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-rw-r--r--libs/audioflinger/AudioMixer.cpp915
1 files changed, 0 insertions, 915 deletions
diff --git a/libs/audioflinger/AudioMixer.cpp b/libs/audioflinger/AudioMixer.cpp
deleted file mode 100644
index 19a442a864ae..000000000000
--- a/libs/audioflinger/AudioMixer.cpp
+++ /dev/null
@@ -1,915 +0,0 @@
-/* //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->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++;
- }
-}
-
- 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, int value)
-{
- switch (target) {
- case TRACK:
- if (name == CHANNEL_COUNT) {
- if ((uint32_t(value) <= MAX_NUM_CHANNELS) && (value)) {
- if (mState.tracks[ mActiveTrack ].channelCount != value) {
- mState.tracks[ mActiveTrack ].channelCount = value;
- LOGV("setParameter(TRACK, CHANNEL_COUNT, %d)", value);
- invalidateState(1<<mActiveTrack);
- }
- return NO_ERROR;
- }
- }
- break;
- case RESAMPLE:
- if (name == SAMPLE_RATE) {
- if (value > 0) {
- track_t& track = mState.tracks[ mActiveTrack ];
- if (track.setResampler(uint32_t(value), mSampleRate)) {
- LOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
- uint32_t(value));
- 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] != value) {
- track.prevVolume[name-VOLUME0] = track.volume[name-VOLUME0] << 16;
- track.volume[name-VOLUME0] = value;
- if (target == VOLUME) {
- track.prevVolume[name-VOLUME0] = value << 16;
- track.volumeInc[name-VOLUME0] = 0;
- } else {
- int32_t d = (value<<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] = value << 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()
-{
- 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;
- }
- }
-}
-
-
-status_t AudioMixer::setBufferProvider(AudioBufferProvider* buffer)
-{
- mState.tracks[ mActiveTrack ].bufferProvider = buffer;
- return NO_ERROR;
-}
-
-
-
-void AudioMixer::process(void* output)
-{
- mState.hook(&mState, output);
-}
-
-
-void AudioMixer::process__validate(state_t* state, void* output)
-{
- 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.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_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, output);
-
- // 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 (!resampling && 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)
-{
- t->resampler->setSampleRate(t->sampleRate);
-
- // ramp gain - resample to temp buffer and scale/mix in 2nd step
- 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);
- }
-
- // 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)
-{
-}
-
-void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp)
-{
- 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
- 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();
-}
-
-void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp)
-{
- int16_t const *in = static_cast<int16_t const *>(t->in);
-
- // 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();
- }
-
- // 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)
-{
- int16_t const *in = static_cast<int16_t const *>(t->in);
-
- // 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();
- }
- // 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, void* output)
-{
- // this assumes output 16 bits stereo, no resampling
- memset(output, 0, state->frameCount*4);
- uint32_t en = state->enabledTracks;
- while (en) {
- const int i = 31 - __builtin_clz(en);
- en &= ~(1<<i);
- track_t& t = state->tracks[i];
- size_t outFrames = state->frameCount;
- while (outFrames) {
- t.buffer.frameCount = outFrames;
- t.bufferProvider->getNextBuffer(&t.buffer);
- if (!t.buffer.raw) break;
- outFrames -= t.buffer.frameCount;
- t.bufferProvider->releaseBuffer(&t.buffer);
- }
- }
-}
-
-// generic code without resampling
-void AudioMixer::process__genericNoResampling(state_t* state, void* output)
-{
- int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
-
- // acquire each track's buffer
- uint32_t enabledTracks = state->enabledTracks;
- uint32_t en = enabledTracks;
- while (en) {
- const int i = 31 - __builtin_clz(en);
- en &= ~(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);
- }
-
- // this assumes output 16 bits stereo, no resampling
- int32_t* out = static_cast<int32_t*>(output);
- size_t numFrames = state->frameCount;
- do {
- memset(outTemp, 0, sizeof(outTemp));
-
- en = enabledTracks;
- while (en) {
- const int i = 31 - __builtin_clz(en);
- en &= ~(1<<i);
- track_t& t = state->tracks[i];
- size_t outFrames = BLOCKSIZE;
-
- 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);
- t.frameCount -= inFrames;
- outFrames -= inFrames;
- }
- if (t.frameCount == 0 && outFrames) {
- t.bufferProvider->releaseBuffer(&t.buffer);
- t.buffer.frameCount = numFrames - (BLOCKSIZE - outFrames);
- t.bufferProvider->getNextBuffer(&t.buffer);
- t.in = t.buffer.raw;
- if (t.in == NULL) {
- enabledTracks &= ~(1<<i);
- break;
- }
- t.frameCount = t.buffer.frameCount;
- }
- }
- }
-
- ditherAndClamp(out, outTemp, BLOCKSIZE);
- out += BLOCKSIZE;
- numFrames -= BLOCKSIZE;
- } while (numFrames);
-
-
- // release each track's buffer
- en = enabledTracks;
- while (en) {
- const int i = 31 - __builtin_clz(en);
- en &= ~(1<<i);
- track_t& t = state->tracks[i];
- t.bufferProvider->releaseBuffer(&t.buffer);
- }
-}
-
-// generic code with resampling
-void AudioMixer::process__genericResampling(state_t* state, void* output)
-{
- int32_t* const outTemp = state->outputTemp;
- const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
- memset(outTemp, 0, size);
-
- int32_t* out = static_cast<int32_t*>(output);
- size_t numFrames = state->frameCount;
-
- uint32_t en = state->enabledTracks;
- while (en) {
- const int i = 31 - __builtin_clz(en);
- en &= ~(1<<i);
- track_t& t = state->tracks[i];
-
- // 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);
- } else {
-
- size_t outFrames = numFrames;
-
- while (outFrames) {
- t.buffer.frameCount = 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;
-
- (t.hook)(&t, outTemp + (numFrames-outFrames)*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp);
- 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, void* output)
-{
- const int i = 31 - __builtin_clz(state->enabledTracks);
- const track_t& t = state->tracks[i];
-
- AudioBufferProvider::Buffer& b(t.buffer);
-
- int32_t* out = static_cast<int32_t*>(output);
- 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
-void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, void* output)
-{
- 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;
-
- int32_t* out = static_cast<int32_t*>(output);
- 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
-
generated by cgit v1.2.3 (git 2.25.1) at 2025-10-31 15:13:22 +0000