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am 23796e3b: Merge "Add lost frame handling in AudioSource" into gingerbread

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Merge commit '23796e3b622c6be4feb15d4d57e58645acfa431b' into gingerbread-plus-aosp

* commit '23796e3b622c6be4feb15d4d57e58645acfa431b':
  Add lost frame handling in AudioSource
This commit is contained in:
James Dong
2010-08-02 18:15:48 -07:00
committed by Android Git Automerger
parent fa32a45af7 23796e3b62
commit 51a260d8c6
2 changed files with 97 additions and 51 deletions
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5
include/media/stagefright/AudioSource.h
View File

@@ -57,11 +57,10 @@ private:
bool mCollectStats;
bool mTrackMaxAmplitude;
int64_t mTotalReadTimeUs;
int64_t mTotalReadBytes;
int64_t mTotalReads;
int64_t mStartTimeUs;
int16_t mMaxAmplitude;
int64_t mPrevSampleTimeUs;
int64_t mNumLostFrames;
MediaBufferGroup *mGroup;

143
media/libstagefright/AudioSource.cpp
View File

@@ -26,8 +26,7 @@
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <cutils/properties.h>
#include <sys/time.h>
#include <time.h>
#include <stdlib.h>
namespace android {
@@ -35,9 +34,8 @@ AudioSource::AudioSource(
int inputSource, uint32_t sampleRate, uint32_t channels)
: mStarted(false),
mCollectStats(false),
mTotalReadTimeUs(0),
mTotalReadBytes(0),
mTotalReads(0),
mPrevSampleTimeUs(0),
mNumLostFrames(0),
mGroup(NULL) {
LOGV("sampleRate: %d, channels: %d", sampleRate, channels);
@@ -110,10 +108,7 @@ status_t AudioSource::stop() {
mStarted = false;
if (mCollectStats) {
LOGI("%lld reads: %.2f bps in %lld us",
mTotalReads,
(mTotalReadBytes * 8000000.0) / mTotalReadTimeUs,
mTotalReadTimeUs);
LOGI("Total lost audio frames: %lld", mNumLostFrames);
}
return OK;
@@ -129,67 +124,113 @@ sp<MetaData> AudioSource::getFormat() {
return meta;
}
/*
* Returns -1 if frame skipping request is too long.
* Returns 0 if there is no need to skip frames.
* Returns 1 if we need to skip frames.
*/
static int skipFrame(int64_t timestampUs,
const MediaSource::ReadOptions *options) {
int64_t skipFrameUs;
if (!options || !options->getSkipFrame(&skipFrameUs)) {
return 0;
}
if (skipFrameUs <= timestampUs) {
return 0;
}
// Safe guard against the abuse of the kSkipFrame_Option.
if (skipFrameUs - timestampUs >= 1E6) {
LOGE("Frame skipping requested is way too long: %lld us",
skipFrameUs - timestampUs);
return -1;
}
LOGV("skipFrame: %lld us > timestamp: %lld us",
skipFrameUs, timestampUs);
return 1;
}
status_t AudioSource::read(
MediaBuffer **out, const ReadOptions *options) {
*out = NULL;
++mTotalReads;
MediaBuffer *buffer;
CHECK_EQ(mGroup->acquire_buffer(&buffer), OK);
int err = 0;
while (mStarted) {
uint32_t numFramesRecorded;
mRecord->getPosition(&numFramesRecorded);
int64_t latency = mRecord->latency() * 1000;
int64_t readTime = systemTime() / 1000;
if (numFramesRecorded == 0) {
if (numFramesRecorded == 0 && mPrevSampleTimeUs == 0) {
// Initial delay
if (mStartTimeUs > 0) {
mStartTimeUs = readTime - mStartTimeUs;
mStartTimeUs = systemTime() / 1000 - mStartTimeUs;
} else {
mStartTimeUs += latency;
// Assume latency is constant.
mStartTimeUs += mRecord->latency() * 1000;
}
}
ssize_t n = 0;
if (mCollectStats) {
n = mRecord->read(buffer->data(), buffer->size());
int64_t endTime = systemTime() / 1000;
mTotalReadTimeUs += (endTime - readTime);
if (n >= 0) {
mTotalReadBytes += n;
}
} else {
n = mRecord->read(buffer->data(), buffer->size());
}
if (n < 0) {
buffer->release();
buffer = NULL;
return (status_t)n;
mPrevSampleTimeUs = mStartTimeUs;
}
uint32_t sampleRate = mRecord->getSampleRate();
int64_t timestampUs = (1000000LL * numFramesRecorded) / sampleRate +
mStartTimeUs;
int64_t skipFrameUs;
if (!options || !options->getSkipFrame(&skipFrameUs)) {
skipFrameUs = timestampUs; // Don't skip frame
}
if (skipFrameUs > timestampUs) {
// Safe guard against the abuse of the kSkipFrame_Option.
if (skipFrameUs - timestampUs >= 1E6) {
LOGE("Frame skipping requested is way too long: %lld us",
skipFrameUs - timestampUs);
// Insert null frames when lost frames are detected.
int64_t timestampUs = mPrevSampleTimeUs;
uint32_t numLostBytes = mRecord->getInputFramesLost() << 1;
#if 0
// Simulate lost frames
numLostBytes = ((rand() * 1.0 / RAND_MAX)) * kMaxBufferSize;
numLostBytes &= 0xFFFFFFFE; // Alignment request
// Reduce the chance to lose
if (rand() * 1.0 / RAND_MAX >= 0.05) {
numLostBytes = 0;
}
#endif
if (numLostBytes > 0) {
// Not expect too many lost frames!
CHECK(numLostBytes <= kMaxBufferSize);
timestampUs += (1000000LL * numLostBytes >> 1) / sampleRate;
CHECK(timestampUs > mPrevSampleTimeUs);
if (mCollectStats) {
mNumLostFrames += (numLostBytes >> 1);
}
if ((err = skipFrame(timestampUs, options)) == -1) {
buffer->release();
return UNKNOWN_ERROR;
} else if (err != 0) {
continue;
}
LOGV("skipFrame: %lld us > timestamp: %lld us, samples %d",
skipFrameUs, timestampUs, numFramesRecorded);
memset(buffer->data(), 0, numLostBytes);
buffer->set_range(0, numLostBytes);
buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
mPrevSampleTimeUs = timestampUs;
*out = buffer;
return OK;
}
ssize_t n = mRecord->read(buffer->data(), buffer->size());
if (n < 0) {
buffer->release();
return (status_t)n;
}
int64_t recordDurationUs = (1000000LL * n >> 1) / sampleRate;
timestampUs += recordDurationUs;
if ((err = skipFrame(timestampUs, options)) == -1) {
buffer->release();
return UNKNOWN_ERROR;
} else if (err != 0) {
continue;
}
@@ -197,7 +238,13 @@ status_t AudioSource::read(
trackMaxAmplitude((int16_t *) buffer->data(), n >> 1);
}
buffer->meta_data()->setInt64(kKeyTime, timestampUs);
buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
CHECK(timestampUs > mPrevSampleTimeUs);
if (mNumLostFrames == 0) {
CHECK_EQ(mPrevSampleTimeUs,
mStartTimeUs + (1000000LL * numFramesRecorded) / sampleRate);
}
mPrevSampleTimeUs = timestampUs;
LOGV("initial delay: %lld, sample rate: %d, timestamp: %lld",
mStartTimeUs, sampleRate, timestampUs);