParcel(const Parcel* parcel) override;
+ status_t writeToParcelable(media::AudioPortConfig* parcelable) const;
+ status_t readFromParcelable(const media::AudioPortConfig& parcelable);
protected:
unsigned int mSamplingRate = 0u;
diff --git a/media/libaudiofoundation/include/media/AudioProfile.h b/media/libaudiofoundation/include/media/AudioProfile.h
index ebe855e..57592bc 100644
--- a/media/libaudiofoundation/include/media/AudioProfile.h
+++ b/media/libaudiofoundation/include/media/AudioProfile.h
@@ -19,8 +19,10 @@
#include <string>
#include <vector>
+#include <android/media/AudioProfile.h>
#include <binder/Parcel.h>
#include <binder/Parcelable.h>
+#include <media/AidlConversion.h>
#include <media/AudioContainers.h>
#include <system/audio.h>
#include <utils/RefBase.h>
@@ -73,6 +75,9 @@
status_t writeToParcel(Parcel* parcel) const override;
status_t readFromParcel(const Parcel* parcel) override;
+ ConversionResult<media::AudioProfile> toParcelable() const;
+ static ConversionResult<sp<AudioProfile>> fromParcelable(const media::AudioProfile& parcelable);
+
private:
std::string mName;
audio_format_t mFormat; // The format for an audio profile should only be set when initialized.
@@ -82,8 +87,17 @@
bool mIsDynamicFormat = false;
bool mIsDynamicChannels = false;
bool mIsDynamicRate = false;
+
+ AudioProfile() = default;
+ AudioProfile& operator=(const AudioProfile& other);
};
+// Conversion routines, according to AidlConversion.h conventions.
+ConversionResult<sp<AudioProfile>>
+aidl2legacy_AudioProfile(const media::AudioProfile& aidl);
+ConversionResult<media::AudioProfile>
+legacy2aidl_AudioProfile(const sp<AudioProfile>& legacy);
+
class AudioProfileVector : public std::vector<sp<AudioProfile>>, public Parcelable
{
public:
@@ -117,4 +131,11 @@
bool operator == (const AudioProfile &left, const AudioProfile &right);
+// Conversion routines, according to AidlConversion.h conventions.
+ConversionResult<AudioProfileVector>
+aidl2legacy_AudioProfileVector(const std::vector<media::AudioProfile>& aidl);
+ConversionResult<std::vector<media::AudioProfile>>
+legacy2aidl_AudioProfileVector(const AudioProfileVector& legacy);
+
+
} // namespace android
diff --git a/media/libaudiofoundation/include/media/DeviceDescriptorBase.h b/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
index 9ad8d3d..8a920b7 100644
--- a/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
+++ b/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
@@ -18,6 +18,7 @@
#include <vector>
+#include <android/media/DeviceDescriptorBase.h>
#include <binder/Parcel.h>
#include <binder/Parcelable.h>
#include <media/AudioContainers.h>
@@ -76,6 +77,9 @@
status_t writeToParcel(Parcel* parcel) const override;
status_t readFromParcel(const Parcel* parcel) override;
+ status_t writeToParcelable(media::DeviceDescriptorBase* parcelable) const;
+ status_t readFromParcelable(const media::DeviceDescriptorBase& parcelable);
+
protected:
AudioDeviceTypeAddr mDeviceTypeAddr;
uint32_t mEncapsulationModes = 0;
@@ -107,4 +111,10 @@
*/
AudioDeviceTypeAddrVector deviceTypeAddrsFromDescriptors(const DeviceDescriptorBaseVector& devices);
+// Conversion routines, according to AidlConversion.h conventions.
+ConversionResult<sp<DeviceDescriptorBase>>
+aidl2legacy_DeviceDescriptorBase(const media::DeviceDescriptorBase& aidl);
+ConversionResult<media::DeviceDescriptorBase>
+legacy2aidl_DeviceDescriptorBase(const sp<DeviceDescriptorBase>& legacy);
+
} // namespace android
diff --git a/media/libaudiohal/Android.bp b/media/libaudiohal/Android.bp
index fab0fea..482f40e 100644
--- a/media/libaudiohal/Android.bp
+++ b/media/libaudiohal/Android.bp
@@ -63,8 +63,6 @@
export_include_dirs: ["include"],
// This is needed because the stream interface includes media/MicrophoneInfo.h
- // which is not in any library but has a dependency on headers from libbinder.
