blob: cf7d90fe7145870864cb27149532f9adf993b4e8 [file] [log] [blame]
** Copyright 2007, The Android Open Source Project
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** See the License for the specific language governing permissions and
** limitations under the License.
#include <pthread.h>
#include <sstream>
#include <stdint.h>
#include <sys/types.h>
#include <unordered_map>
#include <media/AudioBufferProvider.h>
#include <media/AudioResampler.h>
#include <media/AudioResamplerPublic.h>
#include <media/BufferProviders.h>
#include <media/nblog/NBLog.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
// This must match frameworks/av/services/audioflinger/Configuration.h
#define FLOAT_AUX
namespace android {
// ----------------------------------------------------------------------------
class AudioMixer
// 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 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
// set Parameter names
// for target TRACK
CHANNEL_MASK = 0x4000,
FORMAT = 0x4001,
MAIN_BUFFER = 0x4002,
AUX_BUFFER = 0x4003,
MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
// 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.
AudioMixer(size_t frameCount, uint32_t sampleRate)
: mSampleRate(sampleRate)
, mFrameCount(frameCount) {
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);
void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
void process() {
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) {
return true;
return false;
static inline bool isValidChannelMask(audio_channel_mask_t channelMask) {
return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible.
/* 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
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 {
enum {
// 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 {
: bufferProvider(nullptr)
// TODO: move additional initialization here.
// bufferProvider, mInputBufferProvider need not be deleted.
// Ensure the order of destruction of buffer providers as they
// release the upstream provider in the destructor.
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; };
status_t prepareForDownmix();
void unprepareForDownmix();
status_t prepareForReformat();
void unprepareForReformat();
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.
* 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
* match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
* requires reformat. For example, it may convert floating point input to
* PCM_16_bit if that's required by the downmixer.
* 3) mDownmixerBufferProvider: If not NULL, performs the channel remixing to match
* 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.
std::unique_ptr<PassthruBufferProvider> mReformatBufferProvider;
std::unique_ptr<PassthruBufferProvider> mDownmixerBufferProvider;
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;
// 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;
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();
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);
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
// ----------------------------------------------------------------------------
} // namespace android