Merge "Add playback rate to AudioMixer" (e62e1637) · Commits · e / os / android_frameworks_av

include/media/AudioResamplerPublic.h

+8 −0

Original line number	Diff line number	Diff line
		@@ -34,6 +34,14 @@
		// an int32_t of the phase increments, making the resulting sample rate inexact.
		#define AUDIO_RESAMPLER_UP_RATIO_MAX 65536

		#define AUDIO_TIMESTRETCH_SPEED_MIN 0.5f
		#define AUDIO_TIMESTRETCH_SPEED_MAX 2.0f
		#define AUDIO_TIMESTRETCH_SPEED_NORMAL 1.0f

		#define AUDIO_TIMESTRETCH_PITCH_MIN 0.5f
		#define AUDIO_TIMESTRETCH_PITCH_MAX 2.0f
		#define AUDIO_TIMESTRETCH_PITCH_NORMAL 1.0f

		// Returns the source frames needed to resample to destination frames. This is not a precise
		// value and depends on the resampler (and possibly how it handles rounding internally).
		// Nevertheless, this should be an upper bound on the requirements of the resampler.

services/audioflinger/AudioMixer.cpp

+56 −1

Original line number	Diff line number	Diff line
		@@ -123,6 +123,7 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTr
		t->resampler = NULL;
		t->downmixerBufferProvider = NULL;
		t->mReformatBufferProvider = NULL;
		t->mTimestretchBufferProvider = NULL;
		t++;
		}

		@@ -135,6 +136,7 @@ AudioMixer::~AudioMixer()
		delete t->resampler;
		delete t->downmixerBufferProvider;
		delete t->mReformatBufferProvider;
		delete t->mTimestretchBufferProvider;
		t++;
		}
		delete [] mState.outputTemp;
		@@ -213,6 +215,7 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask,
		t->mReformatBufferProvider = NULL;
		t->downmixerBufferProvider = NULL;
		t->mPostDownmixReformatBufferProvider = NULL;
		t->mTimestretchBufferProvider = NULL;
		t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
		t->mFormat = format;
		t->mMixerInFormat = selectMixerInFormat(format);
		@@ -220,6 +223,8 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask,
		t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
		AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
		t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
		t->mSpeed = AUDIO_TIMESTRETCH_SPEED_NORMAL;
		t->mPitch = AUDIO_TIMESTRETCH_PITCH_NORMAL;
		// Check the downmixing (or upmixing) requirements.
		status_t status = t->prepareForDownmix();
		if (status != OK) {
		@@ -412,6 +417,10 @@ void AudioMixer::track_t::reconfigureBufferProviders()
		mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider);
		bufferProvider = mPostDownmixReformatBufferProvider;
		}
		if (mTimestretchBufferProvider) {
		mTimestretchBufferProvider->setBufferProvider(bufferProvider);
		bufferProvider = mTimestretchBufferProvider;
		}
		}

		void AudioMixer::deleteTrackName(int name)
		@@ -432,7 +441,9 @@ void AudioMixer::deleteTrackName(int name)
		mState.tracks[name].unprepareForDownmix();
		// delete the reformatter
		mState.tracks[name].unprepareForReformat();

		// delete the timestretch provider
		delete track.mTimestretchBufferProvider;
		track.mTimestretchBufferProvider = NULL;
		mTrackNames &= ~(1<<name);
		}

		@@ -654,6 +665,26 @@ void AudioMixer::setParameter(int name, int target, int param, void *value)
		}
		}
		break;
		case TIMESTRETCH:
		switch (param) {
		case PLAYBACK_RATE: {
		const float speed = reinterpret_cast<float*>(value)[0];
		const float pitch = reinterpret_cast<float*>(value)[1];
		ALOG_ASSERT(AUDIO_TIMESTRETCH_SPEED_MIN <= speed
		&& speed <= AUDIO_TIMESTRETCH_SPEED_MAX,
		"bad speed %f", speed);
		ALOG_ASSERT(AUDIO_TIMESTRETCH_PITCH_MIN <= pitch
		&& pitch <= AUDIO_TIMESTRETCH_PITCH_MAX,
		"bad pitch %f", pitch);
		if (track.setPlaybackRate(speed, pitch)) {
		ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, %f %f", speed, pitch);
		// invalidateState(1 << name);
		}
		} break;
		default:
		LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
		}
		break;