- header_libs: ["libbinder_headers"],
-
- export_header_lib_headers: ["libbinder_headers"],
+ header_libs: ["av-headers"],
+ export_header_lib_headers: ["av-headers"],
}
diff --git a/media/libaudiohal/impl/Android.bp b/media/libaudiohal/impl/Android.bp
index df006b5..fe47881 100644
--- a/media/libaudiohal/impl/Android.bp
+++ b/media/libaudiohal/impl/Android.bp
@@ -26,6 +26,7 @@
"android.hardware.audio.common-util",
"android.hidl.allocator@1.0",
"android.hidl.memory@1.0",
+ "av-types-aidl-unstable-cpp",
"libaudiofoundation",
"libaudiohal_deathhandler",
"libaudioutils",
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index c2f9376..77c2550 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -48,6 +48,9 @@
namespace {
+using ::android::hardware::audio::common::CPP_VERSION::AudioPort;
+using ::android::hardware::audio::common::CPP_VERSION::AudioPortConfig;
+
status_t deviceAddressFromHal(
audio_devices_t device, const char* halAddress, DeviceAddress* address) {
address->device = AudioDevice(device);
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index 1a7eb6f..f68f65d 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -5,12 +5,14 @@
export_include_dirs: ["include"],
header_libs: [
+ "av-headers",
"libbase_headers",
"libgui_headers",
"libstagefright_headers",
"media_plugin_headers",
],
export_header_lib_headers: [
+ "av-headers",
"libgui_headers",
"libstagefright_headers",
"media_plugin_headers",
diff --git a/media/libmediahelper/Android.bp b/media/libmediahelper/Android.bp
index 6c43fe0..849debf 100644
--- a/media/libmediahelper/Android.bp
+++ b/media/libmediahelper/Android.bp
@@ -9,6 +9,12 @@
enabled: false,
},
},
+ apex_available: [
+ "//apex_available:platform",
+ "com.android.bluetooth.updatable",
+ "com.android.media",
+ "com.android.media.swcodec",
+ ],
}
cc_library {
diff --git a/media/libshmem/Android.bp b/media/libshmem/Android.bp
index b549b5d..0e4ff48 100644
--- a/media/libshmem/Android.bp
+++ b/media/libshmem/Android.bp
@@ -14,6 +14,9 @@
name: "libshmemcompat",
export_include_dirs: ["include"],
srcs: ["ShmemCompat.cpp"],
+ host_supported: true,
+ vendor_available: true,
+ double_loadable: true,
shared_libs: [
"libbinder",
"libshmemutil",
@@ -31,6 +34,9 @@
name: "libshmemutil",
export_include_dirs: ["include"],
srcs: ["ShmemUtil.cpp"],
+ host_supported: true,
+ vendor_available: true,
+ double_loadable: true,
shared_libs: [
"shared-file-region-aidl-unstable-cpp",
],
diff --git a/services/audioflinger/Android.bp b/services/audioflinger/Android.bp
index 7443320..87b5ff4 100644
--- a/services/audioflinger/Android.bp
+++ b/services/audioflinger/Android.bp
@@ -38,6 +38,7 @@
"audioflinger-aidl-unstable-cpp",
"audioclient-types-aidl-unstable-cpp",
"av-types-aidl-unstable-cpp",
+ "effect-aidl-unstable-cpp",
"libaudioclient_aidl_conversion",
"libaudiofoundation",
"libaudiohal",
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 959e858..9ba99bc 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -22,15 +22,6 @@
// Define AUDIO_ARRAYS_STATIC_CHECK to check all audio arrays are correct
#define AUDIO_ARRAYS_STATIC_CHECK 1
-#define VALUE_OR_FATAL(result) \
- ({ \
- auto _tmp = (result); \
- LOG_ALWAYS_FATAL_IF(!_tmp.ok(), \
- "Failed result (%d)", \
- _tmp.error()); \
- std::move(_tmp.value()); \
- })
-
#include "Configuration.h"
#include <dirent.h>
#include <math.h>
@@ -91,6 +82,15 @@
#include "TypedLogger.h"
+#define VALUE_OR_FATAL(result) \
+ ({ \
+ auto _tmp = (result); \
+ LOG_ALWAYS_FATAL_IF(!_tmp.ok(), \
+ "Failed result (%d)", \
+ _tmp.error()); \
+ std::move(_tmp.value()); \
+ })
+
// ----------------------------------------------------------------------------
// Note: the following macro is used for extremely verbose logging message. In
@@ -757,25 +757,11 @@
// IAudioFlinger interface
-sp<IAudioTrack> AudioFlinger::createTrack(const media::CreateTrackRequest& _input,
- media::CreateTrackResponse& _output,
- status_t* status)
+status_t AudioFlinger::createTrack(const media::CreateTrackRequest& _input,
+ media::CreateTrackResponse& _output)
{
// Local version of VALUE_OR_RETURN, specific to this method's calling conventions.
