Merge changes I52ec5b02,I376656e3,I3f005583,Icba4e33d,I78400f6c, ... am: d7a1506715 am: 7594c6e4d7 (f2c360c3) · Commits · e / os / android_frameworks_av

media/audioserver/Android.mk

+1 −0

Original line number	Diff line number	Diff line
		@@ -9,6 +9,7 @@ LOCAL_SHARED_LIBRARIES := \
		libaaudioservice \
		libaudioflinger \
		libaudiopolicyservice \
		libaudioprocessing \
		libbinder \
		libcutils \
		liblog \

media/libaudioclient/include/media/AudioMixer.h

+34 −315

Original line number	Diff line number	Diff line
		@@ -18,87 +18,38 @@
		#ifndef ANDROID_AUDIO_MIXER_H
		#define ANDROID_AUDIO_MIXER_H

		#include <map>
		#include <pthread.h>
		#include <sstream>
		#include <stdint.h>
		#include <sys/types.h>
		#include <unordered_map>
		#include <vector>

		#include <android/os/IExternalVibratorService.h>
		#include <media/AudioBufferProvider.h>
		#include <media/AudioResampler.h>
		#include <media/AudioResamplerPublic.h>
		#include <media/AudioMixerBase.h>
		#include <media/BufferProviders.h>
		#include <system/audio.h>
		#include <utils/Compat.h>
		#include <utils/threads.h>

		// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
		#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT

		// This must match frameworks/av/services/audioflinger/Configuration.h
		#define FLOAT_AUX
		#define MAX_GAIN_INT AudioMixerBase::UNITY_GAIN_INT

		namespace android {

		namespace NBLog {
		class Writer;
		} // namespace NBLog

		// ----------------------------------------------------------------------------

		class AudioMixer
		// AudioMixer extends AudioMixerBase by adding support for down- and up-mixing
		// and time stretch that are implemented via Effects HAL, and adding support
		// for haptic channels which depends on Vibrator service. This is the version
		// that is used by Audioflinger.

		class AudioMixer : public AudioMixerBase
		{
		public:
		// Do not change these unless underlying code changes.
		// This mixer has a hard-coded upper limit of 8 channels for output.
		static constexpr uint32_t MAX_NUM_CHANNELS = FCC_8;
		static constexpr uint32_t MAX_NUM_VOLUMES = FCC_2; // stereo volume only
		// maximum number of channels supported for the content
		static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;

		static const uint16_t UNITY_GAIN_INT = 0x1000;
		static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;

		enum { // names
		// setParameter targets
		TRACK = 0x3000,
		RESAMPLE = 0x3001,
		RAMP_VOLUME = 0x3002, // ramp to new volume
		VOLUME = 0x3003, // don't ramp
		TIMESTRETCH = 0x3004,

		// set Parameter names
		// for target TRACK
		CHANNEL_MASK = 0x4000,
		FORMAT = 0x4001,
		MAIN_BUFFER = 0x4002,
		AUX_BUFFER = 0x4003,
		DOWNMIX_TYPE = 0X4004,
		MIXER_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT\|16_BIT)
		MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
		enum { // extension of AudioMixerBase parameters
		DOWNMIX_TYPE = 0x4004,
		// for haptic
		HAPTIC_ENABLED = 0x4007, // Set haptic data from this track should be played or not.
		HAPTIC_INTENSITY = 0x4008, // Set the intensity to play haptic data.
		// for target RESAMPLE
		SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
		// parameter 'value' is the new sample rate in Hz.
		// Only creates a sample rate converter the first time that
		// the track sample rate is different from the mix sample rate.
		// If the new sample rate is the same as the mix sample rate,
		// and a sample rate converter already exists,
		// then the sample rate converter remains present but is a no-op.
		RESET = 0x4101, // Reset sample rate converter without changing sample rate.
		// This clears out the resampler's input buffer.
		REMOVE = 0x4102, // Remove the sample rate converter on this track name;
		// the track is restored to the mix sample rate.
		// for target RAMP_VOLUME and VOLUME (8 channels max)
		// FIXME use float for these 3 to improve the dynamic range
		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.
		@@ -131,142 +82,23 @@ public:
		}

