Merge "Codec2: Add sys-prop to allow dmabuf heap usage to be forced"
diff --git a/METADATA b/METADATA
index d97975c..1fbda08 100644
--- a/METADATA
+++ b/METADATA
@@ -1,3 +1,7 @@
+# *** THIS PACKAGE HAS SPECIAL LICENSING CONDITIONS. PLEASE
+# CONSULT THE OWNERS AND opensource-licensing@google.com BEFORE
+# DEPENDING ON IT IN YOUR PROJECT. ***
third_party {
- license_type: NOTICE
+ # would be NOTICE save for drm/mediadrm/plugins/clearkey/hidl/
+ license_type: BY_EXCEPTION_ONLY
}
diff --git a/media/libeffects/preprocessing/tests/Android.bp b/media/libeffects/preprocessing/tests/Android.bp
index b439880..5e8255a 100644
--- a/media/libeffects/preprocessing/tests/Android.bp
+++ b/media/libeffects/preprocessing/tests/Android.bp
@@ -19,3 +19,14 @@
"libhardware_headers",
],
}
+
+cc_test {
+ name: "correlation",
+ host_supported: true,
+ srcs: ["correlation.cpp"],
+ cflags: [
+ "-Wall",
+ "-Werror",
+ "-Wextra",
+ ],
+}
diff --git a/media/libeffects/preprocessing/tests/correlation.cpp b/media/libeffects/preprocessing/tests/correlation.cpp
new file mode 100644
index 0000000..b13dcc7
--- /dev/null
+++ b/media/libeffects/preprocessing/tests/correlation.cpp
@@ -0,0 +1,166 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <iostream>
+#include <vector>
+
+constexpr int kMinLoopLimitValue = 1;
+constexpr int kNumPeaks = 3;
+
+/*!
+ \brief Compute the length normalized correlation of two signals
+
+ \sigX Pointer to signal 1
+ \sigY Pointer to signal 2
+ \len Length of signals
+ \enableCrossCorr Flag to be set to 1 if cross-correlation is needed
+
+ \return First value is vector of correlation peak indices
+ Second value is vector of correlation peak values
+*/
+
+static std::pair<std::vector<int>, std::vector<float>> correlation(const int16_t* sigX,
+ const int16_t* sigY, int len,
+ int16_t enableCrossCorr) {
+ float maxCorrVal = 0.f, prevCorrVal = 0.f;
+ int delay = 0, peakIndex = 0, flag = 0;
+ int loopLim = (1 == enableCrossCorr) ? len : kMinLoopLimitValue;
+ std::vector<int> peakIndexVect(kNumPeaks, 0);
+ std::vector<float> peakValueVect(kNumPeaks, 0.f);
+ for (int i = 0; i < loopLim; i++) {
+ float corrVal = 0.f;
+ for (int j = i; j < len; j++) {
+ corrVal += (float)(sigX[j] * sigY[j - i]);
+ }
+ corrVal /= len - i;
+ if (corrVal > maxCorrVal) {
+ delay = i;
+ maxCorrVal = corrVal;
+ }
+ // Correlation peaks are expected to be observed at equal intervals. The interval length is
+ // expected to match with wave period.
+ // The following block of code saves the first kNumPeaks number of peaks and the index at
+ // which they occur.
