Merge "RTP: reduce the latency by overlapping AudioRecord and AudioTrack." into gingerbread (5b1f0c9f) · Commits · e / os / android_frameworks_base

voip/jni/rtp/AudioGroup.cpp

+53 −28

Original line number	Diff line number	Diff line
		@@ -548,10 +548,11 @@ bool AudioGroup::set(int sampleRate, int sampleCount)
		}
		LOGD("reported frame count: output %d, input %d", output, input);

		output = (output + sampleCount - 1) / sampleCount * sampleCount;
		input = (input + sampleCount - 1) / sampleCount * sampleCount;
		if (input < output * 2) {
		input = output * 2;
		if (output < sampleCount * 2) {
		output = sampleCount * 2;
		}
		if (input < sampleCount * 2) {
		input = sampleCount * 2;
		}
		LOGD("adjusted frame count: output %d, input %d", output, input);

		@@ -587,7 +588,7 @@ bool AudioGroup::set(int sampleRate, int sampleCount)
		// Give device socket a reasonable timeout and buffer size.
		timeval tv;
		tv.tv_sec = 0;
		tv.tv_usec = 1000 * sampleCount / sampleRate * 500;
		tv.tv_usec = 1000 * sampleCount / sampleRate * 1000;
		if (setsockopt(pair[0], SOL_SOCKET, SO_RCVTIMEO, &tv, sizeof(tv)) \|\|
		setsockopt(pair[0], SOL_SOCKET, SO_RCVBUF, &output, sizeof(output)) \|\|
		setsockopt(pair[1], SOL_SOCKET, SO_SNDBUF, &output, sizeof(output))) {
		@@ -764,38 +765,62 @@ bool AudioGroup::deviceLoop()
		if (recv(mDeviceSocket, output, sizeof(output), 0) <= 0) {
		memset(output, 0, sizeof(output));
		}
		if (mTrack.write(output, sizeof(output)) != (int)sizeof(output)) {

		int16_t input[mSampleCount];
		int toWrite = mSampleCount;
		int toRead = (mMode == MUTED) ? 0 : mSampleCount;
		int chances = 100;

		while (--chances > 0 && (toWrite > 0 \|\| toRead > 0)) {
		if (toWrite > 0) {
		AudioTrack::Buffer buffer;
		buffer.frameCount = toWrite;

		status_t status = mTrack.obtainBuffer(&buffer, 1);
		if (status == NO_ERROR) {
		memcpy(buffer.i8, &output[mSampleCount - toWrite], buffer.size);
		toWrite -= buffer.frameCount;
		mTrack.releaseBuffer(&buffer);
		} else if (status != TIMED_OUT && status != WOULD_BLOCK) {
		LOGE("cannot write to AudioTrack");
		return false;
		}
		}

		if (mMode != MUTED) {
		uint32_t frameCount = mRecord.frameCount();
		AudioRecord::Buffer input;
		input.frameCount = frameCount;
		if (toRead > 0) {
		AudioRecord::Buffer buffer;
		buffer.frameCount = mRecord.frameCount();

		if (mRecord.obtainBuffer(&input, -1) != NO_ERROR) {
		status_t status = mRecord.obtainBuffer(&buffer, 1);
		if (status == NO_ERROR) {
		int count = (buffer.frameCount < toRead) ?
		buffer.frameCount : toRead;
		memcpy(&input[mSampleCount - toRead], buffer.i8, count * 2);
		toRead -= count;
		if (buffer.frameCount < mRecord.frameCount()) {
		buffer.frameCount = count;
		}
		mRecord.releaseBuffer(&buffer);
		} else if (status != TIMED_OUT && status != WOULD_BLOCK) {
		LOGE("cannot read from AudioRecord");
		return false;
		}
		}
		}

		if (!chances) {
		LOGE("device loop timeout");
		return false;
		}

		if (input.frameCount < (uint32_t)mSampleCount) {
		input.frameCount = 0;
		} else {
		if (mMode != MUTED) {
		if (mMode == NORMAL) {
		send(mDeviceSocket, input.i8, sizeof(output), MSG_DONTWAIT);
		send(mDeviceSocket, input, sizeof(input), MSG_DONTWAIT);
		} else {
		// TODO: Echo canceller runs here.
		send(mDeviceSocket, input.i8, sizeof(output), MSG_DONTWAIT);
		send(mDeviceSocket, input, sizeof(input), MSG_DONTWAIT);
		}
		if (input.frameCount < frameCount) {
		input.frameCount = mSampleCount;
		}
		}

		mRecord.releaseBuffer(&input);
		}

		return true;
		}