Merge "Move validation of frameCount from set to openRecord_l" (59121171) · Commits · e / os / android_frameworks_av

media/libmedia/AudioRecord.cpp

+24 −18

Original line number	Diff line number	Diff line
		@@ -203,23 +203,6 @@ status_t AudioRecord::set(
		mFrameSize = sizeof(uint8_t);
		}

		// validate framecount
		size_t minFrameCount;
		status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
		sampleRate, format, channelMask);
		if (status != NO_ERROR) {
		ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; status %d",
		sampleRate, format, channelMask, status);
		return status;
		}
		ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);

		if (frameCount == 0) {
		frameCount = minFrameCount;
		} else if (frameCount < minFrameCount) {
		ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
		return BAD_VALUE;
		}
		// mFrameCount is initialized in openRecord_l
		mReqFrameCount = frameCount;

		@@ -242,7 +225,7 @@ status_t AudioRecord::set(
		}

		// create the IAudioRecord
		status = openRecord_l(0 /epoch/);
		status_t status = openRecord_l(0 /epoch/);

		if (status != NO_ERROR) {
		if (mAudioRecordThread != 0) {
		@@ -464,6 +447,29 @@ status_t AudioRecord::openRecord_l(size_t epoch)
		size_t frameCount = mReqFrameCount;

		if (!(mFlags & AUDIO_INPUT_FLAG_FAST)) {
		// validate framecount
		// If fast track was not requested, this preserves
		// the old behavior of validating on client side.
		// FIXME Eventually the validation should be done on server side
		// regardless of whether it's a fast or normal track. It's debatable
		// whether to account for the input latency to provision buffers appropriately.
		size_t minFrameCount;
		status = AudioRecord::getMinFrameCount(&minFrameCount,
		mSampleRate, mFormat, mChannelMask);
		if (status != NO_ERROR) {
		ALOGE("getMinFrameCount() failed for sampleRate %u, format %#x, channelMask %#x; "
		"status %d",
		mSampleRate, mFormat, mChannelMask, status);
		return status;
		}

		if (frameCount == 0) {
		frameCount = minFrameCount;
		} else if (frameCount < minFrameCount) {
		ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
		return BAD_VALUE;
		}

		// Make sure that application is notified with sufficient margin before overrun
		if (mNotificationFramesAct == 0 \|\| mNotificationFramesAct > frameCount/2) {
		mNotificationFramesAct = frameCount/2;

services/audioflinger/Threads.cpp

+1 −0

Original line number	Diff line number	Diff line
		@@ -5132,6 +5132,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe
		// to be at least 2 x the record thread frame count and cover audio hardware latency.
		// This is probably too conservative, but legacy application code may depend on it.
		// If you change this calculation, also review the start threshold which is related.
		// FIXME It's not clear how input latency actually matters. Perhaps this should be 0.
		uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream);
		size_t mNormalFrameCount = 2048; // FIXME
		uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);