		default:
		LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
		@@ -699,6 +730,28 @@ bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSam
		return false;
		}

		bool AudioMixer::track_t::setPlaybackRate(float speed, float pitch)
		{
		if (speed == mSpeed && pitch == mPitch) {
		return false;
		}
		mSpeed = speed;
		mPitch = pitch;
		if (mTimestretchBufferProvider == NULL) {
		// TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
		// but if none exists, it is the channel count (1 for mono).
		const int timestretchChannelCount = downmixerBufferProvider != NULL
		? mMixerChannelCount : channelCount;
		mTimestretchBufferProvider = new TimestretchBufferProvider(timestretchChannelCount,
		mMixerInFormat, sampleRate, speed, pitch);
		reconfigureBufferProviders();
		} else {
		reinterpret_cast<TimestretchBufferProvider*>(mTimestretchBufferProvider)
		->setPlaybackRate(speed, pitch);
		}
		return true;
		}

		/* Checks to see if the volume ramp has completed and clears the increment
		* variables appropriately.
		*
		@@ -777,6 +830,8 @@ void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider
		mState.tracks[name].downmixerBufferProvider->reset();
		} else if (mState.tracks[name].mPostDownmixReformatBufferProvider != NULL) {
		mState.tracks[name].mPostDownmixReformatBufferProvider->reset();
		} else if (mState.tracks[name].mTimestretchBufferProvider != NULL) {
		mState.tracks[name].mTimestretchBufferProvider->reset();
		}

		mState.tracks[name].mInputBufferProvider = bufferProvider;

services/audioflinger/AudioMixer.h

+13 −0

Original line number	Diff line number	Diff line
		@@ -73,6 +73,7 @@ public:
		RESAMPLE = 0x3001,
		RAMP_VOLUME = 0x3002, // ramp to new volume
		VOLUME = 0x3003, // don't ramp
		TIMESTRETCH = 0x3004,

		// set Parameter names
		// for target TRACK
		@@ -100,6 +101,9 @@ public:
		VOLUME0 = 0x4200,
		VOLUME1 = 0x4201,
		AUXLEVEL = 0x4210,
		// for target TIMESTRETCH
		PLAYBACK_RATE = 0x4300, // Configure timestretch on this track name;
		// parameter 'value' is a pointer to the new playback rate.
		};


		@@ -213,6 +217,9 @@ private:
		// 16-byte boundary

		/* Buffer providers are constructed to translate the track input data as needed.
		*
		* TODO: perhaps make a single PlaybackConverterProvider class to move
		* all pre-mixer track buffer conversions outside the AudioMixer class.
		*
		* 1) mInputBufferProvider: The AudioTrack buffer provider.
		* 2) mReformatBufferProvider: If not NULL, performs the audio reformat to
		@@ -223,11 +230,13 @@ private:
		* the number of channels required by the mixer sink.
		* 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
		* the downmixer requirements to the mixer engine input requirements.
		* 5) mTimestretchBufferProvider: Adds timestretching for playback rate
		*/
		AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider.
		PassthruBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting.
		PassthruBufferProvider* downmixerBufferProvider; // wrapper for channel conversion.
		PassthruBufferProvider* mPostDownmixReformatBufferProvider;
		PassthruBufferProvider* mTimestretchBufferProvider;

		int32_t sessionId;

		@@ -250,6 +259,9 @@ private:
		audio_channel_mask_t mMixerChannelMask;
		uint32_t mMixerChannelCount;

		float mSpeed;
		float mPitch;

		bool needsRamp() { return (volumeInc[0] \| volumeInc[1] \| auxInc) != 0; }
		bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
		bool doesResample() const { return resampler != NULL; }
		@@ -262,6 +274,7 @@ private:
		void unprepareForDownmix();
		status_t prepareForReformat();
		void unprepareForReformat();
		bool setPlaybackRate(float speed, float pitch);
		void reconfigureBufferProviders();
		};

services/audioflinger/BufferProviders.cpp

+162 −0

Original line number	Diff line number	Diff line
		@@ -20,7 +20,9 @@
		#include <audio_effects/effect_downmix.h>
		#include <audio_utils/primitives.h>
		#include <audio_utils/format.h>
		#include <media/AudioResamplerPublic.h>
		#include <media/EffectsFactoryApi.h>

