AudioTrack: Advance timestamp time during pause
Update MediaPlayer code to match.
This will change the behavior on the AudioTrack Java side,
but is within the API and should make timestamp handling easier.
Test: Photos app with pause and play.
Bug: 30502030
Change-Id: I0e9129ce6af5b920ad71e63ad0c205de7561acf2
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index d45dbcd..f619e1d 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1521,57 +1521,45 @@
}
uint32_t numFramesPlayed;
- int64_t numFramesPlayedAt;
+ int64_t numFramesPlayedAtUs;
AudioTimestamp ts;
- static const int64_t kStaleTimestamp100ms = 100000;
status_t res = mTrack->getTimestamp(ts);
if (res == OK) { // case 1: mixing audio tracks and offloaded tracks.
numFramesPlayed = ts.mPosition;
- numFramesPlayedAt = ts.mTime.tv_sec * 1000000LL + ts.mTime.tv_nsec / 1000;
- const int64_t timestampAge = nowUs - numFramesPlayedAt;
- if (timestampAge > kStaleTimestamp100ms) {
- // This is an audio FIXME.
- // getTimestamp returns a timestamp which may come from audio mixing threads.
- // After pausing, the MixerThread may go idle, thus the mTime estimate may
- // become stale. Assuming that the MixerThread runs 20ms, with FastMixer at 5ms,
- // the max latency should be about 25ms with an average around 12ms (to be verified).
- // For safety we use 100ms.
- ALOGV("getTimestamp: returned stale timestamp nowUs(%lld) numFramesPlayedAt(%lld)",
- (long long)nowUs, (long long)numFramesPlayedAt);
- numFramesPlayedAt = nowUs - kStaleTimestamp100ms;
- }
- //ALOGD("getTimestamp: OK %d %lld", numFramesPlayed, (long long)numFramesPlayedAt);
+ numFramesPlayedAtUs = ts.mTime.tv_sec * 1000000LL + ts.mTime.tv_nsec / 1000;
+ //ALOGD("getTimestamp: OK %d %lld", numFramesPlayed, (long long)numFramesPlayedAtUs);
} else if (res == WOULD_BLOCK) { // case 2: transitory state on start of a new track
numFramesPlayed = 0;
- numFramesPlayedAt = nowUs;
+ numFramesPlayedAtUs = nowUs;
//ALOGD("getTimestamp: WOULD_BLOCK %d %lld",
- // numFramesPlayed, (long long)numFramesPlayedAt);
+ // numFramesPlayed, (long long)numFramesPlayedAtUs);
} else { // case 3: transitory at new track or audio fast tracks.
res = mTrack->getPosition(&numFramesPlayed);
CHECK_EQ(res, (status_t)OK);
- numFramesPlayedAt = nowUs;
- numFramesPlayedAt += 1000LL * mTrack->latency() / 2; /* XXX */
- //ALOGD("getPosition: %u %lld", numFramesPlayed, (long long)numFramesPlayedAt);
+ numFramesPlayedAtUs = nowUs;
+ numFramesPlayedAtUs += 1000LL * mTrack->latency() / 2; /* XXX */
+ //ALOGD("getPosition: %u %lld", numFramesPlayed, (long long)numFramesPlayedAtUs);
}
// CHECK_EQ(numFramesPlayed & (1 << 31), 0); // can't be negative until 12.4 hrs, test
// TODO: remove the (int32_t) casting below as it may overflow at 12.4 hours.
int64_t durationUs = (int64_t)((int32_t)numFramesPlayed * 1000000LL / mSampleRateHz)
- + nowUs - numFramesPlayedAt;
+ + nowUs - numFramesPlayedAtUs;
if (durationUs < 0) {
// Occurs when numFramesPlayed position is very small and the following:
// (1) In case 1, the time nowUs is computed before getTimestamp() is called and
- // numFramesPlayedAt is greater than nowUs by time more than numFramesPlayed.
+ // numFramesPlayedAtUs is greater than nowUs by time more than numFramesPlayed.
// (2) In case 3, using getPosition and adding mAudioSink->latency() to
- // numFramesPlayedAt, by a time amount greater than numFramesPlayed.
+ // numFramesPlayedAtUs, by a time amount greater than numFramesPlayed.
//
// Both of these are transitory conditions.
ALOGV("getPlayedOutDurationUs: negative duration %lld set to zero", (long long)durationUs);
durationUs = 0;
}
ALOGV("getPlayedOutDurationUs(%lld) nowUs(%lld) frames(%u) framesAt(%lld)",
- (long long)durationUs, (long long)nowUs, numFramesPlayed, (long long)numFramesPlayedAt);
+ (long long)durationUs, (long long)nowUs,
+ numFramesPlayed, (long long)numFramesPlayedAtUs);
return durationUs;
}