-#define VALUE_OR_EXIT(expr) \
- ({ \
- auto _tmp = (expr); \
- if (!_tmp.ok()) { \
- *status = _tmp.error(); \
- return nullptr; \
- } \
- std::move(_tmp.value()); \
- })
-
- CreateTrackInput input = VALUE_OR_EXIT(CreateTrackInput::fromAidl(_input));
-
-#undef VALUE_OR_EXIT
-
+ CreateTrackInput input = VALUE_OR_RETURN_STATUS(CreateTrackInput::fromAidl(_input));
CreateTrackOutput output;
sp<PlaybackThread::Track> track;
@@ -1034,17 +1020,14 @@
AudioSystem::moveEffectsToIo(effectIds, effectThreadId);
}
+ output.audioTrack = new TrackHandle(track);
_output = VALUE_OR_FATAL(output.toAidl());
- // return handle to client
- trackHandle = new TrackHandle(track);
-
Exit:
if (lStatus != NO_ERROR && output.outputId != AUDIO_IO_HANDLE_NONE) {
AudioSystem::releaseOutput(portId);
}
- *status = lStatus;
- return trackHandle;
+ return lStatus;
}
uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const
@@ -2018,24 +2001,10 @@
// ----------------------------------------------------------------------------
-sp<media::IAudioRecord> AudioFlinger::createRecord(const media::CreateRecordRequest& _input,
- media::CreateRecordResponse& _output,
- status_t* status)
+status_t AudioFlinger::createRecord(const media::CreateRecordRequest& _input,
+ media::CreateRecordResponse& _output)
{
- // Local version of VALUE_OR_RETURN, specific to this method's calling conventions.
-#define VALUE_OR_EXIT(expr) \
- ({ \
- auto _tmp = (expr); \
- if (!_tmp.ok()) { \
- *status = _tmp.error(); \
- return nullptr; \
- } \
- std::move(_tmp.value()); \
- })
-
- CreateRecordInput input = VALUE_OR_EXIT(CreateRecordInput::fromAidl(_input));
-
-#undef VALUE_OR_EXIT
+ CreateRecordInput input = VALUE_OR_RETURN_STATUS(CreateRecordInput::fromAidl(_input));
CreateRecordOutput output;
sp<RecordThread::RecordTrack> recordTrack;
@@ -2175,11 +2144,9 @@
output.buffers = recordTrack->getBuffers();
output.portId = portId;
+ output.audioRecord = new RecordHandle(recordTrack);
_output = VALUE_OR_FATAL(output.toAidl());
- // return handle to client
- recordHandle = new RecordHandle(recordTrack);
-
Exit:
if (lStatus != NO_ERROR) {
// remove local strong reference to Client before deleting the RecordTrack so that the
@@ -2196,8 +2163,7 @@
}
}
- *status = lStatus;
- return recordHandle;
+ return lStatus;
}
@@ -2602,20 +2568,28 @@
return 0;
}
-status_t AudioFlinger::openOutput(audio_module_handle_t module,
- audio_io_handle_t *output,
- audio_config_t *config,
- const sp<DeviceDescriptorBase>& device,
- uint32_t *latencyMs,
- audio_output_flags_t flags)
+status_t AudioFlinger::openOutput(const media::OpenOutputRequest& request,
+ media::OpenOutputResponse* response)
{
+ audio_module_handle_t module = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_int32_t_audio_module_handle_t(request.module));
+ audio_config_t config = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioConfig_audio_config_t(request.config));
+ sp<DeviceDescriptorBase> device = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_DeviceDescriptorBase(request.device));
+ audio_output_flags_t flags = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_audio_output_flags_mask(request.flags));
+
+ audio_io_handle_t output;
+ uint32_t latencyMs;
+
ALOGI("openOutput() this %p, module %d Device %s, SamplingRate %d, Format %#08x, "
"Channels %#x, flags %#x",
this, module,
device->toString().c_str(),
- config->sample_rate,
- config->format,
- config->channel_mask,
+ config.sample_rate,
+ config.format,
+ config.channel_mask,
flags);
audio_devices_t deviceType = device->type();
@@ -2627,11 +2601,11 @@
Mutex::Autolock _l(mLock);
- sp<ThreadBase> thread = openOutput_l(module, output, config, deviceType, address, flags);
+ sp<ThreadBase> thread = openOutput_l(module, &output, &config, deviceType, address, flags);
if (thread != 0) {
if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
- *latencyMs = playbackThread->latency();
+ latencyMs = playbackThread->latency();
// notify client processes of the new output creation
playbackThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
@@ -2651,6 +2625,10 @@
MmapThread *mmapThread = (MmapThread *)thread.get();
mmapThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
}
+ response->output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
+ response->config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+ response->latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(latencyMs));
+ response->flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_mask(flags));
return NO_ERROR;
}
@@ -2803,22 +2781,36 @@
return NO_ERROR;
}
-status_t AudioFlinger::openInput(audio_module_handle_t module,
- audio_io_handle_t *input,
- audio_config_t *config,
- audio_devices_t *devices,
- const String8& address,
- audio_source_t source,
- audio_input_flags_t flags)
+status_t AudioFlinger::openInput(const media::OpenInputRequest& request,
