/* //device/include/server/AudioFlinger/AudioMixer.cpp
**
** 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
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
#define LOG_TAG "AudioMixer"
//#define LOG_NDEBUG 0
#include <stdint.h>
#include <string.h>
#include <stdlib.h>
#include <sys/types.h>
#include <utils/Errors.h>
#include <utils/Log.h>
#include "AudioMixer.h"
namespace android {
// ----------------------------------------------------------------------------
static inline int16_t clamp16(int32_t sample)
{
if ((sample>>15) ^ (sample>>31))
sample = 0x7FFF ^ (sample>>31);
return sample;
}
// ----------------------------------------------------------------------------
AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
: mActiveTrack(0), mTrackNames(0), mSampleRate(sampleRate)
{
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
mState.outputTemp = 0;
mState.resampleTemp = 0;
mState.hook = process__nop;
track_t* t = mState.tracks;
for (int i=0 ; i<32 ; i++) {
t->needs = 0;
t->volume[0] = UNITY_GAIN;
t->volume[1] = UNITY_GAIN;
t->volumeInc[0] = 0;
t->volumeInc[1] = 0;
t->auxLevel = 0;
t->auxInc = 0;
t->channelCount = 2;
t->enabled = 0;
t->format = 16;
t->buffer.raw = 0;
t->bufferProvider = 0;
t->hook = 0;
t->resampler = 0;
t->sampleRate = mSampleRate;
t->in = 0;
t->mainBuffer = NULL;
t->auxBuffer = NULL;
t++;
}
}
AudioMixer::~AudioMixer()
{
track_t* t = mState.tracks;
for (int i=0 ; i<32 ; i++) {
delete t->resampler;
t++;
}
delete [] mState.outputTemp;
delete [] mState.resampleTemp;
}
int AudioMixer::getTrackName()
{
uint32_t names = mTrackNames;
uint32_t mask = 1;
int n = 0;
while (names & mask) {
mask <<= 1;
n++;
}
if (mask) {
LOGV("add track (%d)", n);
mTrackNames |= mask;
return TRACK0 + n;
}
return -1;
}
void AudioMixer::invalidateState(uint32_t mask)
{
if (mask) {
mState.needsChanged |= mask;
mState.hook = process__validate;
}
}
void AudioMixer::deleteTrackName(int name)
{
name -= TRACK0;
if (uint32_t(name) < MAX_NUM_TRACKS) {
LOGV("deleteTrackName(%d)", name);
track_t& track(mState.tracks[ name ]);
if (track.enabled != 0) {
track.enabled = 0;
invalidateState(1<<name);
}
if (track.resampler) {
// delete the resampler
delete track.resampler;
track.resampler = 0;
track.sampleRate = mSampleRate;
invalidateState(1<<name);
}
track.volumeInc[0] = 0;
track.volumeInc[1] = 0;
mTrackNames &= ~(1<<name);
}
}
status_t AudioMixer::enable(int name)
{
switch (name) {
case MIXING: {
if (mState.tracks[ mActiveTrack ].enabled != 1) {
mState.tracks[ mActiveTrack ].enabled = 1;
LOGV("enable(%d)", mActiveTrack);
invalidateState(1<<mActiveTrack);
}
} break;
default:
return NAME_NOT_FOUND;
}
return NO_ERROR;
}
status_t AudioMixer::disable(int name)
{
switch (name) {
case MIXING: {
if (mState.tracks[ mActiveTrack ].enabled != 0) {
mState.tracks[ mActiveTrack ].enabled = 0;
LOGV("disable(%d)", mActiveTrack);
invalidateState(1<<mActiveTrack);
}
} break;
default:
return NAME_NOT_FOUND;
}
return NO_ERROR;
}
status_t AudioMixer::setActiveTrack(int track)
{
if (uint32_t(track-TRACK0) >= MAX_NUM_TRACKS) {
return BAD_VALUE;
}
mActiveTrack = track - TRACK0;
return NO_ERROR;
}
status_t AudioMixer::setParameter(int target, int name, void *value)
{
int valueInt = (int)value;
int32_t *valueBuf = (int32_t *)value;
switch (target) {
case TRACK:
if (name == CHANNEL_COUNT) {
if ((uint32_t(valueInt) <= MAX_NUM_CHANNELS) && (valueInt)) {
if (mState.tracks[ mActiveTrack ].channelCount != valueInt) {
mState.tracks[ mActiveTrack ].channelCount = valueInt;
LOGV("setParameter(TRACK, CHANNEL_COUNT, %d)", valueInt);
invalidateState(1<<mActiveTrack);
}
return NO_ERROR;
}
}
if (name == MAIN_BUFFER) {
