Merge "AudioPlaybackCapture: solve second output data consumed low issue" into main
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index c8b0aa1..adff1b5 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -3922,7 +3922,8 @@
patchRecord->bufferSize(),
outputFlags,
0ns /* timeout */,
- frameCountToBeReady);
+ frameCountToBeReady,
+ track->getSpeed());
status = patchTrack->initCheck();
if (status != NO_ERROR) {
ALOGE("Secondary output patchTrack init failed: %d", status);
diff --git a/services/audioflinger/IAfTrack.h b/services/audioflinger/IAfTrack.h
index 2302e13..0d8c01e 100644
--- a/services/audioflinger/IAfTrack.h
+++ b/services/audioflinger/IAfTrack.h
@@ -571,10 +571,11 @@
size_t bufferSize,
audio_output_flags_t flags,
const Timeout& timeout = {},
- size_t frameCountToBeReady = 1 /** Default behaviour is to start
+ size_t frameCountToBeReady = 1, /** Default behaviour is to start
* as soon as possible to have
* the lowest possible latency
- * even if it might glitch. */);
+ * even if it might glitch. */
+ float speed = 1.0f);
};
class IAfPatchRecord : public virtual IAfRecordTrack, public virtual IAfPatchTrackBase {
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index 4333cc8..f57470f 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -646,7 +646,8 @@
tempRecordTrack->bufferSize(),
outputFlags,
{} /*timeout*/,
- frameCountToBeReady);
+ frameCountToBeReady,
+ 1.0f);
status = mPlayback.checkTrack(tempPatchTrack.get());
if (status != NO_ERROR) {
return status;
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index b4cb805..15c786e 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -490,10 +490,11 @@
size_t bufferSize,
audio_output_flags_t flags,
const Timeout& timeout = {},
- size_t frameCountToBeReady = 1 /** Default behaviour is to start
+ size_t frameCountToBeReady = 1, /** Default behaviour is to start
* as soon as possible to have
* the lowest possible latency
- * even if it might glitch. */);
+ * even if it might glitch. */
+ float speed = 1.0f);
~PatchTrack() override;
size_t framesReady() const final;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 77abaf6..2c7af7b 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -2441,10 +2441,11 @@
size_t bufferSize,
audio_output_flags_t flags,
const Timeout& timeout,
- size_t frameCountToBeReady /** Default behaviour is to start
+ size_t frameCountToBeReady, /** Default behaviour is to start
* as soon as possible to have
* the lowest possible latency
- * even if it might glitch. */)
+ * even if it might glitch. */
+ float speed)
{
return sp<PatchTrack>::make(
playbackThread,
@@ -2457,7 +2458,8 @@
bufferSize,
flags,
timeout,
- frameCountToBeReady);
+ frameCountToBeReady,
+ speed);
}
PatchTrack::PatchTrack(IAfPlaybackThread* playbackThread,
@@ -2470,17 +2472,26 @@
size_t bufferSize,
audio_output_flags_t flags,
const Timeout& timeout,
- size_t frameCountToBeReady)
+ size_t frameCountToBeReady,
+ float speed)
: Track(playbackThread, NULL, streamType,
audio_attributes_t{} /* currently unused for patch track */,
sampleRate, format, channelMask, frameCount,
buffer, bufferSize, nullptr /* sharedBuffer */,
AUDIO_SESSION_NONE, getpid(), audioServerAttributionSource(getpid()), flags,
- TYPE_PATCH, AUDIO_PORT_HANDLE_NONE, frameCountToBeReady),
- PatchTrackBase(mCblk ? new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true)
- : nullptr,
+ TYPE_PATCH, AUDIO_PORT_HANDLE_NONE, frameCountToBeReady, speed),
+ PatchTrackBase(mCblk ? new AudioTrackClientProxy(mCblk, mBuffer, frameCount, mFrameSize,
+ true /*clientInServer*/) : nullptr,
playbackThread, timeout)
{
+ if (mProxy != nullptr) {
+ sp<AudioTrackClientProxy>::cast(mProxy)->setPlaybackRate({
+ /* .mSpeed = */ speed,
+ /* .mPitch = */ AUDIO_TIMESTRETCH_PITCH_NORMAL,
+ /* .mStretchMode = */ AUDIO_TIMESTRETCH_STRETCH_DEFAULT,
+ /* .mFallbackMode = */ AUDIO_TIMESTRETCH_FALLBACK_FAIL
+ });
+ }
ALOGV("%s(%d): sampleRate %d mPeerTimeout %d.%03d sec",
__func__, mId, sampleRate,
(int)mPeerTimeout.tv_sec,