Fix underruns when fast track denied due to SRC
OpenSL ES requests a fast track. If sample rate conversion is needed,
the request is denied by server, and a larger client buffer is used
to handle the higher latency of a normal track. However the client
notification period was calculated based on buffer being divided into
2 sub-buffers. That resulted in the notification period being too long.
The server pulls chunks that are smaller than half the total buffer.
So now the client uses 3 sub-buffers when there is SRC.
Also removed the 'defer wake' optimization because it was incorrect.
This optimization attempted to reduce the number of wakeups of client,
when server releaseBuffer knows that another releaseBuffer will be
following. But there is no way for the first releaseBuffer to predict
how soon the second releaseBuffer will occur. In some cases it was
a long time, and the client underran. So now the client is woken up
immediately if the total number of available frames to client is >=
the minimum number the client wants to see (the notification period).
Also fix bug where minimum frame count was not being used in the
calculation of notification period.
Bug: 10342804
Change-Id: I3c246f4e7bc3684a344f2cf08268dc082e338e2a
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 744faee..15249a4 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -815,12 +815,29 @@
return NO_INIT;
}
+ // Not all of these values are needed under all conditions, but it is easier to get them all
+
uint32_t afLatency;
- if ((status = AudioSystem::getLatency(output, streamType, &afLatency)) != NO_ERROR) {
+ status = AudioSystem::getLatency(output, streamType, &afLatency);
+ if (status != NO_ERROR) {
ALOGE("getLatency(%d) failed status %d", output, status);
return NO_INIT;
}
+ size_t afFrameCount;
+ status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
+ if (status != NO_ERROR) {
+ ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);
+ return NO_INIT;
+ }
+
+ uint32_t afSampleRate;
+ status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);
+ if (status != NO_ERROR) {
+ ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status);
+ return NO_INIT;
+ }
+
// Client decides whether the track is TIMED (see below), but can only express a preference
// for FAST. Server will perform additional tests.
if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
@@ -836,6 +853,14 @@
}
ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
+ // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
+ // n = 1 fast track; nBuffering is ignored
+ // n = 2 normal track, no sample rate conversion
+ // n = 3 normal track, with sample rate conversion
+ // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering)
+ // n > 3 very high latency or very small notification interval; nBuffering is ignored
+ const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3;
+
mNotificationFramesAct = mNotificationFramesReq;
if (!audio_is_linear_pcm(format)) {
@@ -844,13 +869,6 @@
// Same comment as below about ignoring frameCount parameter for set()
frameCount = sharedBuffer->size();
} else if (frameCount == 0) {
- size_t afFrameCount;
- status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
- if (status != NO_ERROR) {
- ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType,
- status);
- return NO_INIT;
- }
frameCount = afFrameCount;
}
if (mNotificationFramesAct != frameCount) {
@@ -880,26 +898,13 @@
} else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
// FIXME move these calculations and associated checks to server
- uint32_t afSampleRate;
- status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);
- if (status != NO_ERROR) {
- ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType,
- status);
- return NO_INIT;
- }
- size_t afFrameCount;
- status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
- if (status != NO_ERROR) {
- ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);
- return NO_INIT;
- }
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d",
afFrameCount, minBufCount, afSampleRate, afLatency);
- if (minBufCount <= 2) {
- minBufCount = sampleRate == afSampleRate ? 2 : 3;
+ if (minBufCount <= nBuffering) {
+ minBufCount = nBuffering;
}
size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
@@ -909,18 +914,16 @@
if (frameCount == 0) {
frameCount = minFrameCount;
- }
- // Make sure that application is notified with sufficient margin
- // before underrun
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) {
- mNotificationFramesAct = frameCount/2;
- }
- if (frameCount < minFrameCount) {
+ } else if (frameCount < minFrameCount) {
// not ALOGW because it happens all the time when playing key clicks over A2DP
ALOGV("Minimum buffer size corrected from %d to %d",
frameCount, minFrameCount);
frameCount = minFrameCount;
}
+ // Make sure that application is notified with sufficient margin before underrun
+ if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
+ mNotificationFramesAct = frameCount/nBuffering;
+ }
} else {
// For fast tracks, the frame count calculations and checks are done by server
@@ -1001,8 +1004,8 @@
flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
mFlags = flags;
if (sharedBuffer == 0) {
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) {
- mNotificationFramesAct = frameCount/2;
+ if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
+ mNotificationFramesAct = frameCount/nBuffering;
}
}
}