+ if (peakIndex < kNumPeaks) {
+ if (corrVal > prevCorrVal) {
+ peakIndexVect[peakIndex] = i;
+ peakValueVect[peakIndex] = corrVal;
+ flag = 0;
+ } else if (0 == flag) {
+ peakIndex++;
+ flag = 1;
+ }
+ }
+ if (peakIndex == kNumPeaks) break;
+ prevCorrVal = corrVal;
+ }
+ return {peakIndexVect, peakValueVect};
+}
+
+void printUsage() {
+ printf("\nUsage: ");
+ printf("\n correlation <firstFile> <secondFile> [enableCrossCorr]\n");
+ printf("\nwhere, \n <firstFile> is the first file name");
+ printf("\n <secondFile> is the second file name");
+ printf("\n [enableCrossCorr] is flag to set for cross-correlation (Default 1)\n\n");
+}
+
+int main(int argc, const char* argv[]) {
+ if (argc < 3) {
+ printUsage();
+ return EXIT_FAILURE;
+ }
+
+ std::unique_ptr<FILE, decltype(&fclose)> fInput1(fopen(argv[1], "rb"), &fclose);
+ if (fInput1.get() == NULL) {
+ printf("\nError: missing file %s\n", argv[1]);
+ return EXIT_FAILURE;
+ }
+ std::unique_ptr<FILE, decltype(&fclose)> fInput2(fopen(argv[2], "rb"), &fclose);
+ if (fInput2.get() == NULL) {
+ printf("\nError: missing file %s\n", argv[2]);
+ return EXIT_FAILURE;
+ }
+ int16_t enableCrossCorr = (4 == argc) ? atoi(argv[3]) : 1;
+
+ fseek(fInput1.get(), 0L, SEEK_END);
+ unsigned int fileSize1 = ftell(fInput1.get());
+ rewind(fInput1.get());
+ fseek(fInput2.get(), 0L, SEEK_END);
+ unsigned int fileSize2 = ftell(fInput2.get());
+ rewind(fInput2.get());
+ if (fileSize1 != fileSize2) {
+ printf("\nError: File sizes different\n");
+ return EXIT_FAILURE;
+ }
+
+ int numFrames = fileSize1 / sizeof(int16_t);
+ std::unique_ptr<int16_t[]> inBuffer1(new int16_t[numFrames]());
+ std::unique_ptr<int16_t[]> inBuffer2(new int16_t[numFrames]());
+
+ fread(inBuffer1.get(), sizeof(int16_t), numFrames, fInput1.get());
+ fread(inBuffer2.get(), sizeof(int16_t), numFrames, fInput2.get());
+
+ auto pairAutoCorr1 = correlation(inBuffer1.get(), inBuffer1.get(), numFrames, enableCrossCorr);
+ auto pairAutoCorr2 = correlation(inBuffer2.get(), inBuffer2.get(), numFrames, enableCrossCorr);
+
+ // Following code block checks pitch period difference between two input signals. They must
+ // match as AGC applies only gain, no frequency related computation is done.
+ bool pitchMatch = false;
+ for (unsigned i = 0; i < pairAutoCorr1.first.size() - 1; i++) {
+ if (pairAutoCorr1.first[i + 1] - pairAutoCorr1.first[i] !=
+ pairAutoCorr2.first[i + 1] - pairAutoCorr2.first[i]) {
+ pitchMatch = false;
+ break;
+ }
+ pitchMatch = true;
+ }
+ if (pitchMatch) {
+ printf("Auto-correlation : Pitch matched\n");
+ } else {
+ printf("Auto-correlation : Pitch mismatch\n");
+ return EXIT_FAILURE;
+ }
+
+ if (enableCrossCorr) {
+ auto pairCrossCorr =
+ correlation(inBuffer1.get(), inBuffer2.get(), numFrames, enableCrossCorr);
+
+ // Since AGC applies only gain, the pitch information obtained from cross correlation data
+ // of input and output is expected to be same as the input signal's pitch information.
+ pitchMatch = false;
+ for (unsigned i = 0; i < pairCrossCorr.first.size() - 1; i++) {
+ if (pairAutoCorr1.first[i + 1] - pairAutoCorr1.first[i] !=
+ pairCrossCorr.first[i + 1] - pairCrossCorr.first[i]) {
+ pitchMatch = false;
+ break;
+ }
+ pitchMatch = true;
+ }
+ if (pitchMatch) {
+ printf("Cross-correlation : Pitch matched for AGC\n");
+ if (pairAutoCorr1.second[0]) {
+ printf("Expected gain : (maxCrossCorr / maxAutoCorr1) = %f\n",
+ pairCrossCorr.second[0] / pairAutoCorr1.second[0]);
+ }
+ } else {
+ printf("Cross-correlation : Pitch mismatch\n");
+ return EXIT_FAILURE;
+ }
+ }
+
+ return EXIT_SUCCESS;
+}
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 8fe18de..4ac46b7 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1462,6 +1462,20 @@
}
}
+// Update downstream patches for all playback threads attached to an MSD module
+void AudioFlinger::updateDownStreamPatches_l(const struct audio_patch *patch,
+ const std::set<audio_io_handle_t> streams)
+{
+ for (const audio_io_handle_t stream : streams) {
+ PlaybackThread *playbackThread = checkPlaybackThread_l(stream);
+ if (playbackThread == nullptr || !playbackThread->isMsdDevice()) {
+ continue;
+ }
+ playbackThread->setDownStreamPatch(patch);
+ playbackThread->sendIoConfigEvent(AUDIO_OUTPUT_CONFIG_CHANGED);
+ }
+}
+
// Filter reserved keys from setParameters() before forwarding to audio HAL or acting upon.