		AudioMixer(size_t frameCount, uint32_t sampleRate)
		: mSampleRate(sampleRate)
		, mFrameCount(frameCount) {
		: AudioMixerBase(frameCount, sampleRate) {
		pthread_once(&sOnceControl, &sInitRoutine);
		}

		// Create a new track in the mixer.
		//
		// \param name a unique user-provided integer associated with the track.
		// If name already exists, the function will abort.
		// \param channelMask output channel mask.
		// \param format PCM format
		// \param sessionId Session id for the track. Tracks with the same
		// session id will be submixed together.
		//
		// \return OK on success.
		// BAD_VALUE if the format does not satisfy isValidFormat()
		// or the channelMask does not satisfy isValidChannelMask().
		status_t create(
		int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId);

		bool exists(int name) const {
		return mTracks.count(name) > 0;
		}

		// Free an allocated track by name.
		void destroy(int name);

		// Enable or disable an allocated track by name
		void enable(int name);
		void disable(int name);

		void setParameter(int name, int target, int param, void *value);
		bool isValidChannelMask(audio_channel_mask_t channelMask) const override;

		void setParameter(int name, int target, int param, void *value) override;
		void setBufferProvider(int name, AudioBufferProvider* bufferProvider);

		void process() {
		for (const auto &pair : mTracks) {
		// Clear contracted buffer before processing if contracted channels are saved
		const std::shared_ptr<Track> &t = pair.second;
		if (t->mKeepContractedChannels) {
		t->clearContractedBuffer();
		}
		}
		(this->*mHook)();
		processHapticData();
		}

		size_t getUnreleasedFrames(int name) const;

		std::string trackNames() const {
		std::stringstream ss;
		for (const auto &pair : mTracks) {
		ss << pair.first << " ";
		}
		return ss.str();
		}

		void setNBLogWriter(NBLog::Writer *logWriter) {
		mNBLogWriter = logWriter;
		}

		static inline bool isValidFormat(audio_format_t format) {
		switch (format) {
		case AUDIO_FORMAT_PCM_8_BIT:
		case AUDIO_FORMAT_PCM_16_BIT:
		case AUDIO_FORMAT_PCM_24_BIT_PACKED:
		case AUDIO_FORMAT_PCM_32_BIT:
		case AUDIO_FORMAT_PCM_FLOAT:
		return true;
		default:
		return false;
		}
		}

		static inline bool isValidChannelMask(audio_channel_mask_t channelMask) {
		return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible.
		}

		private:

		/* For multi-format functions (calls template functions
		* in AudioMixerOps.h). The template parameters are as follows:
		*
		* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
		* USEFLOATVOL (set to true if float volume is used)
		* ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
		* TO: int32_t (Q4.27) or float
		* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
		* TA: int32_t (Q4.27)
		*/

		enum {
		// FIXME this representation permits up to 8 channels
		NEEDS_CHANNEL_COUNT__MASK = 0x00000007,
		};

		enum {
		NEEDS_CHANNEL_1 = 0x00000000, // mono
		NEEDS_CHANNEL_2 = 0x00000001, // stereo

		// sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT

		NEEDS_MUTE = 0x00000100,
		NEEDS_RESAMPLE = 0x00001000,
		NEEDS_AUX = 0x00010000,
		};

		// hook types
		enum {
		PROCESSTYPE_NORESAMPLEONETRACK, // others set elsewhere
		};

		enum {
		TRACKTYPE_NOP,
		TRACKTYPE_RESAMPLE,
		TRACKTYPE_NORESAMPLE,
		TRACKTYPE_NORESAMPLEMONO,
		};

		// process hook functionality
		using process_hook_t = void(AudioMixer::*)();

		struct Track;
		using hook_t = void(Track::)(int32_t output, size_t numOutFrames, int32_t* temp, int32_t* aux);

		struct Track {
		Track()
		: bufferProvider(nullptr)
		{
		// TODO: move additional initialization here.
		}
		struct Track : public TrackBase {
		Track() : TrackBase() {}

