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am 45cb3cfa: Merge "Handle large audio lost" into gingerbread

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Merge commit '45cb3cfacf3b07ae3c5f15eba9bbbcd060cee261' into gingerbread-plus-aosp

* commit '45cb3cfacf3b07ae3c5f15eba9bbbcd060cee261':
  Handle large audio lost
This commit is contained in:
James Dong
2010-08-12 10:12:28 -07:00
committed by Android Git Automerger
parent f4e26da1af 45cb3cfacf
commit 8aa244eab7
2 changed files with 19 additions and 10 deletions
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3
include/media/stagefright/AudioSource.h
View File

@@ -60,7 +60,8 @@ private:
int64_t mStartTimeUs;
int16_t mMaxAmplitude;
int64_t mPrevSampleTimeUs;
int64_t mNumLostFrames;
int64_t mTotalLostFrames;
int64_t mPrevLostBytes;
MediaBufferGroup *mGroup;

26
media/libstagefright/AudioSource.cpp
View File

@@ -35,7 +35,8 @@ AudioSource::AudioSource(
: mStarted(false),
mCollectStats(false),
mPrevSampleTimeUs(0),
mNumLostFrames(0),
mTotalLostFrames(0),
mPrevLostBytes(0),
mGroup(NULL) {
LOGV("sampleRate: %d, channels: %d", sampleRate, channels);
@@ -108,7 +109,8 @@ status_t AudioSource::stop() {
mStarted = false;
if (mCollectStats) {
LOGI("Total lost audio frames: %lld", mNumLostFrames);
LOGI("Total lost audio frames: %lld",
mTotalLostFrames + (mPrevLostBytes >> 1));
}
return OK;
@@ -186,10 +188,11 @@ status_t AudioSource::read(
// Insert null frames when lost frames are detected.
int64_t timestampUs = mPrevSampleTimeUs;
uint32_t numLostBytes = mRecord->getInputFramesLost() << 1;
numLostBytes += mPrevLostBytes;
#if 0
// Simulate lost frames
numLostBytes = ((rand() * 1.0 / RAND_MAX)) * kMaxBufferSize;
numLostBytes &= 0xFFFFFFFE; // Alignment request
numLostBytes = ((rand() * 1.0 / RAND_MAX)) * 2 * kMaxBufferSize;
numLostBytes &= 0xFFFFFFFE; // Alignment requirement
// Reduce the chance to lose
if (rand() * 1.0 / RAND_MAX >= 0.05) {
@@ -197,13 +200,18 @@ status_t AudioSource::read(
}
#endif
if (numLostBytes > 0) {
// Not expect too many lost frames!
CHECK(numLostBytes <= kMaxBufferSize);
if (numLostBytes > kMaxBufferSize) {
mPrevLostBytes = numLostBytes - kMaxBufferSize;
numLostBytes = kMaxBufferSize;
}
CHECK_EQ(numLostBytes & 1, 0);
timestampUs += ((1000000LL * (numLostBytes >> 1)) +
(sampleRate >> 1)) / sampleRate;
timestampUs += (1000000LL * numLostBytes >> 1) / sampleRate;
CHECK(timestampUs > mPrevSampleTimeUs);
if (mCollectStats) {
mNumLostFrames += (numLostBytes >> 1);
mTotalLostFrames += (numLostBytes >> 1);
}
if ((err = skipFrame(timestampUs, options)) == -1) {
buffer->release();
@@ -240,7 +248,7 @@ status_t AudioSource::read(
buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
CHECK(timestampUs > mPrevSampleTimeUs);
if (mNumLostFrames == 0) {
if (mTotalLostFrames == 0) {
CHECK_EQ(mPrevSampleTimeUs,
mStartTimeUs + (1000000LL * numFramesRecorded) / sampleRate);
}