Merge "[media][sfplugin] fix -Wdangling-gsl"
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 2d769cf..a021866 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -2953,9 +2953,11 @@
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setWriteBlocked(mWriteAckSequence);
}
+ ATRACE_BEGIN("write");
// FIXME We should have an implementation of timestamps for direct output threads.
// They are used e.g for multichannel PCM playback over HDMI.
bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining);
+ ATRACE_END();
if (mUseAsyncWrite &&
((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
@@ -5646,10 +5648,17 @@
minFrames = 1;
}
- if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() &&
+ const size_t framesReady = track->framesReady();
+ const int trackId = track->id();
+ if (ATRACE_ENABLED()) {
+ std::string traceName("nRdy");
+ traceName += std::to_string(trackId);
+ ATRACE_INT(traceName.c_str(), framesReady);
+ }
+ if ((framesReady >= minFrames) && track->isReady() && !track->isPaused() &&
!track->isStopping_2() && !track->isStopped())
{
- ALOGVV("track(%d) s=%08x [OK]", track->id(), cblk->mServer);
+ ALOGVV("track(%d) s=%08x [OK]", trackId, cblk->mServer);
if (track->mFillingUpStatus == Track::FS_FILLED) {
track->mFillingUpStatus = Track::FS_ACTIVE;
@@ -5726,7 +5735,7 @@
// fill a buffer, then remove it from active list.
// Only consider last track started for mixer state control
if (--(track->mRetryCount) <= 0) {
- ALOGV("BUFFER TIMEOUT: remove track(%d) from active list", track->id());
+ ALOGV("BUFFER TIMEOUT: remove track(%d) from active list", trackId);
tracksToRemove->add(track);
// indicate to client process that the track was disabled because of underrun;
// it will then automatically call start() when data is available
@@ -5734,7 +5743,7 @@
} else if (last) {
ALOGW("pause because of UNDERRUN, framesReady = %zu,"
"minFrames = %u, mFormat = %#x",
- track->framesReady(), minFrames, mFormat);
+ framesReady, minFrames, mFormat);
mixerStatus = MIXER_TRACKS_ENABLED;
if (mHwSupportsPause && !mHwPaused && !mStandby) {
doHwPause = true;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 932c32b..7c53ca0 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -2414,7 +2414,6 @@
if (!stream) return NO_INIT; // If there is no stream, RecordThread is not reading.
status_t result = NO_ERROR;
- struct timespec newTimeOut = *timeOut;
size_t bytesRead = 0;
{
ATRACE_NAME("read");
@@ -2437,15 +2436,16 @@
ALOGW_IF(buffer->mFrameCount < bytesRead / mFrameSize,
"Lost %zu frames obtained from HAL", bytesRead / mFrameSize - buffer->mFrameCount);
mUnconsumedFrames = buffer->mFrameCount;
- // Correct newTimeOut by elapsed time.
+ struct timespec newTimeOut;
if (startTimeNs) {
- nsecs_t newTimeOutNs =
- audio_utils_ns_from_timespec(&newTimeOut) - (systemTime() - startTimeNs);
+ // Correct the timeout by elapsed time.
+ nsecs_t newTimeOutNs = audio_utils_ns_from_timespec(timeOut) - (systemTime() - startTimeNs);
if (newTimeOutNs < 0) newTimeOutNs = 0;
newTimeOut.tv_sec = newTimeOutNs / NANOS_PER_SECOND;
newTimeOut.tv_nsec = newTimeOutNs - newTimeOut.tv_sec * NANOS_PER_SECOND;
+ timeOut = &newTimeOut;
}
- return PatchRecord::obtainBuffer(buffer, &newTimeOut);
+ return PatchRecord::obtainBuffer(buffer, timeOut);
stream_error:
stream->standby();