Merge "libaudiopreprocessing: Cleanup PreProcessing.cpp" am: 85f5f27626 (7370827c) · Commits · e / os / android_frameworks_av

media/libeffects/preprocessing/PreProcessing.cpp

+28 −128

Original line number	Diff line number	Diff line
		@@ -105,9 +105,8 @@ struct preproc_session_s {
		webrtc::AudioProcessing* apm; // handle on webRTC audio processing module (APM)
		// Audio Processing module builder
		webrtc::AudioProcessingBuilder ap_builder;
		size_t apmFrameCount; // buffer size for webRTC process (10 ms)
		uint32_t apmSamplingRate; // webRTC APM sampling rate (8/16 or 32 kHz)
		size_t frameCount; // buffer size before input resampler ( <=> apmFrameCount)
		// frameCount represents the size of the buffers used for processing, and must represent 10ms.
		size_t frameCount;
		uint32_t samplingRate; // sampling rate at effect process interface
		uint32_t inChannelCount; // input channel count
		uint32_t outChannelCount; // output channel count
		@@ -119,21 +118,12 @@ struct preproc_session_s {
		webrtc::AudioProcessing::Config config;
		webrtc::StreamConfig inputConfig; // input stream configuration
		webrtc::StreamConfig outputConfig; // output stream configuration
		int16_t* inBuf; // input buffer used when resampling
		size_t inBufSize; // input buffer size in frames
		size_t framesIn; // number of frames in input buffer
		int16_t* outBuf; // output buffer used when resampling
		size_t outBufSize; // output buffer size in frames
		size_t framesOut; // number of frames in output buffer
		uint32_t revChannelCount; // number of channels on reverse stream
		uint32_t revEnabledMsk; // bit field containing IDs of enabled pre processors
		// with reverse channel
		uint32_t revProcessedMsk; // bit field containing IDs of pre processors with reverse
		// channel already processed in current round
		webrtc::StreamConfig revConfig; // reverse stream configuration.
		int16_t* revBuf; // reverse channel input buffer
		size_t revBufSize; // reverse channel input buffer size
		size_t framesRev; // number of frames in reverse channel input buffer
		};

		#ifdef DUAL_MIC_TEST
		@@ -862,9 +852,7 @@ extern "C" int Session_CreateEffect(preproc_session_t* session, int32_t procId,
		ALOGW("Session_CreateEffect could not get apm engine");
		goto error;
		}
		session->apmSamplingRate = kPreprocDefaultSr;
		session->apmFrameCount = (kPreprocDefaultSr) / 100;
		session->frameCount = session->apmFrameCount;
		session->frameCount = kPreprocDefaultSr / 100;
		session->samplingRate = kPreprocDefaultSr;
		session->inChannelCount = kPreProcDefaultCnl;
		session->outChannelCount = kPreProcDefaultCnl;
		@@ -879,12 +867,6 @@ extern "C" int Session_CreateEffect(preproc_session_t* session, int32_t procId,
		session->processedMsk = 0;
		session->revEnabledMsk = 0;
		session->revProcessedMsk = 0;
		session->inBuf = NULL;
		session->inBufSize = 0;
		session->outBuf = NULL;
		session->outBufSize = 0;
		session->revBuf = NULL;
		session->revBufSize = 0;
		}
		status = Effect_Create(&session->effects[procId], session, interface);
		if (status < 0) {
		@@ -908,13 +890,6 @@ int Session_ReleaseEffect(preproc_session_t* session, preproc_effect_t* fx) {
		if (session->createdMsk == 0) {
		delete session->apm;
		session->apm = NULL;
		delete session->inBuf;
		session->inBuf = NULL;
		free(session->outBuf);
		session->outBuf = NULL;
		delete session->revBuf;
		session->revBuf = NULL;

		session->id = 0;
		}

		@@ -934,24 +909,8 @@ int Session_SetConfig(preproc_session_t* session, effect_config_t* config) {
		ALOGV("Session_SetConfig sr %d cnl %08x", config->inputCfg.samplingRate,
		config->inputCfg.channels);

