Multi-client recording continued (4cc0a6a8) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioFlinger.cpp

+0 −2

Original line number	Diff line number	Diff line
		@@ -1847,8 +1847,6 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
		// pre processing modules
		RecordThread *thread = new RecordThread(this,
		input,
		reqSamplingRate,
		reqChannelMask,
		id,
		primaryOutputDevice_l(),
		*pDevices

services/audioflinger/Threads.cpp

+46 −39

Original line number	Diff line number	Diff line
		@@ -4451,8 +4451,6 @@ void AudioFlinger::DuplicatingThread::cacheParameters_l()

		AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
		AudioStreamIn *input,
		uint32_t sampleRate,
		audio_channel_mask_t channelMask,
		audio_io_handle_t id,
		audio_devices_t outDevice,
		audio_devices_t inDevice
		@@ -4463,10 +4461,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
		ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD),
		mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL),
		// mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l()
		mRsmpInRear(0),
		// FIXME these should be per-track, so this is only the initial track?
		mReqChannelCount(popcount(channelMask)),
		mReqSampleRate(sampleRate)
		mRsmpInRear(0)
		#ifdef TEE_SINK
		, mTeeSink(teeSink)
		#endif
		@@ -4598,13 +4593,6 @@ reacquire_wakelock:

		case TrackBase::STARTING_2:
		doBroadcast = true;
		if (mReqChannelCount != activeTrack->channelCount()) {
		ALOGW("wrong channel count");
		mActiveTracks.remove(activeTrack);
		mActiveTracksGen++;
		size--;
		continue;
		}
		mStandby = false;
		activeTrack->mState = TrackBase::ACTIVE;
		break;
		@@ -4715,6 +4703,10 @@ reacquire_wakelock:
		overrun = OVERRUN_TRUE;
		}

		if (framesOut == 0 \|\| framesIn == 0) {
		break;
		}

		if (activeTrack->mResampler == NULL) {
		// no resampling
		if (framesIn > framesOut) {
		@@ -4730,7 +4722,7 @@ reacquire_wakelock:
		part1 = framesIn;
		}
		int8_t src = (int8_t )mRsmpInBuffer + (front * mFrameSize);
		if (mChannelCount == mReqChannelCount) {
		if (mChannelCount == activeTrack->mChannelCount) {
		memcpy(dst, src, part1 * mFrameSize);
		} else if (mChannelCount == 1) {
		upmix_to_stereo_i16_from_mono_i16((int16_t )dst, (int16_t )src,
		@@ -4754,15 +4746,30 @@ reacquire_wakelock:
		// FIXME only re-calculate when it changes, and optimize for common ratios
		double inOverOut = (double) mSampleRate / activeTrack->mSampleRate;
		double outOverIn = (double) activeTrack->mSampleRate / mSampleRate;
		framesInNeeded = framesOut * inOverOut;
		framesInNeeded = ceil(framesOut * inOverOut) + 1;
		if (framesIn < framesInNeeded) {
		ALOGV("not enough to resample: have %u but need %u to produce %u",
		framesIn, framesInNeeded, framesOut);
		size_t newFramesOut = framesIn * outOverIn;
		size_t newFramesInNeeded = newFramesOut * inOverOut;
		LOG_ALWAYS_FATAL_IF(framesIn < newFramesInNeeded);
		ALOGV("not enough to resample: have %u but need %u to produce %u "
		"given in/out ratio of %.4g",
		framesIn, framesInNeeded, framesOut, inOverOut);
		size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0;
		size_t newFramesInNeeded = ceil(newFramesOut * inOverOut) + 1;
		ALOGV("now need %u frames to produce %u given out/in ratio of %.4g",
		newFramesInNeeded, newFramesOut, outOverIn);
		if (framesIn < newFramesInNeeded) {
		ALOGE("failure: have %u but need %u", framesIn, newFramesInNeeded);
		framesOut = 0;
		} else {
		ALOGV("success 2: have %u and need %u to produce %u "
		"given in/out ratio of %.4g",
		framesIn, newFramesInNeeded, newFramesOut, inOverOut);
		LOG_ALWAYS_FATAL_IF(newFramesOut > framesOut);
		framesOut = newFramesOut;
		}
		} else {
		ALOGI("success 1: have %u and need %u to produce %u "
		"given in/out ratio of %.4g",
		framesIn, framesInNeeded, framesOut, inOverOut);
		}