// Some keys are used for audio routing and audio path configuration and should be reserved for use
// by audio policy and audio flinger for functional, privacy and security reasons.
@@ -2534,7 +2548,11 @@
*output, thread.get());
}
mPlaybackThreads.add(*output, thread);
- mPatchPanel.notifyStreamOpened(outHwDev, *output);
+ struct audio_patch patch;
+ mPatchPanel.notifyStreamOpened(outHwDev, *output, &patch);
+ if (thread->isMsdDevice()) {
+ thread->setDownStreamPatch(&patch);
+ }
return thread;
}
}
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index c47afd5..6d2afaa 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -320,6 +320,9 @@
status_t removeEffectFromHal(audio_port_handle_t deviceId,
audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect);
+ void updateDownStreamPatches_l(const struct audio_patch *patch,
+ const std::set<audio_io_handle_t> streams);
+
private:
// FIXME The 400 is temporarily too high until a leak of writers in media.log is fixed.
static const size_t kLogMemorySize = 400 * 1024;
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index b58fd8b..37aa13e 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -413,10 +413,10 @@
*handle = (audio_patch_handle_t) mAudioFlinger.nextUniqueId(AUDIO_UNIQUE_ID_USE_PATCH);
newPatch.mHalHandle = halHandle;
mAudioFlinger.mDeviceEffectManager.createAudioPatch(*handle, newPatch);
- mPatches.insert(std::make_pair(*handle, std::move(newPatch)));
if (insertedModule != AUDIO_MODULE_HANDLE_NONE) {
- addSoftwarePatchToInsertedModules(insertedModule, *handle);
+ addSoftwarePatchToInsertedModules(insertedModule, *handle, &newPatch.mAudioPatch);
}
+ mPatches.insert(std::make_pair(*handle, std::move(newPatch)));
} else {
newPatch.clearConnections(this);
}
@@ -781,10 +781,20 @@
}
void AudioFlinger::PatchPanel::notifyStreamOpened(
- AudioHwDevice *audioHwDevice, audio_io_handle_t stream)
+ AudioHwDevice *audioHwDevice, audio_io_handle_t stream, struct audio_patch *patch)
{
if (audioHwDevice->isInsert()) {
mInsertedModules[audioHwDevice->handle()].streams.insert(stream);
+ if (patch != nullptr) {
+ std::vector <SoftwarePatch> swPatches;
+ getDownstreamSoftwarePatches(stream, &swPatches);
+ if (swPatches.size() > 0) {
+ auto iter = mPatches.find(swPatches[0].getPatchHandle());
+ if (iter != mPatches.end()) {
+ *patch = iter->second.mAudioPatch;
+ }
+ }
+ }
}
}
@@ -813,9 +823,13 @@
}
void AudioFlinger::PatchPanel::addSoftwarePatchToInsertedModules(
- audio_module_handle_t module, audio_patch_handle_t handle)
+ audio_module_handle_t module, audio_patch_handle_t handle,
+ const struct audio_patch *patch)
{
mInsertedModules[module].sw_patches.insert(handle);
+ if (!mInsertedModules[module].streams.empty()) {
+ mAudioFlinger.updateDownStreamPatches_l(patch, mInsertedModules[module].streams);
+ }
}
void AudioFlinger::PatchPanel::removeSoftwarePatchFromInsertedModules(
diff --git a/services/audioflinger/PatchPanel.h b/services/audioflinger/PatchPanel.h
index 89d4eb1..ea38559 100644
--- a/services/audioflinger/PatchPanel.h
+++ b/services/audioflinger/PatchPanel.h
@@ -71,7 +71,8 @@
std::vector<SoftwarePatch> *patches) const;
// Notifies patch panel about all opened and closed streams.