		~Track()
		{
		// bufferProvider, mInputBufferProvider need not be deleted.
		mResampler.reset(nullptr);
		// mInputBufferProvider need not be deleted.
		// Ensure the order of destruction of buffer providers as they
		// release the upstream provider in the destructor.
		mTimestretchBufferProvider.reset(nullptr);
		@@ -277,13 +109,12 @@ private:
		mAdjustChannelsBufferProvider.reset(nullptr);
		}

		bool needsRamp() { return (volumeInc[0] \| volumeInc[1] \| auxInc) != 0; }
		bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
		bool doesResample() const { return mResampler.get() != nullptr; }
		void resetResampler() { if (mResampler.get() != nullptr) mResampler->reset(); }
		void adjustVolumeRamp(bool aux, bool useFloat = false);
		size_t getUnreleasedFrames() const { return mResampler.get() != nullptr ?
		mResampler->getUnreleasedFrames() : 0; };
		uint32_t getOutputChannelCount() override {
		return mDownmixerBufferProvider.get() != nullptr ? mMixerChannelCount : channelCount;
		}
		uint32_t getMixerChannelCount() override {
		return mMixerChannelCount + mMixerHapticChannelCount;
		}

		status_t prepareForDownmix();
		void unprepareForDownmix();
		@@ -297,51 +128,9 @@ private:
		bool setPlaybackRate(const AudioPlaybackRate &playbackRate);
		void reconfigureBufferProviders();

		static hook_t getTrackHook(int trackType, uint32_t channelCount,
		audio_format_t mixerInFormat, audio_format_t mixerOutFormat);

		void track__nop(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);

		template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
		typename TO, typename TI, typename TA>
		void volumeMix(TO out, size_t outFrames, const TI in, TA *aux, bool ramp);

		uint32_t needs;

		// TODO: Eventually remove legacy integer volume settings
		union {
		int16_t volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
		int32_t volumeRL;
		};

		int32_t prevVolume[MAX_NUM_VOLUMES];
		int32_t volumeInc[MAX_NUM_VOLUMES];
		int32_t auxInc;
		int32_t prevAuxLevel;
		int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance

		uint16_t frameCount;

		uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
		uint8_t unused_padding; // formerly format, was always 16
		uint16_t enabled; // actually bool
		audio_channel_mask_t channelMask;

		// actual buffer provider used by the track hooks, see DownmixerBufferProvider below
		// for how the Track buffer provider is wrapped by another one when dowmixing is required
		AudioBufferProvider* bufferProvider;

		mutable AudioBufferProvider::Buffer buffer; // 8 bytes

		hook_t hook;
		const void *mIn; // current location in buffer

		std::unique_ptr<AudioResampler> mResampler;
		uint32_t sampleRate;
		int32_t* mainBuffer;
		int32_t* auxBuffer;

		/* Buffer providers are constructed to translate the track input data as needed.
		* See DownmixerBufferProvider below for how the Track buffer provider
		* is wrapped by another one when dowmixing is required.
		*
		* TODO: perhaps make a single PlaybackConverterProvider class to move
		* all pre-mixer track buffer conversions outside the AudioMixer class.
		@@ -373,27 +162,10 @@ private:
		std::unique_ptr<PassthruBufferProvider> mPostDownmixReformatBufferProvider;
		std::unique_ptr<PassthruBufferProvider> mTimestretchBufferProvider;

		int32_t sessionId;

		audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT\|16_BIT)
		audio_format_t mFormat; // input track format
		audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT\|16_BIT)
		// each track must be converted to this format.
		audio_format_t mDownmixRequiresFormat; // required downmixer format
		// AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
		// AUDIO_FORMAT_INVALID if no required format

		float mVolume[MAX_NUM_VOLUMES]; // floating point set volume
		float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
		float mVolumeInc[MAX_NUM_VOLUMES]; // floating point volume increment

		float mAuxLevel; // floating point set aux level
		float mPrevAuxLevel; // floating point prev aux level
		float mAuxInc; // floating point aux increment

		audio_channel_mask_t mMixerChannelMask;
		uint32_t mMixerChannelCount;

		AudioPlaybackRate mPlaybackRate;

		// Haptic
		@@ -440,76 +212,23 @@ private:
		return 0.0f;
		}
		}

		private:
		// hooks
		void track__genericResample(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
		void track__16BitsStereo(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
		void track__16BitsMono(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);

		void volumeRampStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
		void volumeStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);