if (mState.tracks[ mActiveTrack ].mainBuffer != valueBuf) {
mState.tracks[ mActiveTrack ].mainBuffer = valueBuf;
LOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
invalidateState(1<<mActiveTrack);
}
return NO_ERROR;
}
if (name == AUX_BUFFER) {
if (mState.tracks[ mActiveTrack ].auxBuffer != valueBuf) {
mState.tracks[ mActiveTrack ].auxBuffer = valueBuf;
LOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
invalidateState(1<<mActiveTrack);
}
return NO_ERROR;
}
break;
case RESAMPLE:
if (name == SAMPLE_RATE) {
if (valueInt > 0) {
track_t& track = mState.tracks[ mActiveTrack ];
if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
LOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
uint32_t(valueInt));
invalidateState(1<<mActiveTrack);
}
return NO_ERROR;
}
}
break;
case RAMP_VOLUME:
case VOLUME:
if ((uint32_t(name-VOLUME0) < MAX_NUM_CHANNELS)) {
track_t& track = mState.tracks[ mActiveTrack ];
if (track.volume[name-VOLUME0] != valueInt) {
LOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt);
track.prevVolume[name-VOLUME0] = track.volume[name-VOLUME0] << 16;
track.volume[name-VOLUME0] = valueInt;
if (target == VOLUME) {
track.prevVolume[name-VOLUME0] = valueInt << 16;
track.volumeInc[name-VOLUME0] = 0;
} else {
int32_t d = (valueInt<<16) - track.prevVolume[name-VOLUME0];
int32_t volInc = d / int32_t(mState.frameCount);
track.volumeInc[name-VOLUME0] = volInc;
if (volInc == 0) {
track.prevVolume[name-VOLUME0] = valueInt << 16;
}
}
invalidateState(1<<mActiveTrack);
}
return NO_ERROR;
} else if (name == AUXLEVEL) {
track_t& track = mState.tracks[ mActiveTrack ];
if (track.auxLevel != valueInt) {
LOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt);
track.prevAuxLevel = track.auxLevel << 16;
track.auxLevel = valueInt;
if (target == VOLUME) {
track.prevAuxLevel = valueInt << 16;
track.auxInc = 0;
} else {
int32_t d = (valueInt<<16) - track.prevAuxLevel;
int32_t volInc = d / int32_t(mState.frameCount);
track.auxInc = volInc;
if (volInc == 0) {
track.prevAuxLevel = valueInt << 16;
}
}
invalidateState(1<<mActiveTrack);
}
return NO_ERROR;
}
break;
}
return BAD_VALUE;
}
bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
{
if (value!=devSampleRate || resampler) {
if (sampleRate != value) {
sampleRate = value;
if (resampler == 0) {
resampler = AudioResampler::create(
format, channelCount, devSampleRate);
}
return true;
}
}
return false;
}
bool AudioMixer::track_t::doesResample() const
{
return resampler != 0;
}
inline
void AudioMixer::track_t::adjustVolumeRamp(bool aux)
{
for (int i=0 ; i<2 ; i++) {
if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
volumeInc[i] = 0;
prevVolume[i] = volume[i]<<16;
}
}
if (aux) {
if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) ||
((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) {
auxInc = 0;
prevAuxLevel = auxLevel<<16;
}
}
}
status_t AudioMixer::setBufferProvider(AudioBufferProvider* buffer)
{
mState.tracks[ mActiveTrack ].bufferProvider = buffer;
return NO_ERROR;
}
void AudioMixer::process()
{
mState.hook(&mState);
}
void AudioMixer::process__validate(state_t* state)
{
LOGW_IF(!state->needsChanged,
"in process__validate() but nothing's invalid");
uint32_t changed = state->needsChanged;
state->needsChanged = 0; // clear the validation flag
// recompute which tracks are enabled / disabled
uint32_t enabled = 0;
uint32_t disabled = 0;
while (changed) {
const int i = 31 - __builtin_clz(changed);
const uint32_t mask = 1<<i;
changed &= ~mask;
track_t& t = state->tracks[i];
(t.enabled ? enabled : disabled) |= mask;
}
state->enabledTracks &= ~disabled;
state->enabledTracks |= enabled;
// compute everything we need...
