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** Copyright 2019, 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 <map>
#include <memory>
#include <string>
#include <unordered_map>
#include <vector>
#include <media/AudioBufferProvider.h>
#include <media/AudioResampler.h>
#include <media/AudioResamplerPublic.h>
#include <system/audio.h>
#include <utils/Compat.h>
// This must match frameworks/av/services/audioflinger/Configuration.h
// when used with the Audio Framework.
#define FLOAT_AUX
namespace android {
// ----------------------------------------------------------------------------
// AudioMixerBase is functional on its own if only mixing and resampling
// is needed.
class AudioMixerBase
// 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
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,
// 0x4004 reserved
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,
AudioMixerBase(size_t frameCount, uint32_t sampleRate)
: mSampleRate(sampleRate)
, mFrameCount(frameCount) {
virtual ~AudioMixerBase() {}
virtual bool isValidFormat(audio_format_t format) const;
virtual bool isValidChannelMask(audio_channel_mask_t channelMask) const;
// 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);
virtual void setParameter(int name, int target, int param, void *value);
void process() {
size_t getUnreleasedFrames(int name) const;
std::string trackNames() const;
// Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
// original code will be used for stereo sinks, the new mixer for everything else.
static constexpr bool kUseNewMixer = true;
// Set kUseFloat to true to allow floating input into the mixer engine.
// If kUseNewMixer is false, this is ignored or may be overridden internally
static constexpr bool kUseFloat = true;
#ifdef FLOAT_AUX
using TYPE_AUX = float;
static_assert(kUseNewMixer && kUseFloat,
"kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
using TYPE_AUX = int32_t; // q4.27
/* 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(AudioMixerBase::*)();
struct TrackBase;
using hook_t = void(TrackBase::*)(
int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
struct TrackBase {
: bufferProvider(nullptr)
// TODO: move additional initialization here.
virtual ~TrackBase() {}
virtual uint32_t getOutputChannelCount() { return channelCount; }
virtual uint32_t getMixerChannelCount() { return mMixerChannelCount; }
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 recreateResampler(uint32_t devSampleRate);
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; };
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
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;
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.
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;
// 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);
// preCreateTrack must create an instance of a proper TrackBase descendant.
// postCreateTrack is called after filling out fields of TrackBase. It can
// abort track creation by returning non-OK status. See the implementation
// of create() for details.
virtual std::shared_ptr<TrackBase> preCreateTrack();
virtual status_t postCreateTrack(TrackBase *track __unused) { return OK; }
// preProcess is called before the process hook, postProcess after,
// see the implementation of process() method.
virtual void preProcess() {}
virtual void postProcess() {}
virtual 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 = &AudioMixerBase::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);
// initialization constants
const uint32_t mSampleRate;
const size_t mFrameCount;
process_hook_t mHook = &AudioMixerBase::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<TrackBase>> mTracks;
} // namespace android