- void notifyStreamOpened(AudioHwDevice *audioHwDevice, audio_io_handle_t stream);
+ void notifyStreamOpened(AudioHwDevice *audioHwDevice, audio_io_handle_t stream,
+ struct audio_patch *patch);
void notifyStreamClosed(audio_io_handle_t stream);
void dump(int fd) const;
@@ -226,7 +227,8 @@
AudioHwDevice* findAudioHwDeviceByModule(audio_module_handle_t module);
sp<DeviceHalInterface> findHwDeviceByModule(audio_module_handle_t module);
void addSoftwarePatchToInsertedModules(
- audio_module_handle_t module, audio_patch_handle_t handle);
+ audio_module_handle_t module, audio_patch_handle_t handle,
+ const struct audio_patch *patch);
void removeSoftwarePatchFromInsertedModules(audio_patch_handle_t handle);
void erasePatch(audio_patch_handle_t handle);
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 6b37fd0..927d87e 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1864,7 +1864,8 @@
// index 0 is reserved for normal mixer's submix
mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1),
mHwSupportsPause(false), mHwPaused(false), mFlushPending(false),
- mLeftVolFloat(-1.0), mRightVolFloat(-1.0)
+ mLeftVolFloat(-1.0), mRightVolFloat(-1.0),
+ mDownStreamPatch{}
{
snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id);
mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
@@ -2632,12 +2633,16 @@
ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event);
desc->mIoHandle = mId;
+ struct audio_patch patch = mPatch;
+ if (isMsdDevice()) {
+ patch = mDownStreamPatch;
+ }
switch (event) {
case AUDIO_OUTPUT_OPENED:
case AUDIO_OUTPUT_REGISTERED:
case AUDIO_OUTPUT_CONFIG_CHANGED:
- desc->mPatch = mPatch;
+ desc->mPatch = patch;
desc->mChannelMask = mChannelMask;
desc->mSamplingRate = mSampleRate;
desc->mFormat = mFormat;
@@ -2647,7 +2652,7 @@
desc->mLatency = latency_l();
break;
case AUDIO_CLIENT_STARTED:
- desc->mPatch = mPatch;
+ desc->mPatch = patch;
desc->mPortId = portId;
break;
case AUDIO_OUTPUT_CLOSED:
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 6b33ad5..709a3cc 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -940,6 +940,11 @@
&& outDeviceTypes().count(mTimestampCorrectedDevice) != 0;
}
+ void setDownStreamPatch(const struct audio_patch *patch) {
+ Mutex::Autolock _l(mLock);
+ mDownStreamPatch = *patch;
+ }
+
protected:
// updated by readOutputParameters_l()
size_t mNormalFrameCount; // normal mixer and effects
@@ -1218,6 +1223,10 @@
// volumes last sent to audio HAL with stream->setVolume()
float mLeftVolFloat;
float mRightVolFloat;
+
+ // audio patch used by the downstream software patch.
+ // Only used if ThreadBase::mIsMsdDevice is true.
+ struct audio_patch mDownStreamPatch;
};
class MixerThread : public PlaybackThread {
diff --git a/services/audiopolicy/common/managerdefinitions/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
index b5b10f3..9ba745a 100644
--- a/services/audiopolicy/common/managerdefinitions/include/HwModule.h
+++ b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
@@ -52,8 +52,12 @@
devices.merge(mDynamicDevices);
return devices;
}
+ std::string getTagForDevice(audio_devices_t device,
+ const String8 &address = String8(),
+ audio_format_t codec = AUDIO_FORMAT_DEFAULT);
void addDynamicDevice(const sp<DeviceDescriptor> &device)
{
+ device->setDynamic();
mDynamicDevices.add(device);
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index 8545f8e..f6859c7 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -49,11 +49,13 @@
{
}
+// Let DeviceDescriptorBase initialize the address since it handles specific cases like
+// legacy remote submix where "0" is added as default address.
DeviceDescriptor::DeviceDescriptor(const AudioDeviceTypeAddr &deviceTypeAddr,
const std::string &tagName,
const FormatVector &encodedFormats) :
DeviceDescriptorBase(deviceTypeAddr), mTagName(tagName), mEncodedFormats(encodedFormats),
- mDeclaredAddress(deviceTypeAddr.getAddress())
+ mDeclaredAddress(DeviceDescriptorBase::address())
{
mCurrentEncodedFormat = AUDIO_FORMAT_DEFAULT;
/* If framework runs against a pre 5.0 Audio HAL, encoded formats are absent from the config.