		// AEC implementation is limited to 16kHz
		if (config->inputCfg.samplingRate >= 32000 && !(session->createdMsk & (1 << PREPROC_AEC))) {
		session->apmSamplingRate = 32000;
		} else if (config->inputCfg.samplingRate >= 16000) {
		session->apmSamplingRate = 16000;
		} else if (config->inputCfg.samplingRate >= 8000) {
		session->apmSamplingRate = 8000;
		}


		session->samplingRate = config->inputCfg.samplingRate;
		session->apmFrameCount = session->apmSamplingRate / 100;
		if (session->samplingRate == session->apmSamplingRate) {
		session->frameCount = session->apmFrameCount;
		} else {
		session->frameCount =
		(session->apmFrameCount * session->samplingRate) / session->apmSamplingRate;
		}
		session->frameCount = session->samplingRate / 100;
		session->inChannelCount = inCnl;
		session->outChannelCount = outCnl;
		session->inputConfig.set_sample_rate_hz(session->samplingRate);
		@@ -963,13 +922,6 @@ int Session_SetConfig(preproc_session_t* session, effect_config_t* config) {
		session->revConfig.set_sample_rate_hz(session->samplingRate);
		session->revConfig.set_num_channels(inCnl);

		// force process buffer reallocation
		session->inBufSize = 0;
		session->outBufSize = 0;
		session->framesIn = 0;
		session->framesOut = 0;


		session->state = PREPROC_SESSION_STATE_CONFIG;
		return 0;
		}
		@@ -1004,9 +956,6 @@ int Session_SetReverseConfig(preproc_session_t* session, effect_config_t* config
		}
		uint32_t inCnl = audio_channel_count_from_out_mask(config->inputCfg.channels);
		session->revChannelCount = inCnl;
		// force process buffer reallocation
		session->revBufSize = 0;
		session->framesRev = 0;

		return 0;
		}
		@@ -1023,12 +972,8 @@ void Session_GetReverseConfig(preproc_session_t* session, effect_config_t* confi

		void Session_SetProcEnabled(preproc_session_t* session, uint32_t procId, bool enabled) {
		if (enabled) {
		if (session->enabledMsk == 0) {
		session->framesIn = 0;
		}
		session->enabledMsk \|= (1 << procId);
		if (HasReverseStream(procId)) {
		session->framesRev = 0;
		session->revEnabledMsk \|= (1 << procId);
		}
		} else {
		@@ -1117,43 +1062,24 @@ int PreProcessingFx_Process(effect_handle_t self, audio_buffer_t* inBuffer,
		return -EINVAL;
		}

		if (inBuffer->frameCount != outBuffer->frameCount) {
		ALOGW("inBuffer->frameCount %zu is not equal to outBuffer->frameCount %zu",
		inBuffer->frameCount, outBuffer->frameCount);
		return -EINVAL;
		}

		if (inBuffer->frameCount != session->frameCount) {
		ALOGW("inBuffer->frameCount %zu != %zu representing 10ms at sampling rate %d",
		inBuffer->frameCount, session->frameCount, session->samplingRate);
		return -EINVAL;
		}

		session->processedMsk \|= (1 << effect->procId);

		// ALOGV("PreProcessingFx_Process In %d frames enabledMsk %08x processedMsk %08x",
		// inBuffer->frameCount, session->enabledMsk, session->processedMsk);

		if ((session->processedMsk & session->enabledMsk) == session->enabledMsk) {
		effect->session->processedMsk = 0;
		size_t framesRq = outBuffer->frameCount;
		size_t framesWr = 0;
		if (session->framesOut) {
		size_t fr = session->framesOut;
		if (outBuffer->frameCount < fr) {
		fr = outBuffer->frameCount;
		}
		memcpy(outBuffer->s16, session->outBuf,
		fr * session->outChannelCount * sizeof(int16_t));
		memmove(session->outBuf, session->outBuf + fr * session->outChannelCount,
		(session->framesOut - fr) * session->outChannelCount * sizeof(int16_t));
		session->framesOut -= fr;
		framesWr += fr;
		}
		outBuffer->frameCount = framesWr;
		if (framesWr == framesRq) {
		inBuffer->frameCount = 0;
		return 0;
		}

		size_t fr = session->frameCount - session->framesIn;
		if (inBuffer->frameCount < fr) {
		fr = inBuffer->frameCount;
		}
		session->framesIn += fr;
		inBuffer->frameCount = fr;
		if (session->framesIn < session->frameCount) {
		return 0;
		}
		session->framesIn = 0;
		if (int status = effect->session->apm->ProcessStream(
		(const int16_t* const)inBuffer->s16,
		(const webrtc::StreamConfig)effect->session->inputConfig,
		@@ -1163,34 +1089,6 @@ int PreProcessingFx_Process(effect_handle_t self, audio_buffer_t* inBuffer,
		ALOGE("Process Stream failed with error %d\n", status);
		return status;
		}
		outBuffer->frameCount = inBuffer->frameCount;

