audio flinger: fix sink metadata for telephony software patches
Make sure the proper sink metadata are sent to audio HAL on the capture
stream when a software audio patch is used for telephony between two
audio HAL modules.
Also fix a bug in audio policy manager where the HW audio source client
used for the telephnoy TX patch was specifying a capture source
instead of playback usage in its audio attributes.
Bug: 249808607
Test: disable LE Audio offload and place a call.
Change-Id: Ic9f7dc75a732e4e404fa0b200b6ee2b34f0690df
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index 45dd258..b54b41f 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -313,12 +313,19 @@
patch->sources[0].config_mask & AUDIO_PORT_CONFIG_FLAGS ?
patch->sources[0].flags.input : AUDIO_INPUT_FLAG_NONE;
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
+ audio_source_t source = AUDIO_SOURCE_MIC;
+ // For telephony patches, propagate voice communication use case to record side
+ if (patch->num_sources == 2
+ && patch->sources[1].ext.mix.usecase.stream
+ == AUDIO_STREAM_VOICE_CALL) {
+ source = AUDIO_SOURCE_VOICE_COMMUNICATION;
+ }
sp<ThreadBase> thread = mAudioFlinger.openInput_l(srcModule,
&input,
&config,
device,
address,
- AUDIO_SOURCE_MIC,
+ source,
flags,
outputDevice,
outputDeviceAddress);
@@ -516,9 +523,14 @@
audio_output_flags_t outputFlags = mAudioPatch.sinks[0].config_mask & AUDIO_PORT_CONFIG_FLAGS ?
mAudioPatch.sinks[0].flags.output : AUDIO_OUTPUT_FLAG_NONE;
audio_stream_type_t streamType = AUDIO_STREAM_PATCH;
+ audio_source_t source = AUDIO_SOURCE_DEFAULT;
if (mAudioPatch.num_sources == 2 && mAudioPatch.sources[1].type == AUDIO_PORT_TYPE_MIX) {
// "reuse one existing output mix" case
streamType = mAudioPatch.sources[1].ext.mix.usecase.stream;
+ // For telephony patches, propagate voice communication use case to record side
+ if (streamType == AUDIO_STREAM_VOICE_CALL) {
+ source = AUDIO_SOURCE_VOICE_COMMUNICATION;
+ }
}
if (mPlayback.thread()->hasFastMixer()) {
// Create a fast track if the playback thread has fast mixer to get better performance.
@@ -546,7 +558,8 @@
inChannelMask,
format,
frameCount,
- inputFlags);
+ inputFlags,
+ source);
} else {
// use a pseudo LCM between input and output framecount
int playbackShift = __builtin_ctz(playbackFrameCount);
@@ -566,7 +579,9 @@
frameCount,
nullptr,
(size_t)0 /* bufferSize */,
- inputFlags);
+ inputFlags,
+ {} /* timeout */,
+ source);
}
status = mRecord.checkTrack(tempRecordTrack.get());
if (status != NO_ERROR) {
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index e8552c4..daec57e 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -87,6 +87,10 @@
&& (flags & AUDIO_INPUT_FLAG_HW_AV_SYNC) == 0;
}
+ using SinkMetadatas = std::vector<record_track_metadata_v7_t>;
+ using MetadataInserter = std::back_insert_iterator<SinkMetadatas>;
+ virtual void copyMetadataTo(MetadataInserter& backInserter) const;
+
private:
friend class AudioFlinger; // for mState
@@ -134,7 +138,8 @@
void *buffer,
size_t bufferSize,
audio_input_flags_t flags,
- const Timeout& timeout = {});
+ const Timeout& timeout = {},
+ audio_source_t source = AUDIO_SOURCE_DEFAULT);
virtual ~PatchRecord();
virtual Source* getSource() { return nullptr; }
@@ -166,7 +171,8 @@
audio_channel_mask_t channelMask,
audio_format_t format,
size_t frameCount,
- audio_input_flags_t flags);
+ audio_input_flags_t flags,
+ audio_source_t source = AUDIO_SOURCE_DEFAULT);
Source* getSource() override { return static_cast<Source*>(this); }
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index bce7e25..9beaec6 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -8752,21 +8752,9 @@
return; // nothing to do
}
StreamInHalInterface::SinkMetadata metadata;
+ auto backInserter = std::back_inserter(metadata.tracks);
for (const sp<RecordTrack> &track : mActiveTracks) {
- // Do not forward PatchRecord metadata to audio HAL
- if (track->isPatchTrack()) {
- continue;
- }
- // No track is invalid as this is called after prepareTrack_l in the same critical section
- record_track_metadata_v7_t trackMetadata;
- trackMetadata.base = {
- .source = track->attributes().source,
- .gain = 1, // capture tracks do not have volumes
- };
- trackMetadata.channel_mask = track->channelMask(),
- strncpy(trackMetadata.tags, track->attributes().tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE);
-
- metadata.tracks.push_back(trackMetadata);
+ track->copyMetadataTo(backInserter);
}
mInput->stream->updateSinkMetadata(metadata);
}