int countActiveTracks = 0;
int all16BitsStereoNoResample = 1;
int resampling = 0;
int volumeRamp = 0;
uint32_t en = state->enabledTracks;
while (en) {
const int i = 31 - __builtin_clz(en);
en &= ~(1<<i);
countActiveTracks++;
track_t& t = state->tracks[i];
uint32_t n = 0;
n |= NEEDS_CHANNEL_1 + t.channelCount - 1;
n |= NEEDS_FORMAT_16;
n |= t.doesResample() ? NEEDS_RESAMPLE_ENABLED : NEEDS_RESAMPLE_DISABLED;
if (t.auxLevel != 0 && t.auxBuffer != NULL) {
n |= NEEDS_AUX_ENABLED;
}
if (t.volumeInc[0]|t.volumeInc[1]) {
volumeRamp = 1;
} else if (!t.doesResample() && t.volumeRL == 0) {
n |= NEEDS_MUTE_ENABLED;
}
t.needs = n;
if ((n & NEEDS_MUTE__MASK) == NEEDS_MUTE_ENABLED) {
t.hook = track__nop;
} else {
if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
all16BitsStereoNoResample = 0;
}
if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
all16BitsStereoNoResample = 0;
resampling = 1;
t.hook = track__genericResample;
} else {
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
t.hook = track__16BitsMono;
all16BitsStereoNoResample = 0;
}
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_2){
t.hook = track__16BitsStereo;
}
}
}
}
// select the processing hooks
state->hook = process__nop;
if (countActiveTracks) {
if (resampling) {
if (!state->outputTemp) {
state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
}
if (!state->resampleTemp) {
state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
}
state->hook = process__genericResampling;
} else {
if (state->outputTemp) {
delete [] state->outputTemp;
state->outputTemp = 0;
}
if (state->resampleTemp) {
delete [] state->resampleTemp;
state->resampleTemp = 0;
}
state->hook = process__genericNoResampling;
if (all16BitsStereoNoResample && !volumeRamp) {
if (countActiveTracks == 1) {
state->hook = process__OneTrack16BitsStereoNoResampling;
}
}
}
}
LOGV("mixer configuration change: %d activeTracks (%08x) "
"all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
countActiveTracks, state->enabledTracks,
all16BitsStereoNoResample, resampling, volumeRamp);
state->hook(state);
// Now that the volume ramp has been done, set optimal state and
// track hooks for subsequent mixer process
if (countActiveTracks) {
int allMuted = 1;
uint32_t en = state->enabledTracks;
while (en) {
const int i = 31 - __builtin_clz(en);
en &= ~(1<<i);
track_t& t = state->tracks[i];
if (!t.doesResample() && t.volumeRL == 0)
{
t.needs |= NEEDS_MUTE_ENABLED;
t.hook = track__nop;
} else {
allMuted = 0;
}
}
if (allMuted) {
state->hook = process__nop;
} else if (all16BitsStereoNoResample) {
if (countActiveTracks == 1) {
state->hook = process__OneTrack16BitsStereoNoResampling;
}
}
}
}
static inline
int32_t mulAdd(int16_t in, int16_t v, int32_t a)
{
#if defined(__arm__) && !defined(__thumb__)
int32_t out;
asm( "smlabb %[out], %[in], %[v], %[a] \n"
: [out]"=r"(out)
: [in]"%r"(in), [v]"r"(v), [a]"r"(a)
: );
return out;
#else
return a + in * int32_t(v);
#endif
}
static inline
int32_t mul(int16_t in, int16_t v)
{
#if defined(__arm__) && !defined(__thumb__)
int32_t out;
asm( "smulbb %[out], %[in], %[v] \n"
: [out]"=r"(out)
: [in]"%r"(in), [v]"r"(v)
: );
return out;
#else
return in * int32_t(v);
#endif
}
static inline
int32_t mulAddRL(int left, uint32_t inRL, uint32_t vRL, int32_t a)
{
#if defined(__arm__) && !defined(__thumb__)
int32_t out;
if (left) {
asm( "smlabb %[out], %[inRL], %[vRL], %[a] \n"
: [out]"=r"(out)
: [inRL]"%r"(inRL), [vRL]"r"(vRL), [a]"r"(a)
: );
} else {
asm( "smlatt %[out], %[inRL], %[vRL], %[a] \n"
: [out]"=r"(out)
: [inRL]"%r"(inRL), [vRL]"r"(vRL), [a]"r"(a)
: );
}
