Merge "Limit the amount of audio record data in each buffer" (f65934a7) · Commits · e / os / android_frameworks_base

include/media/stagefright/AudioSource.h

+1 −0

Original line number	Diff line number	Diff line
		@@ -95,6 +95,7 @@ private:
		int32_t startFrame, int32_t rampDurationFrames,
		uint8_t *data, size_t bytes);

		void queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs);
		void releaseQueuedFrames_l();
		void waitOutstandingEncodingFrames_l();
		status_t reset();

media/libstagefright/AudioSource.cpp

+38 −19

Original line number	Diff line number	Diff line
		@@ -282,8 +282,6 @@ status_t AudioSource::dataCallbackTimestamp(
		mPrevSampleTimeUs = mStartTimeUs;
		}

		int64_t timestampUs = mPrevSampleTimeUs;

		size_t numLostBytes = 0;
		if (mNumFramesReceived > 0) { // Ignore earlier frame lost
		// getInputFramesLost() returns the number of lost frames.
		@@ -293,37 +291,58 @@ status_t AudioSource::dataCallbackTimestamp(

		CHECK_EQ(numLostBytes & 1, 0u);
		CHECK_EQ(audioBuffer.size & 1, 0u);
		size_t bufferSize = numLostBytes + audioBuffer.size;
		MediaBuffer *buffer = new MediaBuffer(bufferSize);
		if (numLostBytes > 0) {
		memset(buffer->data(), 0, numLostBytes);
		memcpy((uint8_t *) buffer->data() + numLostBytes,
		audioBuffer.i16, audioBuffer.size);
		// Loss of audio frames should happen rarely; thus the LOGW should
		// not cause a logging spam
		ALOGW("Lost audio record data: %d bytes", numLostBytes);
		}

		while (numLostBytes > 0) {
		size_t bufferSize = numLostBytes;
		if (numLostBytes > kMaxBufferSize) {
		numLostBytes -= kMaxBufferSize;
		bufferSize = kMaxBufferSize;
		} else {
		numLostBytes = 0;
		}
		MediaBuffer *lostAudioBuffer = new MediaBuffer(bufferSize);
		memset(lostAudioBuffer->data(), 0, bufferSize);
		lostAudioBuffer->set_range(0, bufferSize);
		queueInputBuffer_l(lostAudioBuffer, timeUs);
		}

		if (audioBuffer.size == 0) {
		ALOGW("Nothing is available from AudioRecord callback buffer");
		buffer->release();
		return OK;
		}

		const size_t bufferSize = audioBuffer.size;
		MediaBuffer *buffer = new MediaBuffer(bufferSize);
		memcpy((uint8_t *) buffer->data(),
		audioBuffer.i16, audioBuffer.size);
		buffer->set_range(0, bufferSize);
		queueInputBuffer_l(buffer, timeUs);
		return OK;
		}

		buffer->set_range(0, bufferSize);
		timestampUs += ((1000000LL * (bufferSize >> 1)) +
		void AudioSource::queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs) {
		const size_t bufferSize = buffer->range_length();
		const size_t frameSize = mRecord->frameSize();
		const int64_t timestampUs =
		mPrevSampleTimeUs +
		((1000000LL * (bufferSize / frameSize)) +
		(mSampleRate >> 1)) / mSampleRate;

		if (mNumFramesReceived == 0) {
		buffer->meta_data()->setInt64(kKeyAnchorTime, mStartTimeUs);
		}

		buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
		buffer->meta_data()->setInt64(kKeyDriftTime, timeUs - mInitialReadTimeUs);
		mPrevSampleTimeUs = timestampUs;
		mNumFramesReceived += buffer->range_length() / sizeof(int16_t);
		mNumFramesReceived += bufferSize / frameSize;
		mBuffersReceived.push_back(buffer);
		mFrameAvailableCondition.signal();

		return OK;
		}

		void AudioSource::trackMaxAmplitude(int16_t *data, int nSamples) {