| /* //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 |