+ media::OpenInputResponse* response)
{
Mutex::Autolock _l(mLock);
- if (*devices == AUDIO_DEVICE_NONE) {
+ if (request.device.type == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
+ audio_io_handle_t input = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_int32_t_audio_io_handle_t(request.input));
+ audio_config_t config = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioConfig_audio_config_t(request.config));
+ AudioDeviceTypeAddr device = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioDeviceTypeAddress(request.device));
+
sp<ThreadBase> thread = openInput_l(
- module, input, config, *devices, address, source, flags, AUDIO_DEVICE_NONE, String8{});
+ VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_module_handle_t(request.module)),
+ &input,
+ &config,
+ device.mType,
+ device.address().c_str(),
+ VALUE_OR_RETURN_STATUS(aidl2legacy_AudioSourceType_audio_source_t(request.source)),
+ VALUE_OR_RETURN_STATUS(aidl2legacy_audio_input_flags_mask(request.flags)),
+ AUDIO_DEVICE_NONE,
+ String8{});
+
+ response->input = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(input));
+ response->config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+ response->device = request.device;
if (thread != 0) {
// notify client processes of the new input creation
@@ -2832,7 +2824,7 @@
audio_io_handle_t *input,
audio_config_t *config,
audio_devices_t devices,
- const String8& address,
+ const char* address,
audio_source_t source,
audio_input_flags_t flags,
audio_devices_t outputDevice,
@@ -2862,7 +2854,7 @@
sp<DeviceHalInterface> inHwHal = inHwDev->hwDevice();
sp<StreamInHalInterface> inStream;
status_t status = inHwHal->openInputStream(
- *input, devices, &halconfig, flags, address.string(), source,
+ *input, devices, &halconfig, flags, address, source,
outputDevice, outputDeviceAddress, &inStream);
ALOGV("openInput_l() openInputStream returned input %p, devices %#x, SamplingRate %d"
", Format %#x, Channels %#x, flags %#x, status %d addr %s",
@@ -2872,7 +2864,7 @@
halconfig.format,
halconfig.channel_mask,
flags,
- status, address.string());
+ status, address);
// If the input could not be opened with the requested parameters and we can handle the
// conversion internally, try to open again with the proposed parameters.
@@ -2886,7 +2878,7 @@
ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
inStream.clear();
status = inHwHal->openInputStream(
- *input, devices, &halconfig, flags, address.string(), source,
+ *input, devices, &halconfig, flags, address, source,
outputDevice, outputDeviceAddress, &inStream);
// FIXME log this new status; HAL should not propose any further changes
}
@@ -3499,23 +3491,29 @@
return status;
}
-sp<media::IEffect> AudioFlinger::createEffect(
- effect_descriptor_t *pDesc,
- const sp<IEffectClient>& effectClient,
- int32_t priority,
- audio_io_handle_t io,
- audio_session_t sessionId,
- const AudioDeviceTypeAddr& device,
- const String16& opPackageName,
- pid_t pid,
- bool probe,
- status_t *status,
- int *id,
- int *enabled)
-{
- status_t lStatus = NO_ERROR;
+status_t AudioFlinger::createEffect(const media::CreateEffectRequest& request,
+ media::CreateEffectResponse* response) {
+ const sp<IEffectClient>& effectClient = request.client;
+ const int32_t priority = request.priority;
+ const AudioDeviceTypeAddr device = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioDeviceTypeAddress(request.device));
+ const String16 opPackageName = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_string_view_String16(request.opPackageName));
+ pid_t pid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(request.pid));
+ const audio_session_t sessionId = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_int32_t_audio_session_t(request.sessionId));
+ audio_io_handle_t io = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_int32_t_audio_io_handle_t(request.output));
+ const effect_descriptor_t descIn = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_EffectDescriptor_effect_descriptor_t(request.desc));
+ const bool probe = request.probe;
+
sp<EffectHandle> handle;
- effect_descriptor_t desc;
+ effect_descriptor_t descOut;
+ int enabledOut = 0;
+ int idOut = -1;
+
+ status_t lStatus = NO_ERROR;
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
if (pid == -1 || !isAudioServerOrMediaServerUid(callingUid)) {
@@ -3527,12 +3525,7 @@
}
ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d, factory %p",
- pid, effectClient.get(), priority, sessionId, io, mEffectsFactoryHal.get());
-
- if (pDesc == NULL) {
- lStatus = BAD_VALUE;
- goto Exit;
- }
+ pid, effectClient.get(), priority, sessionId, io, mEffectsFactoryHal.get());
if (mEffectsFactoryHal == 0) {
ALOGE("%s: no effects factory hal", __func__);
@@ -3589,7 +3582,7 @@
// otherwise no preference.
uint32_t preferredType = (sessionId == AUDIO_SESSION_OUTPUT_MIX ?
