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dc3ac85009fbd549faec6a46295a9d34a19e01ac
frameworks_base/services/audioflinger/AudioMixer.cpp
Glenn Kasten dc3ac85009 Constructor initialization and const fields 
In constructors, initialize member fields in the initialization list
rather than constructor body where possible.  This allows more fields
to be const, provided they are never modified.

Also initialize POD fields in constructor, unless it's obvious they
don't need to be initialized.  In that case, put a comment instead.

Remove explicit clear() in destructors on fields that are now const.

Give AudioSessionRef a default constructor, so it's immutable fields can
be marked const.

Add comment about ~TrackBase() trick.

Initialize fields in declaration order to make it easier to confirm that
all fields are set.

Move initialization of mHardwareStatus from onFirstRef() to constructor.

Use NULL not 0 to initialize raw pointers in initialization list.

Rename field mClient to mAudioFlingerClient, and getter from client()
to audioFlingerClient().

Change-Id: Ib36cf6ed32f3cd19003f40a5d84046eb4c122052
2012-02-03 07:37:14 -08:00

1138 lines
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/* //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 <assert.h>
#include <stdint.h>
#include <string.h>
#include <stdlib.h>
#include <sys/types.h>
#include <utils/Errors.h>
#include <utils/Log.h>
#include <cutils/bitops.h>
#include <cutils/compiler.h>
#include <system/audio.h>
#include <audio_utils/primitives.h>
#include "AudioMixer.h"
namespace android {
// ----------------------------------------------------------------------------
AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
: mTrackNames(0), mSampleRate(sampleRate)
{
// AudioMixer is not yet capable of multi-channel beyond stereo
assert(2 == MAX_NUM_CHANNELS);
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
mState.hook = process__nop;
mState.outputTemp = NULL;
mState.resampleTemp = NULL;
// mState.reserved
track_t* t = mState.tracks;
for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
t->needs = 0;
t->volume[0] = UNITY_GAIN;
t->volume[1] = UNITY_GAIN;
// no initialization needed
// t->prevVolume[0]
// t->prevVolume[1]
t->volumeInc[0] = 0;
t->volumeInc[1] = 0;
t->auxLevel = 0;
t->auxInc = 0;
// no initialization needed
// t->prevAuxLevel
// t->frameCount
t->channelCount = 2;
t->enabled = 0;
t->format = 16;
t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
t->bufferProvider = NULL;
t->buffer.raw = NULL;
// t->buffer.frameCount
t->hook = NULL;
t->in = NULL;
t->resampler = NULL;
t->sampleRate = mSampleRate;
t->mainBuffer = NULL;
t->auxBuffer = NULL;
t++;
}
}
AudioMixer::~AudioMixer()
{
track_t* t = mState.tracks;
for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
delete t->resampler;
t++;
}
delete [] mState.outputTemp;
delete [] mState.resampleTemp;
}
int AudioMixer::getTrackName()
{
uint32_t names = ~mTrackNames;
if (names != 0) {
int n = __builtin_ctz(names);
ALOGV("add track (%d)", n);
mTrackNames |= 1 << n;
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;
assert(uint32_t(name) < MAX_NUM_TRACKS);
ALOGV("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 = NULL;
track.sampleRate = mSampleRate;
invalidateState(1<<name);
}
track.volumeInc[0] = 0;
track.volumeInc[1] = 0;
mTrackNames &= ~(1<<name);
}
void AudioMixer::enable(int name)
{
name -= TRACK0;
assert(uint32_t(name) < MAX_NUM_TRACKS);
track_t& track = mState.tracks[name];
if (track.enabled != 1) {
track.enabled = 1;
ALOGV("enable(%d)", name);
invalidateState(1 << name);
}
}
void AudioMixer::disable(int name)
{
name -= TRACK0;
assert(uint32_t(name) < MAX_NUM_TRACKS);
track_t& track = mState.tracks[name];
if (track.enabled != 0) {
track.enabled = 0;
ALOGV("disable(%d)", name);
invalidateState(1 << name);
}
}
void AudioMixer::setParameter(int name, int target, int param, void *value)
{
name -= TRACK0;
assert(uint32_t(name) < MAX_NUM_TRACKS);
track_t& track = mState.tracks[name];
int valueInt = (int)value;
int32_t *valueBuf = (int32_t *)value;
switch (target) {
case TRACK:
switch (param) {
case CHANNEL_MASK: {
uint32_t mask = (uint32_t)value;
if (track.channelMask != mask) {
uint8_t channelCount = popcount(mask);
assert((channelCount <= MAX_NUM_CHANNELS) && (channelCount));
track.channelMask = mask;
track.channelCount = channelCount;
ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
invalidateState(1 << name);
}
} break;
case MAIN_BUFFER:
if (track.mainBuffer != valueBuf) {
track.mainBuffer = valueBuf;
ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
invalidateState(1 << name);
}
break;
case AUX_BUFFER:
if (track.auxBuffer != valueBuf) {
track.auxBuffer = valueBuf;
ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
invalidateState(1 << name);
}
break;
default:
// bad param
assert(false);
}
break;
case RESAMPLE:
switch (param) {
case SAMPLE_RATE:
assert(valueInt > 0);
if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
uint32_t(valueInt));
invalidateState(1 << name);
}
break;
case RESET:
track.resetResampler();
invalidateState(1 << name);
break;
default:
// bad param
assert(false);
}
break;
case RAMP_VOLUME:
case VOLUME:
switch (param) {
case VOLUME0:
case VOLUME1:
if (track.volume[param-VOLUME0] != valueInt) {
ALOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt);
track.prevVolume[param-VOLUME0] = track.volume[param-VOLUME0] << 16;
