diff options
Diffstat (limited to 'libs/audioflinger/AudioMixer.cpp')
-rw-r--r-- | libs/audioflinger/AudioMixer.cpp | 913 |
1 files changed, 0 insertions, 913 deletions
diff --git a/libs/audioflinger/AudioMixer.cpp b/libs/audioflinger/AudioMixer.cpp deleted file mode 100644 index b03467f..0000000 --- a/libs/audioflinger/AudioMixer.cpp +++ /dev/null @@ -1,913 +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" - -#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; -} - -inline -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) { - memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t)); - 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 - |