EFFECT_FLAG_TYPE_AUXILIARY : EFFECT_FLAG_TYPE_MASK);
- lStatus = getEffectDescriptor(&pDesc->uuid, &pDesc->type, preferredType, &desc);
+ lStatus = getEffectDescriptor(&descIn.uuid, &descIn.type, preferredType, &descOut);
if (lStatus < 0) {
ALOGW("createEffect() error %d from getEffectDescriptor", lStatus);
goto Exit;
@@ -3597,20 +3590,20 @@
// Do not allow auxiliary effects on a session different from 0 (output mix)
if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
- (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+ (descOut.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
lStatus = INVALID_OPERATION;
goto Exit;
}
// check recording permission for visualizer
- if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
+ if ((memcmp(&descOut.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
// TODO: Do we need to start/stop op - i.e. is there recording being performed?
!recordingAllowed(opPackageName, pid, callingUid)) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
- const bool hapticPlaybackRequired = EffectModule::isHapticGenerator(&desc.type);
+ const bool hapticPlaybackRequired = EffectModule::isHapticGenerator(&descOut.type);
if (hapticPlaybackRequired
&& (sessionId == AUDIO_SESSION_DEVICE
|| sessionId == AUDIO_SESSION_OUTPUT_MIX
@@ -3620,13 +3613,11 @@
goto Exit;
}
- // return effect descriptor
- *pDesc = desc;
if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
// if the output returned by getOutputForEffect() is removed before we lock the
// mutex below, the call to checkPlaybackThread_l(io) below will detect it
// and we will exit safely
- io = AudioSystem::getOutputForEffect(&desc);
+ io = AudioSystem::getOutputForEffect(&descOut);
ALOGV("createEffect got output %d", io);
}
@@ -3636,15 +3627,15 @@
sp<Client> client = registerPid(pid);
ALOGV("%s device type %#x address %s", __func__, device.mType, device.getAddress());
handle = mDeviceEffectManager.createEffect_l(
- &desc, device, client, effectClient, mPatchPanel.patches_l(),
- enabled, &lStatus, probe);
+ &descOut, device, client, effectClient, mPatchPanel.patches_l(),
+ &enabledOut, &lStatus, probe);
if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
// remove local strong reference to Client with mClientLock held
Mutex::Autolock _cl(mClientLock);
client.clear();
} else {
// handle must be valid here, but check again to be safe.
- if (handle.get() != nullptr && id != nullptr) *id = handle->id();
+ if (handle.get() != nullptr) idOut = handle->id();
}
goto Register;
}
@@ -3674,8 +3665,8 @@
// Detect if the effect is created after an AudioRecord is destroyed.
if (getOrphanEffectChain_l(sessionId).get() != nullptr) {
ALOGE("%s: effect %s with no specified io handle is denied because the AudioRecord"
- " for session %d no longer exists",
- __func__, desc.name, sessionId);
+ " for session %d no longer exists",
+ __func__, descOut.name, sessionId);
lStatus = PERMISSION_DENIED;
goto Exit;
}
@@ -3689,7 +3680,7 @@
if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) {
io = mPlaybackThreads.keyAt(0);
}
- ALOGV("createEffect() got io %d for effect %s", io, desc.name);
+ ALOGV("createEffect() got io %d for effect %s", io, descOut.name);
} else if (checkPlaybackThread_l(io) != nullptr) {
// allow only one effect chain per sessionId on mPlaybackThreads.
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
@@ -3709,7 +3700,7 @@
mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId);
if ((sessionType & ThreadBase::EFFECT_SESSION) != 0) {
ALOGE("%s: effect %s io %d denied because session %d effect exists on io %d",
- __func__, desc.name, (int)io, (int)sessionId, (int)checkIo);
+ __func__, descOut.name, (int) io, (int) sessionId, (int) checkIo);
android_errorWriteLog(0x534e4554, "123237974");
lStatus = BAD_VALUE;
goto Exit;
@@ -3756,14 +3747,14 @@
}
}
handle = thread->createEffect_l(client, effectClient, priority, sessionId,
- &desc, enabled, &lStatus, pinned, probe);
+ &descOut, &enabledOut, &lStatus, pinned, probe);
if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
// remove local strong reference to Client with mClientLock held
Mutex::Autolock _cl(mClientLock);
client.clear();
} else {
// handle must be valid here, but check again to be safe.
- if (handle.get() != nullptr && id != nullptr) *id = handle->id();
+ if (handle.get() != nullptr) idOut = handle->id();
// Invalidate audio session when haptic playback is created.
if (hapticPlaybackRequired && oriThread != nullptr) {
// invalidateTracksForAudioSession will trigger locking the thread.