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index 2967014..3a143b0 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -41,6 +41,14 @@
}
}
+std::string HwModule::getTagForDevice(audio_devices_t device, const String8 &address,
+ audio_format_t codec)
+{
+ DeviceVector declaredDevices = getDeclaredDevices();
+ sp<DeviceDescriptor> deviceDesc = declaredDevices.getDevice(device, address, codec);
+ return deviceDesc ? deviceDesc->getTagName() : std::string{};
+}
+
status_t HwModule::addOutputProfile(const std::string& name, const audio_config_t *config,
audio_devices_t device, const String8& address)
{
@@ -49,7 +57,8 @@
profile->addAudioProfile(new AudioProfile(config->format, config->channel_mask,
config->sample_rate));
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device, "" /*tagName*/, address.string());
+ sp<DeviceDescriptor> devDesc =
+ new DeviceDescriptor(device, getTagForDevice(device), address.string());
addDynamicDevice(devDesc);
// Reciprocally attach the device to the module
devDesc->attach(this);
@@ -116,7 +125,8 @@
profile->addAudioProfile(new AudioProfile(config->format, config->channel_mask,
config->sample_rate));
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device, "" /*tagName*/, address.string());
+ sp<DeviceDescriptor> devDesc =
+ new DeviceDescriptor(device, getTagForDevice(device), address.string());
addDynamicDevice(devDesc);
// Reciprocally attach the device to the module
devDesc->attach(this);
diff --git a/services/audiopolicy/config/audio_policy_configuration_bluetooth_legacy_hal.xml b/services/audiopolicy/config/audio_policy_configuration_bluetooth_legacy_hal.xml
index b4cc1d3..5f4e5f2 100644
--- a/services/audiopolicy/config/audio_policy_configuration_bluetooth_legacy_hal.xml
+++ b/services/audiopolicy/config/audio_policy_configuration_bluetooth_legacy_hal.xml
@@ -185,6 +185,9 @@
<!-- Hearing aid Audio HAL -->
<xi:include href="hearing_aid_audio_policy_configuration.xml"/>
+ <!-- Le Audio Audio HAL -->
+ <xi:include href="le_audio_policy_configuration.xml"/>
+
<!-- MSD Audio HAL (optional) -->
<xi:include href="msd_audio_policy_configuration.xml"/>
diff --git a/services/audiopolicy/config/bluetooth_audio_policy_configuration.xml b/services/audiopolicy/config/bluetooth_audio_policy_configuration.xml
index ce78eb0..7238317 100644
--- a/services/audiopolicy/config/bluetooth_audio_policy_configuration.xml
+++ b/services/audiopolicy/config/bluetooth_audio_policy_configuration.xml
@@ -10,6 +10,12 @@
samplingRates="24000,16000"
channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
</mixPort>
+ <!-- Le Audio Audio Ports -->
+ <mixPort name="le audio output" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT,AUDIO_FORMAT_PCM_24_BIT,AUDIO_FORMAT_PCM_32_BIT"
+ samplingRates="8000,16000,24000,32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
</mixPorts>
<devicePorts>
<!-- A2DP Audio Ports -->
@@ -30,6 +36,9 @@
</devicePort>
<!-- Hearing AIDs Audio Ports -->
<devicePort tagName="BT Hearing Aid Out" type="AUDIO_DEVICE_OUT_HEARING_AID" role="sink"/>
+ <!-- BLE Audio Ports -->
+ <devicePort tagName="BLE Headset Out" type="AUDIO_DEVICE_OUT_BLE_HEADSET" role="sink"/>
+ <devicePort tagName="BLE Speaker Out" type="AUDIO_DEVICE_OUT_BLE_SPEAKER" role="sink"/>
</devicePorts>
<routes>
<route type="mix" sink="BT A2DP Out"
@@ -40,5 +49,9 @@
sources="a2dp output"/>
<route type="mix" sink="BT Hearing Aid Out"
sources="hearing aid output"/>
+ <route type="mix" sink="BLE Headset Out"
+ sources="le audio output"/>
+ <route type="mix" sink="BLE Speaker Out"
+ sources="le audio output"/>
</routes>
</module>
diff --git a/services/audiopolicy/config/le_audio_policy_configuration.xml b/services/audiopolicy/config/le_audio_policy_configuration.xml
new file mode 100644