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index ac8909f..2a77d22 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1477,7 +1477,7 @@
}
}
- metadata.channel_mask = mChannelMask,
+ metadata.channel_mask = mChannelMask;
strncpy(metadata.tags, mAttr.tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE);
*backInserter++ = metadata;
}
@@ -2021,7 +2021,6 @@
{
Buffer *pInBuffer;
Buffer inBuffer;
- bool outputBufferFull = false;
inBuffer.frameCount = frames;
inBuffer.raw = data;
@@ -2051,7 +2050,6 @@
ALOGV("%s(%d): thread %d no more output buffers; status %d",
__func__, mId,
(int)mThreadIoHandle, status);
- outputBufferFull = true;
break;
}
uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
@@ -2747,6 +2745,25 @@
}
}
+void AudioFlinger::RecordThread::RecordTrack::copyMetadataTo(MetadataInserter& backInserter) const
+{
+
+ // Do not forward PatchRecord metadata with unspecified audio source
+ if (mAttr.source == AUDIO_SOURCE_DEFAULT) {
+ return;
+ }
+
+ // No track is invalid as this is called after prepareTrack_l in the same critical section
+ record_track_metadata_v7_t metadata;
+ metadata.base = {
+ .source = mAttr.source,
+ .gain = 1, // capture tracks do not have volumes
+ };
+ metadata.channel_mask = mChannelMask;
+ strncpy(metadata.tags, mAttr.tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE);
+
+ *backInserter++ = metadata;
+}
// ----------------------------------------------------------------------------
#undef LOG_TAG
@@ -2760,9 +2777,10 @@
void *buffer,
size_t bufferSize,
audio_input_flags_t flags,
- const Timeout& timeout)
+ const Timeout& timeout,
+ audio_source_t source)
: RecordTrack(recordThread, NULL,
- audio_attributes_t{} /* currently unused for patch track */,
+ audio_attributes_t{ .source = source } ,
sampleRate, format, channelMask, frameCount,
buffer, bufferSize, AUDIO_SESSION_NONE, getpid(),
audioServerAttributionSource(getpid()), flags, TYPE_PATCH),
@@ -2873,9 +2891,10 @@
audio_channel_mask_t channelMask,
audio_format_t format,
size_t frameCount,
- audio_input_flags_t flags)
+ audio_input_flags_t flags,
+ audio_source_t source)
: PatchRecord(recordThread, sampleRate, channelMask, format, frameCount,
- nullptr /*buffer*/, 0 /*bufferSize*/, flags),
+ nullptr /*buffer*/, 0 /*bufferSize*/, flags, {} /* timeout */, source),
mPatchRecordAudioBufferProvider(*this),
mSinkBuffer(allocAligned(32, mFrameCount * mFrameSize)),
mStubBuffer(allocAligned(32, mFrameCount * mFrameSize))
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index f9415fe..49a6159 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -788,7 +788,8 @@
ALOGV("%s between source %s and sink %s", __func__,
srcDevice->toString().c_str(), sinkDevice->toString().c_str());
auto callTxSourceClientPortId = PolicyAudioPort::getNextUniqueId();
- const audio_attributes_t aa = { .source = AUDIO_SOURCE_VOICE_COMMUNICATION };
+ const auto aa = mEngine->getAttributesForStreamType(AUDIO_STREAM_VOICE_CALL);
+
struct audio_port_config source = {};
srcDevice->toAudioPortConfig(&source);
mCallTxSourceClient = new InternalSourceClientDescriptor(
@@ -4534,7 +4535,7 @@
// In case of Hw bridge, it is a Work Around. The mixPort used is the one declared
// in config XML to reach the sink so that is can be declared as available.
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- sp<SwAudioOutputDescriptor> outputDesc = nullptr;
+ sp<SwAudioOutputDescriptor> outputDesc;
if (!sourceDesc->isInternal()) {
// take care of dynamic routing for SwOutput selection,
audio_attributes_t attributes = sourceDesc->attributes();
@@ -4604,7 +4605,8 @@
audio_port_config srcMixPortConfig = {};
outputDesc->toAudioPortConfig(&srcMixPortConfig, nullptr);
// for volume control, we may need a valid stream
- srcMixPortConfig.ext.mix.usecase.stream = !sourceDesc->isInternal() ?
+ srcMixPortConfig.ext.mix.usecase.stream =
+ (!sourceDesc->isInternal() || isCallTxAudioSource(sourceDesc)) ?
mEngine->getStreamTypeForAttributes(sourceDesc->attributes()) :
AUDIO_STREAM_PATCH;
patchBuilder.addSource(srcMixPortConfig);
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 87e6974..2c1db79 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -639,6 +639,10 @@
return mCallRxSourceClient != nullptr && source == mCallRxSourceClient;
}
+ bool isCallTxAudioSource(const sp<SourceClientDescriptor> &source) {
+ return mCallTxSourceClient != nullptr && source == mCallTxSourceClient;
+ }
+
void connectTelephonyRxAudioSource();
void disconnectTelephonyAudioSource(sp<SourceClientDescriptor> &clientDesc);