return out;
#else
if (left) {
return a + int16_t(inRL&0xFFFF) * int16_t(vRL&0xFFFF);
} else {
return a + int16_t(inRL>>16) * int16_t(vRL>>16);
}
#endif
}
static inline
int32_t mulRL(int left, uint32_t inRL, uint32_t vRL)
{
#if defined(__arm__) && !defined(__thumb__)
int32_t out;
if (left) {
asm( "smulbb %[out], %[inRL], %[vRL] \n"
: [out]"=r"(out)
: [inRL]"%r"(inRL), [vRL]"r"(vRL)
: );
} else {
asm( "smultt %[out], %[inRL], %[vRL] \n"
: [out]"=r"(out)
: [inRL]"%r"(inRL), [vRL]"r"(vRL)
: );
}
return out;
#else
if (left) {
return int16_t(inRL&0xFFFF) * int16_t(vRL&0xFFFF);
} else {
return int16_t(inRL>>16) * int16_t(vRL>>16);
}
#endif
}
void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
{
t->resampler->setSampleRate(t->sampleRate);
// ramp gain - resample to temp buffer and scale/mix in 2nd step
if (aux != NULL) {
// always resample with unity gain when sending to auxiliary buffer to be able
// to apply send level after resampling
// TODO: modify each resampler to support aux channel?
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
volumeRampStereo(t, out, outFrameCount, temp, aux);
} else {
volumeStereo(t, out, outFrameCount, temp, aux);
}
} else {
if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
volumeRampStereo(t, out, outFrameCount, temp, aux);
}
// constant gain
else {
t->resampler->setVolume(t->volume[0], t->volume[1]);
t->resampler->resample(out, outFrameCount, t->bufferProvider);
}
}
}
void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
{
}
void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
{
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
//LOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
// ramp volume
if UNLIKELY(aux != NULL) {
int32_t va = t->prevAuxLevel;
const int32_t vaInc = t->auxInc;
int32_t l;
int32_t r;
do {
l = (*temp++ >> 12);
r = (*temp++ >> 12);
*out++ += (vl >> 16) * l;
*out++ += (vr >> 16) * r;
*aux++ += (va >> 17) * (l + r);
vl += vlInc;
vr += vrInc;
va += vaInc;
} while (--frameCount);
t->prevAuxLevel = va;
} else {
do {
*out++ += (vl >> 16) * (*temp++ >> 12);
*out++ += (vr >> 16) * (*temp++ >> 12);
vl += vlInc;
vr += vrInc;
} while (--frameCount);
}
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->adjustVolumeRamp((aux != NULL));
}
void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
{
const int16_t vl = t->volume[0];
const int16_t vr = t->volume[1];
if UNLIKELY(aux != NULL) {
const int16_t va = (int16_t)t->auxLevel;
do {
int16_t l = (int16_t)(*temp++ >> 12);
int16_t r = (int16_t)(*temp++ >> 12);
out[0] = mulAdd(l, vl, out[0]);
int16_t a = (int16_t)(((int32_t)l + r) >> 1);
out[1] = mulAdd(r, vr, out[1]);
out += 2;
aux[0] = mulAdd(a, va, aux[0]);
aux++;
} while (--frameCount);
} else {
do {
int16_t l = (int16_t)(*temp++ >> 12);
int16_t r = (int16_t)(*temp++ >> 12);
out[0] = mulAdd(l, vl, out[0]);
out[1] = mulAdd(r, vr, out[1]);
out += 2;
} while (--frameCount);
}
}
void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
{
int16_t const *in = static_cast<int16_t const *>(t->in);
if UNLIKELY(aux != NULL) {
int32_t l;
int32_t r;
// ramp gain
if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
int32_t va = t->prevAuxLevel;
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
const int32_t vaInc = t->auxInc;
// LOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
do {
l = (int32_t)*in++;
r = (int32_t)*in++;
*out++ += (vl >> 16) * l;
*out++ += (vr >> 16) * r;
*aux++ += (va >> 17) * (l + r);
vl += vlInc;
vr += vrInc;
va += vaInc;
} while (--frameCount);
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->prevAuxLevel = va;