index 0000000..a3dc72b
--- /dev/null
+++ b/services/audiopolicy/config/le_audio_policy_configuration.xml
@@ -0,0 +1,19 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Le Audio HAL Audio Policy Configuration file -->
+<module name="bluetooth" halVersion="2.1">
+ <mixPorts>
+ <mixPort name="le audio output" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT,AUDIO_FORMAT_PCM_24_BIT,AUDIO_FORMAT_PCM_32_BIT"
+ samplingRates="8000,16000,24000,32000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="BLE Headset Out" type="AUDIO_DEVICE_OUT_BLE_HEADSET" role="sink"/>
+ <devicePort tagName="BLE Speaker Out" type="AUDIO_DEVICE_OUT_BLE_SPEAKER" role="sink"/>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="BLE Headset Out" sources="le audio output"/>
+ <route type="mix" sink="BLE Speaker Out" sources="le audio output"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index a192083..7179355 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -1041,7 +1041,6 @@
if (*output != AUDIO_IO_HANDLE_NONE && setMsdPatch(device) == NO_ERROR) {
ALOGV("%s() Using MSD devices %s instead of devices %s",
__func__, msdDevices.toString().c_str(), outputDevices.toString().c_str());
- outputDevices = msdDevices;
} else {
*output = AUDIO_IO_HANDLE_NONE;
}
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index ca2164b..d379239 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -326,6 +326,7 @@
sp<DeviceDescriptor> mMsdOutputDevice;
sp<DeviceDescriptor> mMsdInputDevice;
+ sp<DeviceDescriptor> mDefaultOutputDevice;
};
void AudioPolicyManagerTestMsd::SetUpManagerConfig() {
@@ -380,17 +381,21 @@
primaryEncodedOutputProfile->addSupportedDevice(config.getDefaultOutputDevice());
config.getHwModules().getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY)->
addOutputProfile(primaryEncodedOutputProfile);
+
+ mDefaultOutputDevice = config.getDefaultOutputDevice();
}
void AudioPolicyManagerTestMsd::TearDown() {
mMsdOutputDevice.clear();
mMsdInputDevice.clear();
+ mDefaultOutputDevice.clear();
AudioPolicyManagerTest::TearDown();
}
TEST_F(AudioPolicyManagerTestMsd, InitSuccess) {
ASSERT_TRUE(mMsdOutputDevice);
ASSERT_TRUE(mMsdInputDevice);
+ ASSERT_TRUE(mDefaultOutputDevice);
}
TEST_F(AudioPolicyManagerTestMsd, Dump) {
@@ -409,7 +414,7 @@
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
- ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
+ ASSERT_EQ(selectedDeviceId, mDefaultOutputDevice->getId());
ASSERT_EQ(1, patchCount.deltaFromSnapshot());
}
@@ -418,7 +423,7 @@
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 48000);
- ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
+ ASSERT_EQ(selectedDeviceId, mDefaultOutputDevice->getId());
ASSERT_EQ(1, patchCount.deltaFromSnapshot());
}
@@ -427,11 +432,11 @@
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
- ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
+ ASSERT_EQ(selectedDeviceId, mDefaultOutputDevice->getId());
ASSERT_EQ(1, patchCount.deltaFromSnapshot());
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 48000);
- ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
+ ASSERT_EQ(selectedDeviceId, mDefaultOutputDevice->getId());
ASSERT_EQ(1, patchCount.deltaFromSnapshot());
}
@@ -453,7 +458,7 @@
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT,
nullptr /*output*/, &portId);
- ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
+ ASSERT_EQ(selectedDeviceId, mDefaultOutputDevice->getId());
ASSERT_EQ(1, patchCount.deltaFromSnapshot());
mManager->releaseOutput(portId);
ASSERT_EQ(1, patchCount.deltaFromSnapshot());
@@ -475,7 +480,7 @@
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
- ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
+ ASSERT_EQ(selectedDeviceId, mDefaultOutputDevice->getId());
ASSERT_EQ(0, patchCount.deltaFromSnapshot());
}
}