@@ -3786,9 +3777,14 @@
handle.clear();
}
+ response->id = idOut;
+ response->enabled = enabledOut != 0;
+ response->effect = handle;
+ response->desc = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_effect_descriptor_t_EffectDescriptor(descOut));
+
Exit:
- *status = lStatus;
- return handle;
+ return lStatus;
}
status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput,
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index b7206b0..6dfc48f 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -33,6 +33,7 @@
#include <sys/types.h>
#include <limits.h>
+#include <android/media/BnAudioTrack.h>
#include <android/media/IAudioFlingerClient.h>
#include <android/media/IAudioTrackCallback.h>
#include <android/os/BnExternalVibrationController.h>
@@ -43,7 +44,6 @@
#include <cutils/properties.h>
#include <media/IAudioFlinger.h>
-#include <media/IAudioTrack.h>
#include <media/AudioSystem.h>
#include <media/AudioTrack.h>
#include <media/MmapStreamInterface.h>
@@ -135,13 +135,11 @@
virtual status_t dump(int fd, const Vector<String16>& args);
// IAudioFlinger interface, in binder opcode order
- virtual sp<IAudioTrack> createTrack(const media::CreateTrackRequest& input,
- media::CreateTrackResponse& output,
- status_t* status) override;
+ status_t createTrack(const media::CreateTrackRequest& input,
+ media::CreateTrackResponse& output) override;
- virtual sp<media::IAudioRecord> createRecord(const media::CreateRecordRequest& input,
- media::CreateRecordResponse& output,
- status_t* status) override;
+ status_t createRecord(const media::CreateRecordRequest& input,
+ media::CreateRecordResponse& output) override;
virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const;
virtual audio_format_t format(audio_io_handle_t output) const;
@@ -182,12 +180,8 @@
virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
audio_channel_mask_t channelMask) const;
- virtual status_t openOutput(audio_module_handle_t module,
- audio_io_handle_t *output,
- audio_config_t *config,
- const sp<DeviceDescriptorBase>& device,
- uint32_t *latencyMs,
- audio_output_flags_t flags);
+ virtual status_t openOutput(const media::OpenOutputRequest& request,
+ media::OpenOutputResponse* response);
virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
audio_io_handle_t output2);
@@ -198,13 +192,8 @@
virtual status_t restoreOutput(audio_io_handle_t output);
- virtual status_t openInput(audio_module_handle_t module,
- audio_io_handle_t *input,
- audio_config_t *config,
- audio_devices_t *device,
- const String8& address,
- audio_source_t source,
- audio_input_flags_t flags);
+ virtual status_t openInput(const media::OpenInputRequest& request,
+ media::OpenInputResponse* response);
virtual status_t closeInput(audio_io_handle_t input);
@@ -233,19 +222,8 @@
uint32_t preferredTypeFlag,
effect_descriptor_t *descriptor) const;
- virtual sp<media::IEffect> createEffect(
- effect_descriptor_t *pDesc,
- const sp<media::IEffectClient>& effectClient,
- int32_t priority,
- audio_io_handle_t io,
- audio_session_t sessionId,
- const AudioDeviceTypeAddr& device,
- const String16& opPackageName,
- pid_t pid,
- bool probe,
- status_t *status /*non-NULL*/,
- int *id,
- int *enabled);
+ virtual status_t createEffect(const media::CreateEffectRequest& request,
+ media::CreateEffectResponse* response);
virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput,
audio_io_handle_t dstOutput);
@@ -626,27 +604,30 @@
}
// server side of the client's IAudioTrack
- class TrackHandle : public android::BnAudioTrack {
+ class TrackHandle : public android::media::BnAudioTrack {
public:
explicit TrackHandle(const sp<PlaybackThread::Track>& track);
virtual ~TrackHandle();
- virtual sp<IMemory> getCblk() const;
- virtual status_t start();
- virtual void stop();
- virtual void flush();
- virtual void pause();
- virtual status_t attachAuxEffect(int effectId);
- virtual status_t setParameters(const String8& keyValuePairs);
- virtual status_t selectPresentation(int presentationId, int programId);
- virtual media::VolumeShaper::Status applyVolumeShaper(
- const sp<media::VolumeShaper::Configuration>& configuration,
- const sp<media::VolumeShaper::Operation>& operation) override;
- virtual sp<media::VolumeShaper::State> getVolumeShaperState(int id) override;
- virtual status_t getTimestamp(AudioTimestamp& timestamp);
- virtual void signal(); // signal playback thread for a change in control block
- virtual status_t onTransact(
- uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
+ binder::Status getCblk(std::optional<media::SharedFileRegion>* _aidl_return) override;
+ binder::Status start(int32_t* _aidl_return) override;
+ binder::Status stop() override;
+ binder::Status flush() override;
