Add secondary output to audio tracks
The secondary are returned from mixes (from DAP loopback&render),
from getOutputForAttr.
All getOutputForAttr* of the stack are update.
Internal getOutputForAttr use descriptor, external one use handles.
The secondary output are saved in each track and the track is
invalidated if the list of secondary output changes.
In audio flinger, create a pair of recordTrack & patchTrack to pipe
the intercepted audio audio to the secondary output.
Test: adb shell audiorecorder --target /data/file.raw
Bug: 111453086
Change-Id: Id6523d9e383c15a0e39313d5f355df809b7e72fe
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index bc99099..befabc5 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -292,13 +292,16 @@
fullConfig.sample_rate = config->sample_rate;
fullConfig.channel_mask = config->channel_mask;
fullConfig.format = config->format;
+ std::vector<audio_io_handle_t> secondaryOutputs;
ret = AudioSystem::getOutputForAttr(attr, &io,
actualSessionId,
&streamType, client.clientPid, client.clientUid,
&fullConfig,
(audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ |
AUDIO_OUTPUT_FLAG_DIRECT),
- deviceId, &portId);
+ deviceId, &portId, &secondaryOutputs);
+ ALOGW_IF(!secondaryOutputs.empty(),
+ "%s does not support secondary outputs, ignoring them", __func__);
} else {
ret = AudioSystem::getInputForAttr(attr, &io,
actualSessionId,
@@ -678,6 +681,7 @@
status_t lStatus;
audio_stream_type_t streamType;
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+ std::vector<audio_io_handle_t> secondaryOutputs;
bool updatePid = (input.clientInfo.clientPid == -1);
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
@@ -712,7 +716,7 @@
lStatus = AudioSystem::getOutputForAttr(&input.attr, &output.outputId, sessionId, &streamType,
clientPid, clientUid, &input.config, input.flags,
- &output.selectedDeviceId, &portId);
+ &output.selectedDeviceId, &portId, &secondaryOutputs);
if (lStatus != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
ALOGE("createTrack() getOutputForAttr() return error %d or invalid output handle", lStatus);
@@ -785,6 +789,61 @@
output.afLatencyMs = thread->latency();
output.portId = portId;
+ if (lStatus == NO_ERROR) {
+ // Connect secondary outputs. Failure on a secondary output must not imped the primary
+ // Any secondary output setup failure will lead to a desync between the AP and AF until
+ // the track is destroyed.
+ TeePatches teePatches;
+ for (audio_io_handle_t secondaryOutput : secondaryOutputs) {
+ PlaybackThread *secondaryThread = checkPlaybackThread_l(secondaryOutput);
+ if (secondaryThread == NULL) {
+ ALOGE("no playback thread found for secondary output %d", output.outputId);
+ continue;
+ }
+
+ size_t frameCount = std::lcm(thread->frameCount(), secondaryThread->frameCount());
+
+ using namespace std::chrono_literals;
+ auto inChannelMask = audio_channel_mask_out_to_in(input.config.channel_mask);
+ sp patchRecord = new RecordThread::PatchRecord(nullptr /* thread */,
+ output.sampleRate,
+ inChannelMask,
+ input.config.format,
+ frameCount,
+ NULL /* buffer */,
+ (size_t)0 /* bufferSize */,
+ AUDIO_INPUT_FLAG_DIRECT,
+ 0ns /* timeout */);
+ status_t status = patchRecord->initCheck();
+ if (status != NO_ERROR) {
+ ALOGE("Secondary output patchRecord init failed: %d", status);
+ continue;
+ }
+ sp patchTrack = new PlaybackThread::PatchTrack(secondaryThread,
+ streamType,
+ output.sampleRate,
+ input.config.channel_mask,
+ input.config.format,
+ frameCount,
+ patchRecord->buffer(),
+ patchRecord->bufferSize(),
+ output.flags,
+ 0ns /* timeout */);
+ status = patchTrack->initCheck();
+ if (status != NO_ERROR) {
+ ALOGE("Secondary output patchTrack init failed: %d", status);
+ continue;
+ }
+ teePatches.push_back({patchRecord, patchTrack});
+ secondaryThread->addPatchTrack(patchTrack);
+ patchTrack->setPeerProxy(patchRecord.get());
+ patchRecord->setPeerProxy(patchTrack.get());
+
+ patchTrack->start(); // patchRecord is NOT started as it has no thread
+ }
+ track->setTeePatches(std::move(teePatches));
+ }
+
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
if (lStatus == NO_ERROR && effectThread != NULL) {
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index f16a196..1441e15 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -24,6 +24,7 @@
#include <chrono>
#include <deque>
#include <map>
+#include <numeric>
#include <optional>
#include <set>
#include <string>
@@ -528,6 +529,9 @@
class EffectChain;
struct AudioStreamIn;
+ struct TeePatch;
+ using TeePatches = std::vector<TeePatch>;
+
struct stream_type_t {
stream_type_t()
@@ -727,6 +731,11 @@
audioHwDev(dev), stream(in), flags(flags) {}
};
+ struct TeePatch {
+ sp<RecordThread::PatchRecord> patchRecord;
+ sp<PlaybackThread::PatchTrack> patchTrack;