		#include <utils/Log.h>

		#include "Configuration.h"
		@@ -358,5 +360,165 @@ void ReformatBufferProvider::copyFrames(void dst, const void src, size_t frame
		memcpy_by_audio_format(dst, mOutputFormat, src, mInputFormat, frames * mChannelCount);
		}

		TimestretchBufferProvider::TimestretchBufferProvider(int32_t channelCount,
		audio_format_t format, uint32_t sampleRate, float speed, float pitch) :
		mChannelCount(channelCount),
		mFormat(format),
		mSampleRate(sampleRate),
		mFrameSize(channelCount * audio_bytes_per_sample(format)),
		mSpeed(speed),
		mPitch(pitch),
		mLocalBufferFrameCount(0),
		mLocalBufferData(NULL),
		mRemaining(0)
		{
		ALOGV("TimestretchBufferProvider(%p)(%u, %#x, %u %f %f)",
		this, channelCount, format, sampleRate, speed, pitch);
		mBuffer.frameCount = 0;
		}

		TimestretchBufferProvider::~TimestretchBufferProvider()
		{
		ALOGV("~TimestretchBufferProvider(%p)", this);
		if (mBuffer.frameCount != 0) {
		mTrackBufferProvider->releaseBuffer(&mBuffer);
		}
		free(mLocalBufferData);
		}

		status_t TimestretchBufferProvider::getNextBuffer(
		AudioBufferProvider::Buffer *pBuffer, int64_t pts)
		{
		ALOGV("TimestretchBufferProvider(%p)::getNextBuffer(%p (%zu), %lld)",
		this, pBuffer, pBuffer->frameCount, pts);

		// BYPASS
		//return mTrackBufferProvider->getNextBuffer(pBuffer, pts);

		// check if previously processed data is sufficient.
		if (pBuffer->frameCount <= mRemaining) {
		ALOGV("previous sufficient");
		pBuffer->raw = mLocalBufferData;
		return OK;
		}

		// do we need to resize our buffer?
		if (pBuffer->frameCount > mLocalBufferFrameCount) {
		void *newmem;
		if (posix_memalign(&newmem, 32, pBuffer->frameCount * mFrameSize) == OK) {
		if (mRemaining != 0) {
		memcpy(newmem, mLocalBufferData, mRemaining * mFrameSize);
		}
		free(mLocalBufferData);
		mLocalBufferData = newmem;
		mLocalBufferFrameCount = pBuffer->frameCount;
		}
		}

		// need to fetch more data
		const size_t outputDesired = pBuffer->frameCount - mRemaining;
		mBuffer.frameCount = mSpeed == AUDIO_TIMESTRETCH_SPEED_NORMAL
		? outputDesired : outputDesired * mSpeed + 1;

		status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer, pts);

		ALOG_ASSERT(res == OK \|\| mBuffer.frameCount == 0);
		if (res != OK \|\| mBuffer.frameCount == 0) { // not needed by API spec, but to be safe.
		ALOGD("buffer error");
		if (mRemaining == 0) {
		pBuffer->raw = NULL;
		pBuffer->frameCount = 0;
		return res;
		} else { // return partial count
		pBuffer->raw = mLocalBufferData;
		pBuffer->frameCount = mRemaining;
		return OK;
		}
		}

		// time-stretch the data
		size_t dstAvailable = min(mLocalBufferFrameCount - mRemaining, outputDesired);
		size_t srcAvailable = mBuffer.frameCount;
		processFrames((uint8_t)mLocalBufferData + mRemaining mFrameSize, &dstAvailable,
		mBuffer.raw, &srcAvailable);

		// release all data consumed
		mBuffer.frameCount = srcAvailable;
		mTrackBufferProvider->releaseBuffer(&mBuffer);