t->adjustVolumeRamp(true);
}
// constant gain
else {
const uint32_t vrl = t->volumeRL;
const int16_t va = (int16_t)t->auxLevel;
do {
uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
in += 2;
out[0] = mulAddRL(1, rl, vrl, out[0]);
out[1] = mulAddRL(0, rl, vrl, out[1]);
out += 2;
aux[0] = mulAdd(a, va, aux[0]);
aux++;
} while (--frameCount);
}
} else {
// ramp gain
if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
// LOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
do {
*out++ += (vl >> 16) * (int32_t) *in++;
*out++ += (vr >> 16) * (int32_t) *in++;
vl += vlInc;
vr += vrInc;
} while (--frameCount);
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->adjustVolumeRamp(false);
}
// constant gain
else {
const uint32_t vrl = t->volumeRL;
do {
uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
in += 2;
out[0] = mulAddRL(1, rl, vrl, out[0]);
out[1] = mulAddRL(0, rl, vrl, out[1]);
out += 2;
} while (--frameCount);
}
}
t->in = in;
}
void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
{
int16_t const *in = static_cast<int16_t const *>(t->in);
if UNLIKELY(aux != NULL) {
// ramp gain
if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
int32_t va = t->prevAuxLevel;
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
const int32_t vaInc = t->auxInc;
// LOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
do {
int32_t l = *in++;
*out++ += (vl >> 16) * l;
*out++ += (vr >> 16) * l;
*aux++ += (va >> 16) * l;
vl += vlInc;
vr += vrInc;
va += vaInc;
} while (--frameCount);
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->prevAuxLevel = va;
t->adjustVolumeRamp(true);
}
// constant gain
else {
const int16_t vl = t->volume[0];
const int16_t vr = t->volume[1];
const int16_t va = (int16_t)t->auxLevel;
do {
int16_t l = *in++;
out[0] = mulAdd(l, vl, out[0]);
out[1] = mulAdd(l, vr, out[1]);
out += 2;
aux[0] = mulAdd(l, va, aux[0]);
aux++;
} while (--frameCount);
}
} else {
// ramp gain
if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
const int32_t vrInc = t->volumeInc[1];
// LOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
// t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
// (vl + vlInc*frameCount)/65536.0f, frameCount);
do {
int32_t l = *in++;
*out++ += (vl >> 16) * l;
*out++ += (vr >> 16) * l;
vl += vlInc;
vr += vrInc;
} while (--frameCount);
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
t->adjustVolumeRamp(false);
}
// constant gain
else {
const int16_t vl = t->volume[0];
const int16_t vr = t->volume[1];
do {
int16_t l = *in++;
out[0] = mulAdd(l, vl, out[0]);
out[1] = mulAdd(l, vr, out[1]);
out += 2;
} while (--frameCount);
}
}
t->in = in;
}
void AudioMixer::ditherAndClamp(int32_t* out, int32_t const *sums, size_t c)
{
for (size_t i=0 ; i<c ; i++) {
int32_t l = *sums++;
int32_t r = *sums++;
int32_t nl = l >> 12;
int32_t nr = r >> 12;
l = clamp16(nl);
r = clamp16(nr);
*out++ = (r<<16) | (l & 0xFFFF);
}
}
// no-op case
void AudioMixer::process__nop(state_t* state)
{
uint32_t e0 = state->enabledTracks;
size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
while (e0) {
// process by group of tracks with same output buffer to
// avoid multiple memset() on same buffer
uint32_t e1 = e0, e2 = e0;
int i = 31 - __builtin_clz(e1);
track_t& t1 = state->tracks[i];
e2 &= ~(1<<i);
while (e2) {
i = 31 - __builtin_clz(e2);
e2 &= ~(1<<i);
track_t& t2 = state->tracks[i];
if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
e1 &= ~(1<<i);
}
}
e0 &= ~(e1);
memset(t1.mainBuffer, 0, bufSize);
while (e1) {
i = 31 - __builtin_clz(e1);
e1 &= ~(1<<i);
t1 = state->tracks[i];