		// multi-format track hooks
		template <int MIXTYPE, typename TO, typename TI, typename TA>
		void track__Resample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
		template <int MIXTYPE, typename TO, typename TI, typename TA>
		void track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
		};

		// TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore.
		static constexpr int BLOCKSIZE = 16;

		bool setChannelMasks(int name,
		audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);

		// Called when track info changes and a new process hook should be determined.
		void invalidate() {
		mHook = &AudioMixer::process__validate;
		inline std::shared_ptr<Track> getTrack(int name) {
		return std::static_pointer_cast<Track>(mTracks[name]);
		}

		void process__validate();
		void process__nop();
		void process__genericNoResampling();
		void process__genericResampling();
		void process__oneTrack16BitsStereoNoResampling();

		template <int MIXTYPE, typename TO, typename TI, typename TA>
		void process__noResampleOneTrack();
		std::shared_ptr<TrackBase> preCreateTrack() override;
		status_t postCreateTrack(TrackBase *track) override;

		void processHapticData();
		void preProcess() override;
		void postProcess() override;

		static process_hook_t getProcessHook(int processType, uint32_t channelCount,
		audio_format_t mixerInFormat, audio_format_t mixerOutFormat);

		static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
		void *in, audio_format_t mixerInFormat, size_t sampleCount);
		bool setChannelMasks(int name,
		audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) override;

		static void sInitRoutine();

		// initialization constants
		const uint32_t mSampleRate;
		const size_t mFrameCount;

		NBLog::Writer *mNBLogWriter = nullptr; // associated NBLog::Writer

		process_hook_t mHook = &AudioMixer::process__nop; // one of process__*, never nullptr

		// the size of the type (int32_t) should be the largest of all types supported
		// by the mixer.
		std::unique_ptr<int32_t[]> mOutputTemp;
		std::unique_ptr<int32_t[]> mResampleTemp;

		// track names grouped by main buffer, in no particular order of main buffer.
		// however names for a particular main buffer are in order (by construction).
		std::unordered_map<void * /* mainBuffer /, std::vector<int / name */>> mGroups;

		// track names that are enabled, in increasing order (by construction).
		std::vector<int /* name */> mEnabled;

		// track smart pointers, by name, in increasing order of name.
		std::map<int /* name */, std::shared_ptr<Track>> mTracks;

		static pthread_once_t sOnceControl; // initialized in constructor by first new
		};

media/libaudioprocessing/Android.bp

+18 −12

Original line number	Diff line number	Diff line
		@@ -3,20 +3,13 @@ cc_defaults {

		export_include_dirs: ["include"],

		header_libs: ["libaudioclient_headers"],

		shared_libs: [
		"libaudiohal",
		"libaudioutils",
		"libcutils",
		"liblog",
		"libnbaio",
		"libnblog",
		"libsonic",
		"libutils",
		"libvibrator",
		],

		header_libs: [
		"libbase_headers",
		],

		cflags: [
		@@ -33,18 +26,31 @@ cc_library_shared {
		defaults: ["libaudioprocessing_defaults"],

		srcs: [
		"AudioMixer.cpp",
		"BufferProviders.cpp",
		"RecordBufferConverter.cpp",
		],
		whole_static_libs: ["libaudioprocessing_arm"],

		header_libs: [
		"libbase_headers",
		],

		shared_libs: [
		"libaudiohal",
		"libsonic",
		"libvibrator",
		],

		whole_static_libs: ["libaudioprocessing_base"],
		}

		cc_library_static {
		name: "libaudioprocessing_arm",
		name: "libaudioprocessing_base",
		defaults: ["libaudioprocessing_defaults"],
		vendor_available: true,

		srcs: [
		"AudioMixer.cpp",
		"AudioMixerBase.cpp",
		"AudioResampler.cpp",
		"AudioResamplerCubic.cpp",
		"AudioResamplerSinc.cpp",

media/libaudioprocessing/AudioMixer.cpp

+82 −1575

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media/libaudioprocessing/AudioMixerBase.cpp

0 → 100644

+1692 −0

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