Merge "Store server latency, sample rate, framecount information in AudioTrack" into mnc-dev (0eafa9d8) · Commits · e / os / android_frameworks_av

include/media/AudioTrack.h

+7 −0

Original line number	Diff line number	Diff line
		@@ -801,6 +801,13 @@ protected:
		size_t mReqFrameCount; // frame count to request the first or next time
		// a new IAudioTrack is needed, non-decreasing

		// The following AudioFlinger server-side values are cached in createAudioTrack_l().
		// These values can be used for informational purposes until the track is invalidated,
		// whereupon restoreTrack_l() calls createTrack_l() to update the values.
		uint32_t mAfLatency; // AudioFlinger latency in ms
		size_t mAfFrameCount; // AudioFlinger frame count
		uint32_t mAfSampleRate; // AudioFlinger sample rate

		// constant after constructor or set()
		audio_format_t mFormat; // as requested by client, not forced to 16-bit
		audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies

media/libmedia/AudioTrack.cpp

+12 −38

Original line number	Diff line number	Diff line
		@@ -1111,31 +1111,27 @@ status_t AudioTrack::createTrack_l()
		// we must release it ourselves if anything goes wrong.

		// Not all of these values are needed under all conditions, but it is easier to get them all

		uint32_t afLatency;
		status = AudioSystem::getLatency(output, &afLatency);
		status = AudioSystem::getLatency(output, &mAfLatency);
		if (status != NO_ERROR) {
		ALOGE("getLatency(%d) failed status %d", output, status);
		goto release;
		}
		ALOGV("createTrack_l() output %d afLatency %u", output, afLatency);
		ALOGV("createTrack_l() output %d afLatency %u", output, mAfLatency);

		size_t afFrameCount;
		status = AudioSystem::getFrameCount(output, &afFrameCount);
		status = AudioSystem::getFrameCount(output, &mAfFrameCount);
		if (status != NO_ERROR) {
		ALOGE("getFrameCount(output=%d) status %d", output, status);
		goto release;
		}

		uint32_t afSampleRate;
		status = AudioSystem::getSamplingRate(output, &afSampleRate);
		status = AudioSystem::getSamplingRate(output, &mAfSampleRate);
		if (status != NO_ERROR) {
		ALOGE("getSamplingRate(output=%d) status %d", output, status);
		goto release;
		}
		if (mSampleRate == 0) {
		mSampleRate = afSampleRate;
		mOriginalSampleRate = afSampleRate;
		mSampleRate = mAfSampleRate;
		mOriginalSampleRate = mAfSampleRate;
		}
		// Client decides whether the track is TIMED (see below), but can only express a preference
		// for FAST. Server will perform additional tests.
		@@ -1148,9 +1144,9 @@ status_t AudioTrack::createTrack_l()
		// use case 3: obtain/release mode
		(mTransfer == TRANSFER_OBTAIN)) &&
		// matching sample rate
		(mSampleRate == afSampleRate))) {
		(mSampleRate == mAfSampleRate))) {
		ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d, track %u Hz, output %u Hz",
		mTransfer, mSampleRate, afSampleRate);
		mTransfer, mSampleRate, mAfSampleRate);
		// once denied, do not request again if IAudioTrack is re-created
		mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
		}
		@@ -1171,7 +1167,7 @@ status_t AudioTrack::createTrack_l()
		// Same comment as below about ignoring frameCount parameter for set()
		frameCount = mSharedBuffer->size();
		} else if (frameCount == 0) {
		frameCount = afFrameCount;
		frameCount = mAfFrameCount;
		}
		if (mNotificationFramesAct != frameCount) {
		mNotificationFramesAct = frameCount;
		@@ -1207,7 +1203,7 @@ status_t AudioTrack::createTrack_l()
		if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
		// for normal tracks precompute the frame count based on speed.
		const size_t minFrameCount = calculateMinFrameCount(
		afLatency, afFrameCount, afSampleRate, mSampleRate,
		mAfLatency, mAfFrameCount, mAfSampleRate, mSampleRate,
		mPlaybackRate.mSpeed);
		if (frameCount < minFrameCount) {
		frameCount = minFrameCount;
		@@ -1357,7 +1353,7 @@ status_t AudioTrack::createTrack_l()
		mAudioTrack->attachAuxEffect(mAuxEffectId);
		// FIXME doesn't take into account speed or future sample rate changes (until restoreTrack)
		// FIXME don't believe this lie
		mLatency = afLatency + (1000*frameCount) / mSampleRate;
		mLatency = mAfLatency + (1000*frameCount) / mSampleRate;

		mFrameCount = frameCount;
		// If IAudioTrack is re-created, don't let the requested frameCount
		@@ -2089,30 +2085,8 @@ bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) co
		if (mStaticProxy != 0) {
		return true; // static tracks do not have issues with buffer sizing.
		}
		status_t status;
		uint32_t afLatency;
		status = AudioSystem::getLatency(mOutput, &afLatency);
		if (status != NO_ERROR) {
		ALOGE("getLatency(%d) failed status %d", mOutput, status);
		return false;
		}

		size_t afFrameCount;
		status = AudioSystem::getFrameCount(mOutput, &afFrameCount);
		if (status != NO_ERROR) {
		ALOGE("getFrameCount(output=%d) status %d", mOutput, status);
		return false;
		}

		uint32_t afSampleRate;
		status = AudioSystem::getSamplingRate(mOutput, &afSampleRate);
		if (status != NO_ERROR) {
		ALOGE("getSamplingRate(output=%d) status %d", mOutput, status);
		return false;
		}

		const size_t minFrameCount =
		calculateMinFrameCount(afLatency, afFrameCount, afSampleRate, sampleRate, speed);
		calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed);
		ALOGV("isSampleRateSpeedAllowed_l mFrameCount %zu minFrameCount %zu",
		mFrameCount, minFrameCount);
		return mFrameCount >= minFrameCount;