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** 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 <stdint.h>
#include <sys/types.h>
#include <utils/threads.h>
#include <media/AudioBufferProvider.h>
#include "AudioResampler.h"
#include <hardware/audio_effect.h>
#include <system/audio.h>
#include <media/nbaio/NBLog.h>
// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
namespace android {
// ----------------------------------------------------------------------------
class AudioMixer
AudioMixer(size_t frameCount, uint32_t sampleRate,
uint32_t maxNumTracks = MAX_NUM_TRACKS);
/*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed
// This mixer has a hard-coded upper limit of 32 active track inputs.
// Adding support for > 32 tracks would require more than simply changing this value.
static const uint32_t MAX_NUM_TRACKS = 32;
// maximum number of channels supported by the mixer
// This mixer has a hard-coded upper limit of 8 channels for output.
static const uint32_t MAX_NUM_CHANNELS = 8;
static const uint32_t MAX_NUM_VOLUMES = 2; // stereo volume only
// maximum number of channels supported for the content
static const uint16_t UNITY_GAIN_INT = 0x1000;
static const float UNITY_GAIN_FLOAT = 1.0f;
enum { // names
// track names (MAX_NUM_TRACKS units)
TRACK0 = 0x1000,
// 0x2000 is unused
// 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 all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
// Allocate a track name. Returns new track name if successful, -1 on failure.
// The failure could be because of an invalid channelMask or format, or that
// the track capacity of the mixer is exceeded.
int getTrackName(audio_channel_mask_t channelMask,
audio_format_t format, int sessionId);
// Free an allocated track by name
void deleteTrackName(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(int64_t pts);
uint32_t trackNames() const { return mTrackNames; }
size_t getUnreleasedFrames(int name) const;
static inline bool isValidPcmTrackFormat(audio_format_t format) {
return format == AUDIO_FORMAT_PCM_16_BIT ||
format == AUDIO_FORMAT_PCM_32_BIT ||
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,
struct state_t;
struct track_t;
class CopyBufferProvider;
typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
int32_t* aux);
static const int BLOCKSIZE = 16; // 4 cache lines
struct track_t {
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];
// 16-byte boundary
int32_t volumeInc[MAX_NUM_VOLUMES];
int32_t auxInc;
int32_t prevAuxLevel;
// 16-byte boundary
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;
// 16-byte boundary
mutable AudioBufferProvider::Buffer buffer; // 8 bytes
hook_t hook;
const void* in; // current location in buffer
// 16-byte boundary
AudioResampler* resampler;
uint32_t sampleRate;
int32_t* mainBuffer;
int32_t* auxBuffer;
// 16-byte boundary
AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider.
CopyBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting.
CopyBufferProvider* downmixerBufferProvider; // wrapper for channel conversion.
int32_t sessionId;
// 16-byte boundary
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
// 16-byte boundary
audio_channel_mask_t mMixerChannelMask;
uint32_t mMixerChannelCount;
bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
bool doesResample() const { return resampler != NULL; }
void resetResampler() { if (resampler != NULL) resampler->reset(); }
void adjustVolumeRamp(bool aux, bool useFloat = false);
size_t getUnreleasedFrames() const { return resampler != NULL ?
resampler->getUnreleasedFrames() : 0; };
typedef void (*process_hook_t)(state_t* state, int64_t pts);
// pad to 32-bytes to fill cache line
struct state_t {
uint32_t enabledTracks;
uint32_t needsChanged;
size_t frameCount;
process_hook_t hook; // one of process__*, never NULL
int32_t *outputTemp;
int32_t *resampleTemp;
NBLog::Writer* mLog;
int32_t reserved[1];
// FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
track_t tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
// Base AudioBufferProvider class used for DownMixerBufferProvider, RemixBufferProvider,
// and ReformatBufferProvider.
// It handles a private buffer for use in converting format or channel masks from the
// input data to a form acceptable by the mixer.
// TODO: Make a ResamplerBufferProvider when integers are entirely removed from the
// processing pipeline.
class CopyBufferProvider : public AudioBufferProvider {
// Use a private buffer of bufferFrameCount frames (each frame is outputFrameSize bytes).
// If bufferFrameCount is 0, no private buffer is created and in-place modification of
// the upstream buffer provider's buffers is performed by copyFrames().
CopyBufferProvider(size_t inputFrameSize, size_t outputFrameSize,
size_t bufferFrameCount);
virtual ~CopyBufferProvider();
// Overrides AudioBufferProvider methods
virtual status_t getNextBuffer(Buffer* buffer, int64_t pts);
virtual void releaseBuffer(Buffer* buffer);
// Other public methods
// call this to release the buffer to the upstream provider.