		if (session->outBufSize < session->framesOut + session->frameCount) {
		int16_t* buf;
		session->outBufSize = session->framesOut + session->frameCount;
		buf = (int16_t*)realloc(
		session->outBuf,
		session->outBufSize * session->outChannelCount * sizeof(int16_t));
		if (buf == NULL) {
		session->framesOut = 0;
		free(session->outBuf);
		session->outBuf = NULL;
		return -ENOMEM;
		}
		session->outBuf = buf;
		}

		fr = session->framesOut;
		if (framesRq - framesWr < fr) {
		fr = framesRq - framesWr;
		}
		memcpy(outBuffer->s16 + framesWr * session->outChannelCount, session->outBuf,
		fr * session->outChannelCount * sizeof(int16_t));
		memmove(session->outBuf, session->outBuf + fr * session->outChannelCount,
		(session->framesOut - fr) * session->outChannelCount * sizeof(int16_t));
		session->framesOut -= fr;
		outBuffer->frameCount += fr;

		return 0;
		} else {
		return -ENODATA;
		@@ -1565,6 +1463,18 @@ int PreProcessingFx_ProcessReverse(effect_handle_t self, audio_buffer_t* inBuffe
		return -EINVAL;
		}

		if (inBuffer->frameCount != outBuffer->frameCount) {
		ALOGW("inBuffer->frameCount %zu is not equal to outBuffer->frameCount %zu",
		inBuffer->frameCount, outBuffer->frameCount);
		return -EINVAL;
		}

		if (inBuffer->frameCount != session->frameCount) {
		ALOGW("inBuffer->frameCount %zu != %zu representing 10ms at sampling rate %d",
		inBuffer->frameCount, session->frameCount, session->samplingRate);
		return -EINVAL;
		}

		session->revProcessedMsk \|= (1 << effect->procId);

		// ALOGV("PreProcessingFx_ProcessReverse In %d frames revEnabledMsk %08x revProcessedMsk
		@@ -1573,16 +1483,6 @@ int PreProcessingFx_ProcessReverse(effect_handle_t self, audio_buffer_t* inBuffe

		if ((session->revProcessedMsk & session->revEnabledMsk) == session->revEnabledMsk) {
		effect->session->revProcessedMsk = 0;
		size_t fr = session->frameCount - session->framesRev;
		if (inBuffer->frameCount < fr) {
		fr = inBuffer->frameCount;
		}
		session->framesRev += fr;
		inBuffer->frameCount = fr;
		if (session->framesRev < session->frameCount) {
		return 0;
		}
		session->framesRev = 0;
		if (int status = effect->session->apm->ProcessReverseStream(
		(const int16_t* const)inBuffer->s16,
		(const webrtc::StreamConfig)effect->session->revConfig,

media/libeffects/preprocessing/README.md

0 → 100644

+7 −0

Original line number	Diff line number	Diff line
		# Preprocessing effects

		## Limitations
		- Preprocessing effects currently work on 10ms worth of data and do not support
		arbitrary frame counts. This limiation comes from the underlying effects in
		webrtc modules
		- There is currently no api to communicate this requirement

media/libeffects/preprocessing/tests/PreProcessingTest.cpp

+4 −3

Original line number	Diff line number	Diff line
		@@ -451,8 +451,8 @@ int main(int argc, const char* argv[]) {
		}
		audio_buffer_t inputBuffer, outputBuffer;
		audio_buffer_t farInBuffer{};
		inputBuffer.frameCount = samplesRead;
		outputBuffer.frameCount = samplesRead;
		inputBuffer.frameCount = frameLength;
		outputBuffer.frameCount = frameLength;
		inputBuffer.s16 = in.data();
		outputBuffer.s16 = out.data();

		@@ -472,7 +472,7 @@ int main(int argc, const char* argv[]) {
		}
		}

		farInBuffer.frameCount = samplesRead;
		farInBuffer.frameCount = frameLength;
		farInBuffer.s16 = farIn.data();
		}

		@@ -519,6 +519,7 @@ int main(int argc, const char* argv[]) {
		}
		frameCounter += frameLength;
		}
		printf("frameCounter: [%d]\n", frameCounter);
		// Release all the effect handles created
		for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
		if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(effectHandle[i]);