+ binder::Status pause() override;
+ binder::Status attachAuxEffect(int32_t effectId, int32_t* _aidl_return) override;
+ binder::Status setParameters(const std::string& keyValuePairs,
+ int32_t* _aidl_return) override;
+ binder::Status selectPresentation(int32_t presentationId, int32_t programId,
+ int32_t* _aidl_return) override;
+ binder::Status getTimestamp(media::AudioTimestampInternal* timestamp,
+ int32_t* _aidl_return) override;
+ binder::Status signal() override;
+ binder::Status applyVolumeShaper(const media::VolumeShaperConfiguration& configuration,
+ const media::VolumeShaperOperation& operation,
+ int32_t* _aidl_return) override;
+ binder::Status getVolumeShaperState(
+ int32_t id,
+ std::optional<media::VolumeShaperState>* _aidl_return) override;
private:
const sp<PlaybackThread::Track> mTrack;
@@ -661,7 +642,7 @@
int /*audio_session_t*/ triggerSession);
virtual binder::Status stop();
virtual binder::Status getActiveMicrophones(
- std::vector<media::MicrophoneInfo>* activeMicrophones);
+ std::vector<media::MicrophoneInfoData>* activeMicrophones);
virtual binder::Status setPreferredMicrophoneDirection(
int /*audio_microphone_direction_t*/ direction);
virtual binder::Status setPreferredMicrophoneFieldDimension(float zoom);
@@ -707,7 +688,7 @@
audio_io_handle_t *input,
audio_config_t *config,
audio_devices_t device,
- const String8& address,
+ const char* address,
audio_source_t source,
audio_input_flags_t flags,
audio_devices_t outputDevice,
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 1a12a5f..68b709f 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -54,6 +54,7 @@
namespace android {
+using binder::Status;
using media::VolumeShaper;
// ----------------------------------------------------------------------------
// TrackBase
@@ -319,64 +320,98 @@
mTrack->destroy();
}
-sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
- return mTrack->getCblk();
+Status AudioFlinger::TrackHandle::getCblk(
+ std::optional<media::SharedFileRegion>* _aidl_return) {
+ *_aidl_return = legacy2aidl_NullableIMemory_SharedFileRegion(mTrack->getCblk()).value();
+ return Status::ok();
}
-status_t AudioFlinger::TrackHandle::start() {
- return mTrack->start();
+Status AudioFlinger::TrackHandle::start(int32_t* _aidl_return) {
+ *_aidl_return = mTrack->start();
+ return Status::ok();
}
-void AudioFlinger::TrackHandle::stop() {
+Status AudioFlinger::TrackHandle::stop() {
mTrack->stop();
+ return Status::ok();
}
-void AudioFlinger::TrackHandle::flush() {
+Status AudioFlinger::TrackHandle::flush() {
mTrack->flush();
+ return Status::ok();
}
-void AudioFlinger::TrackHandle::pause() {
+Status AudioFlinger::TrackHandle::pause() {
mTrack->pause();
+ return Status::ok();
}
-status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
-{
- return mTrack->attachAuxEffect(EffectId);
+Status AudioFlinger::TrackHandle::attachAuxEffect(int32_t effectId,
+ int32_t* _aidl_return) {
+ *_aidl_return = mTrack->attachAuxEffect(effectId);
+ return Status::ok();
}
-status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) {
- return mTrack->setParameters(keyValuePairs);
+Status AudioFlinger::TrackHandle::setParameters(const std::string& keyValuePairs,
+ int32_t* _aidl_return) {
+ *_aidl_return = mTrack->setParameters(String8(keyValuePairs.c_str()));
+ return Status::ok();
}
-status_t AudioFlinger::TrackHandle::selectPresentation(int presentationId, int programId) {
- return mTrack->selectPresentation(presentationId, programId);
+Status AudioFlinger::TrackHandle::selectPresentation(int32_t presentationId, int32_t programId,
+ int32_t* _aidl_return) {
+ *_aidl_return = mTrack->selectPresentation(presentationId, programId);
+ return Status::ok();
}
-VolumeShaper::Status AudioFlinger::TrackHandle::applyVolumeShaper(
- const sp<VolumeShaper::Configuration>& configuration,
- const sp<VolumeShaper::Operation>& operation) {
- return mTrack->applyVolumeShaper(configuration, operation);
+Status AudioFlinger::TrackHandle::getTimestamp(media::AudioTimestampInternal* timestamp,
+ int32_t* _aidl_return) {
+ AudioTimestamp legacy;
+ *_aidl_return = mTrack->getTimestamp(legacy);
+ if (*_aidl_return != OK) {
+ return Status::ok();
+ }
+ *timestamp = legacy2aidl_AudioTimestamp(legacy).value();
+ return Status::ok();
}
-sp<VolumeShaper::State> AudioFlinger::TrackHandle::getVolumeShaperState(int id) {
- return mTrack->getVolumeShaperState(id);
+Status AudioFlinger::TrackHandle::signal() {
+ mTrack->signal();
+ return Status::ok();
}
-status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp)
-{
- return mTrack->getTimestamp(timestamp);
+Status AudioFlinger::TrackHandle::applyVolumeShaper(
+ const media::VolumeShaperConfiguration& configuration,