// treat it as an audio discontinuity for future samples.
virtual void reset();
// this function should be supplied by the derived class. It converts
// #frames in the *src pointer to the *dst pointer. It is public because
// some providers will allow this to work on arbitrary buffers outside
// of the internal buffers.
virtual void copyFrames(void *dst, const void *src, size_t frames) = 0;
// set the upstream buffer provider. Consider calling "reset" before this function.
void setBufferProvider(AudioBufferProvider *p) {
mTrackBufferProvider = p;
AudioBufferProvider* mTrackBufferProvider;
const size_t mInputFrameSize;
const size_t mOutputFrameSize;
AudioBufferProvider::Buffer mBuffer;
const size_t mLocalBufferFrameCount;
void* mLocalBufferData;
size_t mConsumed;
// DownmixerBufferProvider wraps a track AudioBufferProvider to provide
// position dependent downmixing by an Audio Effect.
class DownmixerBufferProvider : public CopyBufferProvider {
DownmixerBufferProvider(audio_channel_mask_t inputChannelMask,
audio_channel_mask_t outputChannelMask, audio_format_t format,
uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount);
virtual ~DownmixerBufferProvider();
virtual void copyFrames(void *dst, const void *src, size_t frames);
bool isValid() const { return mDownmixHandle != NULL; }
static status_t init();
static bool isMultichannelCapable() { return sIsMultichannelCapable; }
effect_handle_t mDownmixHandle;
effect_config_t mDownmixConfig;
// effect descriptor for the downmixer used by the mixer
static effect_descriptor_t sDwnmFxDesc;
// indicates whether a downmix effect has been found and is usable by this mixer
static bool sIsMultichannelCapable;
// FIXME: should we allow effects outside of the framework?
// We need to here. A special ioId that must be <= -2 so it does not map to a session.
static const int32_t SESSION_ID_INVALID_AND_IGNORED = -2;
// RemixBufferProvider wraps a track AudioBufferProvider to perform an
// upmix or downmix to the proper channel count and mask.
class RemixBufferProvider : public CopyBufferProvider {
RemixBufferProvider(audio_channel_mask_t inputChannelMask,
audio_channel_mask_t outputChannelMask, audio_format_t format,
size_t bufferFrameCount);
virtual void copyFrames(void *dst, const void *src, size_t frames);
const audio_format_t mFormat;
const size_t mSampleSize;
const size_t mInputChannels;
const size_t mOutputChannels;
int8_t mIdxAry[sizeof(uint32_t)*8]; // 32 bits => channel indices
// ReformatBufferProvider wraps a track AudioBufferProvider to convert the input data
// to an acceptable mixer input format type.
class ReformatBufferProvider : public CopyBufferProvider {
ReformatBufferProvider(int32_t channels,
audio_format_t inputFormat, audio_format_t outputFormat,
size_t bufferFrameCount);
virtual void copyFrames(void *dst, const void *src, size_t frames);
const int32_t mChannels;
const audio_format_t mInputFormat;
const audio_format_t mOutputFormat;
// bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
uint32_t mTrackNames;
// bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
// but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
const uint32_t mConfiguredNames;
const uint32_t mSampleRate;
NBLog::Writer mDummyLog;
void setLog(NBLog::Writer* log);
state_t mState __attribute__((aligned(32)));
// Call after changing either the enabled status of a track, or parameters of an enabled track.
// OK to call more often than that, but unnecessary.
void invalidateState(uint32_t mask);
bool setChannelMasks(int name,
audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
// TODO: remove unused trackName/trackNum from functions below.
static status_t initTrackDownmix(track_t* pTrack, int trackName);
static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
static status_t prepareTrackForReformat(track_t* pTrack, int trackNum);
static void unprepareTrackForReformat(track_t* pTrack, int trackName);
static void reconfigureBufferProviders(track_t* pTrack);
static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
int32_t* aux);
static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
int32_t* aux);
static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
int32_t* aux);
static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
int32_t* aux);
static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
int32_t* aux);
static void process__validate(state_t* state, int64_t pts);
static void process__nop(state_t* state, int64_t pts);
static void process__genericNoResampling(state_t* state, int64_t pts);
static void process__genericResampling(state_t* state, int64_t pts);
static void process__OneTrack16BitsStereoNoResampling(state_t* state,
int64_t pts);
static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
int outputFrameIndex);
static uint64_t sLocalTimeFreq;
static pthread_once_t sOnceControl;
static void sInitRoutine();
/* multi-format volume mixing function (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)
template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
typename TO, typename TI, typename TA>
static void volumeMix(TO *out, size_t outFrames,
const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t);
// multi-format process hooks
template <int MIXTYPE, typename TO, typename TI, typename TA>
static void process_NoResampleOneTrack(state_t* state, int64_t pts);
// multi-format track hooks
template <int MIXTYPE, typename TO, typename TI, typename TA>
static void track__Resample(track_t* t, TO* out, size_t frameCount,
TO* temp __unused, TA* aux);
template <int MIXTYPE, typename TO, typename TI, typename TA>
static void track__NoResample(track_t* t, TO* out, size_t frameCount,
TO* temp __unused, TA* aux);
static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
void *in, audio_format_t mixerInFormat, size_t sampleCount);
// hook types
enum {
enum {
// functions for determining the proper process and track hooks.
static process_hook_t getProcessHook(int processType, uint32_t channelCount,
audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
static hook_t getTrackHook(int trackType, uint32_t channelCount,
audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
// ----------------------------------------------------------------------------
}; // namespace android