		// update buffer vars with the actual data processed and return with buffer
		mRemaining += dstAvailable;

		pBuffer->raw = mLocalBufferData;
		pBuffer->frameCount = mRemaining;

		return OK;
		}

		void TimestretchBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer)
		{
		ALOGV("TimestretchBufferProvider(%p)::releaseBuffer(%p (%zu))",
		this, pBuffer, pBuffer->frameCount);

		// BYPASS
		//return mTrackBufferProvider->releaseBuffer(pBuffer);

		// LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount");
		if (pBuffer->frameCount < mRemaining) {
		memcpy(mLocalBufferData,
		(uint8_t)mLocalBufferData + pBuffer->frameCount mFrameSize,
		(mRemaining - pBuffer->frameCount) * mFrameSize);
		mRemaining -= pBuffer->frameCount;
		} else if (pBuffer->frameCount == mRemaining) {
		mRemaining = 0;
		} else {
		LOG_ALWAYS_FATAL("Releasing more frames(%zu) than available(%zu)",
		pBuffer->frameCount, mRemaining);
		}

		pBuffer->raw = NULL;
		pBuffer->frameCount = 0;
		}

		void TimestretchBufferProvider::reset()
		{
		mRemaining = 0;
		}

		status_t TimestretchBufferProvider::setPlaybackRate(float speed, float pitch)
		{
		mSpeed = speed;
		mPitch = pitch;
		return OK;
		}

		void TimestretchBufferProvider::processFrames(void dstBuffer, size_t dstFrames,
		const void srcBuffer, size_t srcFrames)
		{
		ALOGV("processFrames(%zu %zu) remaining(%zu)", dstFrames, srcFrames, mRemaining);
		// Note dstFrames is the required number of frames.

		// Ensure consumption from src is as expected.
		const size_t targetSrc = dstFrames mSpeed;
		if (*srcFrames < targetSrc) { // limit dst frames to that possible
		dstFrames = srcFrames / mSpeed;
		} else if (*srcFrames > targetSrc + 1) {
		*srcFrames = targetSrc + 1;
		}

		// Do the time stretch by memory copy without any local buffer.
		if (dstFrames <= srcFrames) {
		size_t copySize = mFrameSize * *dstFrames;
		memcpy(dstBuffer, srcBuffer, copySize);
		} else {
		// cyclically repeat the source.
		for (size_t count = 0; count < dstFrames; count += srcFrames) {
		size_t remaining = min(srcFrames, dstFrames - count);
		memcpy((uint8_t)dstBuffer + mFrameSize count,
		srcBuffer, mFrameSize * *srcFrames);
		}
		}
		}

		// ----------------------------------------------------------------------------
		} // namespace android

services/audioflinger/BufferProviders.h

+39 −0

Original line number	Diff line number	Diff line
		@@ -146,6 +146,45 @@ protected:
		const audio_format_t mOutputFormat;
		};

		// TimestretchBufferProvider derives from PassthruBufferProvider for time stretching
		class TimestretchBufferProvider : public PassthruBufferProvider {
		public:
		TimestretchBufferProvider(int32_t channelCount,
		audio_format_t format, uint32_t sampleRate, float speed, float pitch);
		virtual ~TimestretchBufferProvider();

		// Overrides AudioBufferProvider methods
		virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
		virtual void releaseBuffer(Buffer* buffer);

		// Overrides PassthruBufferProvider
		virtual void reset();

		virtual status_t setPlaybackRate(float speed, float pitch);

		// processes frames
		// dstBuffer is where to place the data
		// dstFrames [in/out] is the desired frames (return with actual placed in buffer)
		// srcBuffer is the source data
		// srcFrames [in/out] is the available source frames (return with consumed)
		virtual void processFrames(void dstBuffer, size_t dstFrames,
		const void srcBuffer, size_t srcFrames);

		protected:
		const uint32_t mChannelCount;
		const audio_format_t mFormat;
		const uint32_t mSampleRate; // const for now (TODO change this)
		const size_t mFrameSize;
		float mSpeed;
		float mPitch;

		private:
		AudioBufferProvider::Buffer mBuffer;
		size_t mLocalBufferFrameCount;
		void *mLocalBufferData;
		size_t mRemaining;
		};

		// ----------------------------------------------------------------------------
		} // namespace android