		// reallocate mRsmpOutBuffer as needed; we will grow but never shrink
		if (activeTrack->mRsmpOutFrameCount < framesOut) {
		@@ -4779,7 +4786,7 @@ reacquire_wakelock:
		/this/ /* AudioBufferProvider* */);
		// ditherAndClamp() works as long as all buffers returned by
		// activeTrack->getNextBuffer() are 32 bit aligned which should be always true.
		if (mReqChannelCount == 1) {
		if (activeTrack->mChannelCount == 1) {
		// temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t
		ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer,
		framesOut);
		@@ -4826,9 +4833,6 @@ reacquire_wakelock:
		}

		if (framesOut == 0) {
		if (overrun == OVERRUN_UNKNOWN) {
		overrun = OVERRUN_TRUE;
		}
		break;
		}
		}
		@@ -4849,7 +4853,6 @@ reacquire_wakelock:
		activeTrack->clearOverflow();
		break;
		case OVERRUN_UNKNOWN:
		LOG_FATAL("OVERRUN_UNKNOWN");
		break;
		}

		@@ -5025,7 +5028,7 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac
		} else {
		// do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs
		recordTrack->mFramesToDrop = -
		((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000);
		((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000);
		}
		}

		@@ -5042,6 +5045,9 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac
		return status;
		}

		// TODO consider other ways of handling this, such as changing the state to :STARTING and
		// adding the track to mActiveTracks after returning from AudioSystem::startInput(),
		// or using a separate command thread
		recordTrack->mState = TrackBase::STARTING_1;
		mActiveTracks.add(recordTrack);
		mActiveTracksGen++;
		@@ -5176,8 +5182,6 @@ void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& a

		if (mActiveTracks.size() > 0) {
		fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize);
		fdprintf(fd, " Out channel count: %u\n", mReqChannelCount);
		fdprintf(fd, " Out sample rate: %u\n", mReqSampleRate);
		} else {
		fdprintf(fd, " No active record clients\n");
		}
		@@ -5260,7 +5264,7 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
		}
		if (part1 == 0) {
		// Higher-level should keep mRsmpInBuffer full, and not call resampler if empty
		ALOGE("RecordThread::getNextBuffer() starved");
		LOG_FATAL("RecordThread::getNextBuffer() starved");
		buffer->raw = NULL;
		buffer->frameCount = 0;
		activeTrack->mRsmpInUnrel = 0;
		@@ -5299,11 +5303,14 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l()
		AudioParameter param = AudioParameter(keyValuePair);
		int value;
		audio_format_t reqFormat = mFormat;
		uint32_t reqSamplingRate = mReqSampleRate;
		audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount);
		uint32_t samplingRate = mSampleRate;
		audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount);

		// TODO Investigate when this code runs. Check with audio policy when a sample rate and
		// channel count change can be requested. Do we mandate the first client defines the
		// HAL sampling rate and channel count or do we allow changes on the fly?
		if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
		reqSamplingRate = value;
		samplingRate = value;
		reconfig = true;
		}
		if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
		@@ -5319,7 +5326,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l()
		if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) {
		status = BAD_VALUE;
		} else {
		reqChannelMask = mask;
		channelMask = mask;
		reconfig = true;
		}
		}
		@@ -5384,11 +5391,11 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l()
		reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
		reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
		(mInput->stream->common.get_sample_rate(&mInput->stream->common)
		<= (2 * reqSamplingRate)) &&
		<= (2 * samplingRate)) &&
		popcount(mInput->stream->common.get_channels(&mInput->stream->common))
		<= FCC_2 &&
		(reqChannelMask == AUDIO_CHANNEL_IN_MONO \|\|
		reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) {
		(channelMask == AUDIO_CHANNEL_IN_MONO \|\|
		channelMask == AUDIO_CHANNEL_IN_STEREO)) {
		status = NO_ERROR;
		}
		if (status == NO_ERROR) {
		@@ -5468,8 +5475,8 @@ void AudioFlinger::RecordThread::readInputParameters_l()
		// Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
		mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount];

		// mReqSampleRate and mReqChannelCount are constant due to AudioRecord API constraints.
		// But if mSampleRate or mChannelCount changes, how will that affect active tracks?
		// AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.
		// But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?
		}

		uint32_t AudioFlinger::RecordThread::getInputFramesLost()

services/audioflinger/Threads.h

+0 −6

Original line number	Diff line number	Diff line
		@@ -861,8 +861,6 @@ public:

		RecordThread(const sp<AudioFlinger>& audioFlinger,
		AudioStreamIn *input,
		uint32_t sampleRate,
		audio_channel_mask_t channelMask,
		audio_io_handle_t id,
		audio_devices_t outDevice,
		audio_devices_t inDevice
		@@ -959,10 +957,6 @@ private:
		// rolling index that is never cleared
		int32_t mRsmpInRear; // last filled frame + 1

		// client's requested configuration, which may differ from the HAL configuration
		const uint32_t mReqChannelCount;
		const uint32_t mReqSampleRate;

		// For dumpsys
		const sp<NBAIO_Sink> mTeeSink;
		};