size_t outFrames = state->frameCount;
while (outFrames) {
t1.buffer.frameCount = outFrames;
t1.bufferProvider->getNextBuffer(&t1.buffer);
if (!t1.buffer.raw) break;
outFrames -= t1.buffer.frameCount;
t1.bufferProvider->releaseBuffer(&t1.buffer);
}
}
}
}
// generic code without resampling
void AudioMixer::process__genericNoResampling(state_t* state)
{
int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
// acquire each track's buffer
uint32_t enabledTracks = state->enabledTracks;
uint32_t e0 = enabledTracks;
while (e0) {
const int i = 31 - __builtin_clz(e0);
e0 &= ~(1<<i);
track_t& t = state->tracks[i];
t.buffer.frameCount = state->frameCount;
t.bufferProvider->getNextBuffer(&t.buffer);
t.frameCount = t.buffer.frameCount;
t.in = t.buffer.raw;
// t.in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
if (t.in == NULL)
enabledTracks &= ~(1<<i);
}
e0 = enabledTracks;
while (e0) {
// process by group of tracks with same output buffer to
// optimize cache use
uint32_t e1 = e0, e2 = e0;
int j = 31 - __builtin_clz(e1);
track_t& t1 = state->tracks[j];
e2 &= ~(1<<j);
while (e2) {
j = 31 - __builtin_clz(e2);
e2 &= ~(1<<j);
track_t& t2 = state->tracks[j];
if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
e1 &= ~(1<<j);
}
}
e0 &= ~(e1);
// this assumes output 16 bits stereo, no resampling
int32_t *out = t1.mainBuffer;
size_t numFrames = 0;
do {
memset(outTemp, 0, sizeof(outTemp));
e2 = e1;
while (e2) {
const int i = 31 - __builtin_clz(e2);
e2 &= ~(1<<i);
track_t& t = state->tracks[i];
size_t outFrames = BLOCKSIZE;
int32_t *aux = NULL;
if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
aux = t.auxBuffer + numFrames;
}
while (outFrames) {
size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
if (inFrames) {
(t.hook)(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux);
t.frameCount -= inFrames;
outFrames -= inFrames;
if UNLIKELY(aux != NULL) {
aux += inFrames;
}
}
if (t.frameCount == 0 && outFrames) {
t.bufferProvider->releaseBuffer(&t.buffer);
t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames);
t.bufferProvider->getNextBuffer(&t.buffer);
t.in = t.buffer.raw;
if (t.in == NULL) {
enabledTracks &= ~(1<<i);
e1 &= ~(1<<i);
break;
}
t.frameCount = t.buffer.frameCount;
}
}
}
ditherAndClamp(out, outTemp, BLOCKSIZE);
out += BLOCKSIZE;
numFrames += BLOCKSIZE;
} while (numFrames < state->frameCount);
}
// release each track's buffer
e0 = enabledTracks;
while (e0) {
const int i = 31 - __builtin_clz(e0);
e0 &= ~(1<<i);
track_t& t = state->tracks[i];
t.bufferProvider->releaseBuffer(&t.buffer);
}
}
// generic code with resampling
void AudioMixer::process__genericResampling(state_t* state)
{
int32_t* const outTemp = state->outputTemp;
const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
memset(outTemp, 0, size);
size_t numFrames = state->frameCount;
uint32_t e0 = state->enabledTracks;
while (e0) {
// process by group of tracks with same output buffer
// to optimize cache use
uint32_t e1 = e0, e2 = e0;
int j = 31 - __builtin_clz(e1);
track_t& t1 = state->tracks[j];
e2 &= ~(1<<j);
while (e2) {
j = 31 - __builtin_clz(e2);
e2 &= ~(1<<j);
track_t& t2 = state->tracks[j];
if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
e1 &= ~(1<<j);
}
}
e0 &= ~(e1);
int32_t *out = t1.mainBuffer;
while (e1) {
const int i = 31 - __builtin_clz(e1);
e1 &= ~(1<<i);
track_t& t = state->tracks[i];
int32_t *aux = NULL;
if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
aux = t.auxBuffer;
}
// this is a little goofy, on the resampling case we don't
// acquire/release the buffers because it's done by
// the resampler.
