Add lost frame handling in AudioSource
- Also collect stats on lost audio frames instead of time spent on reading
Change-Id: I6380b143e4fbdcd894491aaae523331e90d0f04f
diff --git a/include/media/stagefright/AudioSource.h b/include/media/stagefright/AudioSource.h
index 628200d..f1d45d2 100644
--- a/include/media/stagefright/AudioSource.h
+++ b/include/media/stagefright/AudioSource.h
@@ -57,11 +57,10 @@
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;
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index 50c0edc..99978e8 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -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 @@
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 @@
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 @@
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 @@
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);