track.volume[param-VOLUME0] = valueInt;
if (target == VOLUME) {
track.prevVolume[param-VOLUME0] = valueInt << 16;
track.volumeInc[param-VOLUME0] = 0;
} else {
int32_t d = (valueInt<<16) - track.prevVolume[param-VOLUME0];
int32_t volInc = d / int32_t(mState.frameCount);
track.volumeInc[param-VOLUME0] = volInc;
if (volInc == 0) {
track.prevVolume[param-VOLUME0] = valueInt << 16;
}
}
invalidateState(1 << name);
}
break;
case AUXLEVEL:
if (track.auxLevel != valueInt) {
ALOGV("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 << name);
}
break;
default:
// bad param
assert(false);
}
break;
default:
// bad target
assert(false);
}
}
bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
{
if (value!=devSampleRate || resampler) {
if (sampleRate != value) {
sampleRate = value;
if (resampler == NULL) {
resampler = AudioResampler::create(
format, channelCount, devSampleRate);
}
return true;
}
}
return false;
}
bool AudioMixer::track_t::doesResample() const
{
return resampler != NULL;
}
void AudioMixer::track_t::resetResampler()
{
if (resampler != NULL) {
resampler->reset();
}
}
inline
void AudioMixer::track_t::adjustVolumeRamp(bool aux)
{
for (uint32_t i=0 ; i<MAX_NUM_CHANNELS ; 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;
}
}
}
size_t AudioMixer::track_t::getUnreleasedFrames()
{
if (resampler != NULL) {
return resampler->getUnreleasedFrames();
}
return 0;
}
size_t AudioMixer::getUnreleasedFrames(int name)
{
name -= TRACK0;
if (uint32_t(name) < MAX_NUM_TRACKS) {
track_t& track(mState.tracks[name]);
return track.getUnreleasedFrames();
}
return 0;
}
void AudioMixer::setBufferProvider(int name, AudioBufferProvider* buffer)
{
name -= TRACK0;
assert(uint32_t(name) < MAX_NUM_TRACKS);
mState.tracks[name].bufferProvider = buffer;
}
void AudioMixer::process()
{
mState.hook(&mState);
}
void AudioMixer::process__validate(state_t* state)
{
ALOGW_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 = NULL;
}
if (state->resampleTemp) {
delete [] state->resampleTemp;
state->resampleTemp = NULL;
}
state->hook = process__genericNoResampling;
if (all16BitsStereoNoResample && !volumeRamp) {
if (countActiveTracks == 1) {
state->hook = process__OneTrack16BitsStereoNoResampling;
}
}
}
}
ALOGV("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;
}
}
}
}
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 (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
volumeRampStereo(t, out, outFrameCount, temp, aux);
} else {
volumeStereo(t, out, outFrameCount, temp, aux);
}
} else {
if (CC_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];
//ALOGD("[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 (CC_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 (CC_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)
{
const int16_t *in = static_cast<const int16_t *>(t->in);
if (CC_UNLIKELY(aux != NULL)) {
int32_t l;
int32_t r;
// ramp gain
if (CC_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;
// ALOGD("[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<const uint32_t *>(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 (CC_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];
// ALOGD("[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<const uint32_t *>(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)
{
const int16_t *in = static_cast<int16_t const *>(t->in);
if (CC_UNLIKELY(aux != NULL)) {
// ramp gain
if (CC_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;
// ALOGD("[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 (CC_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];
// ALOGD("[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;
}
// 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 (CC_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 (CC_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 (CC_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 (CC_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)
{
// this const just means that local variable outTemp doesn't change
int32_t* const outTemp = state->outputTemp;
const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
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 (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<j);
}
}
e0 &= ~(e1);
int32_t *out = t1.mainBuffer;
memset(outTemp, 0, size);
while (e1) {
const int i = 31 - __builtin_clz(e1);
e1 &= ~(1<<i);
track_t& t = state->tracks[i];
int32_t *aux = NULL;
if (CC_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 (CC_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);
const int16_t *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));
ALOGE_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 (CC_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<const uint32_t *>(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<const uint32_t *>(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);
}
}
#if 0
// 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);
const int16_t *in0;
const int16_t vl0 = t0.volume[0];
const int16_t vr0 = t0.volume[1];
size_t frameCount0 = 0;
const int16_t *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;
const int16_t *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);
}
}
delete [] buff;
}
#endif
// ----------------------------------------------------------------------------
}; // namespace android
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