if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
(t.hook)(&t, outTemp, numFrames, state->resampleTemp, aux);
} else {
size_t outFrames = 0;
while (outFrames < numFrames) {
t.buffer.frameCount = numFrames - outFrames;
t.bufferProvider->getNextBuffer(&t.buffer);
t.in = t.buffer.raw;
// t.in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
if (t.in == NULL) break;
if UNLIKELY(aux != NULL) {
aux += outFrames;
}
(t.hook)(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux);
outFrames += t.buffer.frameCount;
t.bufferProvider->releaseBuffer(&t.buffer);
}
}
}
ditherAndClamp(out, outTemp, numFrames);
}
}
// one track, 16 bits stereo without resampling is the most common case
void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
{
const int i = 31 - __builtin_clz(state->enabledTracks);
const track_t& t = state->tracks[i];
AudioBufferProvider::Buffer& b(t.buffer);
int32_t* out = t.mainBuffer;
size_t numFrames = state->frameCount;
const int16_t vl = t.volume[0];
const int16_t vr = t.volume[1];
const uint32_t vrl = t.volumeRL;
while (numFrames) {
b.frameCount = numFrames;
t.bufferProvider->getNextBuffer(&b);
int16_t const *in = b.i16;
// in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
if (in == NULL || ((unsigned long)in & 3)) {
memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t));
LOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: buffer %p track %d, channels %d, needs %08x",
in, i, t.channelCount, t.needs);
return;
}
size_t outFrames = b.frameCount;
if (UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
// volume is boosted, so we might need to clamp even though
// we process only one track.
do {
uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
in += 2;
int32_t l = mulRL(1, rl, vrl) >> 12;
int32_t r = mulRL(0, rl, vrl) >> 12;
// clamping...
l = clamp16(l);
r = clamp16(r);
*out++ = (r<<16) | (l & 0xFFFF);
} while (--outFrames);
} else {
do {
uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
in += 2;
int32_t l = mulRL(1, rl, vrl) >> 12;
int32_t r = mulRL(0, rl, vrl) >> 12;
*out++ = (r<<16) | (l & 0xFFFF);
} while (--outFrames);
}
numFrames -= b.frameCount;
t.bufferProvider->releaseBuffer(&b);
}
}
// 2 tracks is also a common case
// NEVER used in current implementation of process__validate()
// only use if the 2 tracks have the same output buffer
void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
{
int i;
uint32_t en = state->enabledTracks;
i = 31 - __builtin_clz(en);
const track_t& t0 = state->tracks[i];
AudioBufferProvider::Buffer& b0(t0.buffer);
en &= ~(1<<i);
i = 31 - __builtin_clz(en);
const track_t& t1 = state->tracks[i];
AudioBufferProvider::Buffer& b1(t1.buffer);
int16_t const *in0;
const int16_t vl0 = t0.volume[0];
const int16_t vr0 = t0.volume[1];
size_t frameCount0 = 0;
int16_t const *in1;
const int16_t vl1 = t1.volume[0];
const int16_t vr1 = t1.volume[1];
size_t frameCount1 = 0;
//FIXME: only works if two tracks use same buffer
int32_t* out = t0.mainBuffer;
size_t numFrames = state->frameCount;
int16_t const *buff = NULL;
while (numFrames) {
if (frameCount0 == 0) {
b0.frameCount = numFrames;
t0.bufferProvider->getNextBuffer(&b0);
if (b0.i16 == NULL) {
if (buff == NULL) {
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
}
in0 = buff;
b0.frameCount = numFrames;
} else {
in0 = b0.i16;
}
frameCount0 = b0.frameCount;
}
if (frameCount1 == 0) {
b1.frameCount = numFrames;
t1.bufferProvider->getNextBuffer(&b1);
if (b1.i16 == NULL) {
if (buff == NULL) {
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
}
in1 = buff;
b1.frameCount = numFrames;
} else {
in1 = b1.i16;
}
frameCount1 = b1.frameCount;
}
size_t outFrames = frameCount0 < frameCount1?frameCount0:frameCount1;
numFrames -= outFrames;
frameCount0 -= outFrames;
frameCount1 -= outFrames;
do {
int32_t l0 = *in0++;
int32_t r0 = *in0++;
l0 = mul(l0, vl0);
r0 = mul(r0, vr0);
int32_t l = *in1++;
int32_t r = *in1++;
l = mulAdd(l, vl1, l0) >> 12;
r = mulAdd(r, vr1, r0) >> 12;
// clamping...
l = clamp16(l);
r = clamp16(r);
*out++ = (r<<16) | (l & 0xFFFF);
} while (--outFrames);
if (frameCount0 == 0) {
t0.bufferProvider->releaseBuffer(&b0);
}
if (frameCount1 == 0) {
t1.bufferProvider->releaseBuffer(&b1);
}
}
if (buff != NULL) {
delete [] buff;
}
}
// ----------------------------------------------------------------------------
}; // namespace android