+ const media::VolumeShaperOperation& operation,
+ int32_t* _aidl_return) {
+ sp<VolumeShaper::Configuration> conf = new VolumeShaper::Configuration();
+ *_aidl_return = conf->readFromParcelable(configuration);
+ if (*_aidl_return != OK) {
+ return Status::ok();
+ }
+
+ sp<VolumeShaper::Operation> op = new VolumeShaper::Operation();
+ *_aidl_return = op->readFromParcelable(operation);
+ if (*_aidl_return != OK) {
+ return Status::ok();
+ }
+
+ *_aidl_return = mTrack->applyVolumeShaper(conf, op);
+ return Status::ok();
}
-
-void AudioFlinger::TrackHandle::signal()
-{
- return mTrack->signal();
-}
-
-status_t AudioFlinger::TrackHandle::onTransact(
- uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
- return BnAudioTrack::onTransact(code, data, reply, flags);
+Status AudioFlinger::TrackHandle::getVolumeShaperState(
+ int32_t id,
+ std::optional<media::VolumeShaperState>* _aidl_return) {
+ sp<VolumeShaper::State> legacy = mTrack->getVolumeShaperState(id);
+ if (legacy == nullptr) {
+ _aidl_return->reset();
+ return Status::ok();
+ }
+ media::VolumeShaperState aidl;
+ legacy->writeToParcelable(&aidl);
+ *_aidl_return = aidl;
+ return Status::ok();
}
// ----------------------------------------------------------------------------
@@ -2112,10 +2147,15 @@
}
binder::Status AudioFlinger::RecordHandle::getActiveMicrophones(
- std::vector<media::MicrophoneInfo>* activeMicrophones) {
+ std::vector<media::MicrophoneInfoData>* activeMicrophones) {
ALOGV("%s()", __func__);
- return binder::Status::fromStatusT(
- mRecordTrack->getActiveMicrophones(activeMicrophones));
+ std::vector<media::MicrophoneInfo> mics;
+ status_t status = mRecordTrack->getActiveMicrophones(&mics);
+ activeMicrophones->resize(mics.size());
+ for (size_t i = 0; status == OK && i < mics.size(); ++i) {
+ status = mics[i].writeToParcelable(&activeMicrophones->at(i));
+ }
+ return binder::Status::fromStatusT(status);
}
binder::Status AudioFlinger::RecordHandle::setPreferredMicrophoneDirection(
diff --git a/services/audiopolicy/service/Android.bp b/services/audiopolicy/service/Android.bp
index 8a7a1b2..e823efa 100644
--- a/services/audiopolicy/service/Android.bp
+++ b/services/audiopolicy/service/Android.bp
@@ -15,6 +15,7 @@
shared_libs: [
"libaudioclient",
+ "libaudioclient_aidl_conversion",
"libaudiofoundation",
"libaudiopolicymanager",
"libaudioutils",
@@ -28,6 +29,8 @@
"libmediautils",
"libsensorprivacy",
"libutils",
+ "audioclient-types-aidl-unstable-cpp",
+ "audioflinger-aidl-unstable-cpp",
"capture_state_listener-aidl-cpp",
],
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index 1cc6d48..dd128a2 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -50,7 +50,22 @@
ALOGW("%s: could not get AudioFlinger", __func__);
return PERMISSION_DENIED;
}
- return af->openOutput(module, output, config, device, latencyMs, flags);
+
+ media::OpenOutputRequest request;
+ media::OpenOutputResponse response;
+
+ request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
+ request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(*config));
+ request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
+ request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_mask(flags));
+
+ status_t status = af->openOutput(request, &response);
+ if (status == OK) {
+ *output = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_io_handle_t(response.output));
+ *config = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioConfig_audio_config_t(response.config));
+ *latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral<uint32_t>(response.latencyMs));
+ }
+ return status;
}
audio_io_handle_t AudioPolicyService::AudioPolicyClient::openDuplicateOutput(
@@ -111,7 +126,22 @@
return PERMISSION_DENIED;
}
- return af->openInput(module, input, config, device, address, source, flags);
+ AudioDeviceTypeAddr deviceTypeAddr(*device, address.c_str());
+
+ media::OpenInputRequest request;
+ request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
+ request.input = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(*input));
+ request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(*config));
+ request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(deviceTypeAddr));
+ request.source = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_source_t_AudioSourceType(source));
+ request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_input_flags_mask(flags));
+
+ media::OpenInputResponse response;
+ status_t status = af->openInput(request, &response);
+ if (status == OK) {
+ *input = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_module_handle_t(response.input));
+ }
+ return status;
}
status_t AudioPolicyService::AudioPolicyClient::closeInput(audio_io_handle_t input)