+ };
+
// for mAudioSessionRefs only
struct AudioSessionRef {
AudioSessionRef(audio_session_t sessionid, pid_t pid) :
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 33048fc..4683525 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -43,9 +43,8 @@
void appendDumpHeader(String8& result);
void appendDump(String8& result, bool active);
- virtual status_t start(AudioSystem::sync_event_t event =
- AudioSystem::SYNC_EVENT_NONE,
- audio_session_t triggerSession = AUDIO_SESSION_NONE);
+ virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
+ audio_session_t triggerSession = AUDIO_SESSION_NONE);
virtual void stop();
void pause();
@@ -129,6 +128,8 @@
}
sp<os::ExternalVibration> getExternalVibration() const { return mExternalVibration; }
+ void setTeePatches(TeePatches teePatches);
+
protected:
// for numerous
friend class PlaybackThread;
@@ -139,8 +140,8 @@
DISALLOW_COPY_AND_ASSIGN(Track);
// AudioBufferProvider interface
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
- // releaseBuffer() not overridden
+ status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) override;
+ void releaseBuffer(AudioBufferProvider::Buffer* buffer) override;
// ExtendedAudioBufferProvider interface
virtual size_t framesReady() const;
@@ -220,6 +221,8 @@
sp<os::ExternalVibration> mExternalVibration;
private:
+ void interceptBuffer(const AudioBufferProvider::Buffer& buffer);
+
// The following fields are only for fast tracks, and should be in a subclass
int mFastIndex; // index within FastMixerState::mFastTracks[];
// either mFastIndex == -1 if not isFastTrack()
@@ -239,6 +242,7 @@
audio_output_flags_t mFlags;
// If the last track change was notified to the client with readAndClearHasChanged
std::atomic_flag mChangeNotified = ATOMIC_FLAG_INIT;
+ TeePatches mTeePatches;
}; // end of Track
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 5a70864..dd1eabf 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -8464,6 +8464,7 @@
audio_output_flags_t flags =
(audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ | AUDIO_OUTPUT_FLAG_DIRECT);
audio_port_handle_t deviceId = mDeviceId;
+ std::vector<audio_io_handle_t> secondaryOutputs;
ret = AudioSystem::getOutputForAttr(&mAttr, &io,
mSessionId,
&stream,
@@ -8472,7 +8473,10 @@
&config,
flags,
&deviceId,
- &portId);
+ &portId,
+ &secondaryOutputs);
+ ALOGD_IF(!secondaryOutputs.empty(),
+ "MmapThread::start does not support secondary outputs, ignoring them");
} else {
audio_config_base_t config;
config.sample_rate = mSampleRate;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 863dc9e..37c3a2d 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -99,7 +99,7 @@
mId(android_atomic_inc(&nextTrackId)),
mTerminated(false),
mType(type),
- mThreadIoHandle(thread->id()),
+ mThreadIoHandle(thread ? thread->id() : AUDIO_IO_HANDLE_NONE),
mPortId(portId),
mIsInvalid(false)
{
@@ -670,8 +670,7 @@
}
// AudioBufferProvider interface
-status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
- AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
ServerProxy::Buffer buf;
size_t desiredFrames = buffer->frameCount;
@@ -686,10 +685,39 @@
} else {
mAudioTrackServerProxy->tallyUnderrunFrames(0);
}
-
return status;
}
+void AudioFlinger::PlaybackThread::Track::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ interceptBuffer(*buffer);
+ TrackBase::releaseBuffer(buffer);
+}
+
+// TODO: compensate for time shift between HW modules.
+void AudioFlinger::PlaybackThread::Track::interceptBuffer(
+ const AudioBufferProvider::Buffer& buffer) {
+ for (auto& sink : mTeePatches) {
+ RecordThread::PatchRecord& patchRecord = *sink.patchRecord;
+ AudioBufferProvider::Buffer patchBuffer;
+ patchBuffer.frameCount = buffer.frameCount;
+ auto status = patchRecord.getNextBuffer(&patchBuffer);
+ if (status != NO_ERROR) {
+ ALOGW("%s PathRecord getNextBuffer failed with error %d: %s",
+ __func__, status, strerror(-status));
+ continue;
+ }
+ // FIXME: On buffer wrap, the frame count will be less then requested,
+ // retry to write the rest. (unlikely due to lcm buffer sizing)
+ ALOGW_IF(patchBuffer.frameCount != buffer.frameCount,
+ "%s PatchRecord can not provide big enough buffer %zu/%zu, dropping %zu frames",
+ __func__, patchBuffer.frameCount, buffer.frameCount,
+ buffer.frameCount - patchBuffer.frameCount);
+ memcpy(patchBuffer.raw, buffer.raw, patchBuffer.frameCount * mFrameSize);
+ patchRecord.releaseBuffer(&patchBuffer);
+ }
+}
+
// releaseBuffer() is not overridden
// ExtendedAudioBufferProvider interface
@@ -1081,6 +1109,10 @@
};
}
+void AudioFlinger::PlaybackThread::Track::setTeePatches(TeePatches teePatches) {
+ mTeePatches = std::move(teePatches);
+}
+
status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
{
if (!isOffloaded() && !isDirect()) {