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-rw-r--r--libs/audioflinger/AudioMixer.cpp694
1 files changed, 487 insertions, 207 deletions
diff --git a/libs/audioflinger/AudioMixer.cpp b/libs/audioflinger/AudioMixer.cpp
index 19a442a..8aaa325 100644
--- a/libs/audioflinger/AudioMixer.cpp
+++ b/libs/audioflinger/AudioMixer.cpp
@@ -56,6 +56,8 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
t->volume[1] = UNITY_GAIN;
t->volumeInc[0] = 0;
t->volumeInc[1] = 0;
+ t->auxLevel = 0;
+ t->auxInc = 0;
t->channelCount = 2;
t->enabled = 0;
t->format = 16;
@@ -65,6 +67,8 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
t->resampler = 0;
t->sampleRate = mSampleRate;
t->in = 0;
+ t->mainBuffer = NULL;
+ t->auxBuffer = NULL;
t++;
}
}
@@ -169,28 +173,48 @@ status_t AudioMixer::setActiveTrack(int track)
return NO_ERROR;
}
-status_t AudioMixer::setParameter(int target, int name, int value)
+status_t AudioMixer::setParameter(int target, int name, void *value)
{
+ int valueInt = (int)value;
+ int32_t *valueBuf = (int32_t *)value;
+
switch (target) {
case TRACK:
if (name == CHANNEL_COUNT) {
- if ((uint32_t(value) <= MAX_NUM_CHANNELS) && (value)) {
- if (mState.tracks[ mActiveTrack ].channelCount != value) {
- mState.tracks[ mActiveTrack ].channelCount = value;
- LOGV("setParameter(TRACK, CHANNEL_COUNT, %d)", value);
+ if ((uint32_t(valueInt) <= MAX_NUM_CHANNELS) && (valueInt)) {
+ if (mState.tracks[ mActiveTrack ].channelCount != valueInt) {
+ mState.tracks[ mActiveTrack ].channelCount = valueInt;
+ LOGV("setParameter(TRACK, CHANNEL_COUNT, %d)", valueInt);
invalidateState(1<<mActiveTrack);
}
return NO_ERROR;
}
}
+ if (name == MAIN_BUFFER) {
+ if (mState.tracks[ mActiveTrack ].mainBuffer != valueBuf) {
+ mState.tracks[ mActiveTrack ].mainBuffer = valueBuf;
+ LOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
+ invalidateState(1<<mActiveTrack);
+ }
+ return NO_ERROR;
+ }
+ if (name == AUX_BUFFER) {
+ if (mState.tracks[ mActiveTrack ].auxBuffer != valueBuf) {
+ mState.tracks[ mActiveTrack ].auxBuffer = valueBuf;
+ LOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
+ invalidateState(1<<mActiveTrack);
+ }
+ return NO_ERROR;
+ }
+
break;
case RESAMPLE:
if (name == SAMPLE_RATE) {
- if (value > 0) {
+ if (valueInt > 0) {
track_t& track = mState.tracks[ mActiveTrack ];
- if (track.setResampler(uint32_t(value), mSampleRate)) {
+ if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
LOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
- uint32_t(value));
+ uint32_t(valueInt));
invalidateState(1<<mActiveTrack);
}
return NO_ERROR;
@@ -201,18 +225,39 @@ status_t AudioMixer::setParameter(int target, int name, int value)
case VOLUME:
if ((uint32_t(name-VOLUME0) < MAX_NUM_CHANNELS)) {
track_t& track = mState.tracks[ mActiveTrack ];
- if (track.volume[name-VOLUME0] != value) {
+ if (track.volume[name-VOLUME0] != valueInt) {
+ LOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt);
track.prevVolume[name-VOLUME0] = track.volume[name-VOLUME0] << 16;
- track.volume[name-VOLUME0] = value;
+ track.volume[name-VOLUME0] = valueInt;
if (target == VOLUME) {
- track.prevVolume[name-VOLUME0] = value << 16;
+ track.prevVolume[name-VOLUME0] = valueInt << 16;
track.volumeInc[name-VOLUME0] = 0;
} else {
- int32_t d = (value<<16) - track.prevVolume[name-VOLUME0];
+ int32_t d = (valueInt<<16) - track.prevVolume[name-VOLUME0];
int32_t volInc = d / int32_t(mState.frameCount);
track.volumeInc[name-VOLUME0] = volInc;
if (volInc == 0) {
- track.prevVolume[name-VOLUME0] = value << 16;
+ track.prevVolume[name-VOLUME0] = valueInt << 16;
+ }
+ }
+ invalidateState(1<<mActiveTrack);
+ }
+ return NO_ERROR;
+ } else if (name == AUXLEVEL) {
+ track_t& track = mState.tracks[ mActiveTrack ];
+ if (track.auxLevel != valueInt) {
+ LOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt);
+ track.prevAuxLevel = track.auxLevel << 16;
+ track.auxLevel = valueInt;
+ if (target == VOLUME) {
+ track.prevAuxLevel = valueInt << 16;
+ track.auxInc = 0;
+ } else {
+ int32_t d = (valueInt<<16) - track.prevAuxLevel;
+ int32_t volInc = d / int32_t(mState.frameCount);
+ track.auxInc = volInc;
+ if (volInc == 0) {
+ track.prevAuxLevel = valueInt << 16;
}
}
invalidateState(1<<mActiveTrack);
@@ -245,7 +290,7 @@ bool AudioMixer::track_t::doesResample() const
}
inline
-void AudioMixer::track_t::adjustVolumeRamp()
+void AudioMixer::track_t::adjustVolumeRamp(bool aux)
{
for (int i=0 ; i<2 ; i++) {
if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
@@ -254,6 +299,13 @@ void AudioMixer::track_t::adjustVolumeRamp()
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;
+ }
+ }
}
@@ -265,13 +317,13 @@ status_t AudioMixer::setBufferProvider(AudioBufferProvider* buffer)
-void AudioMixer::process(void* output)
+void AudioMixer::process()
{
- mState.hook(&mState, output);
+ mState.hook(&mState);
}
-void AudioMixer::process__validate(state_t* state, void* output)
+void AudioMixer::process__validate(state_t* state)
{
LOGW_IF(!state->needsChanged,
"in process__validate() but nothing's invalid");
@@ -308,7 +360,10 @@ void AudioMixer::process__validate(state_t* state, void* output)
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) {
@@ -319,6 +374,9 @@ void AudioMixer::process__validate(state_t* state, void* output)
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;
@@ -369,7 +427,7 @@ void AudioMixer::process__validate(state_t* state, void* output)
countActiveTracks, state->enabledTracks,
all16BitsStereoNoResample, resampling, volumeRamp);
- state->hook(state, output);
+ state->hook(state);
// Now that the volume ramp has been done, set optimal state and
// track hooks for subsequent mixer process
@@ -390,7 +448,7 @@ void AudioMixer::process__validate(state_t* state, void* output)
}
if (allMuted) {
state->hook = process__nop;
- } else if (!resampling && all16BitsStereoNoResample) {
+ } else if (all16BitsStereoNoResample) {
if (countActiveTracks == 1) {
state->hook = process__OneTrack16BitsStereoNoResampling;
}
@@ -481,30 +539,44 @@ int32_t mulRL(int left, uint32_t inRL, uint32_t vRL)
}
-void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp)
+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 UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+ 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);
- volumeRampStereo(t, out, outFrameCount, temp);
- }
+ if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
+ volumeRampStereo(t, out, outFrameCount, temp, aux);
+ } else {
+ volumeStereo(t, out, outFrameCount, temp, aux);
+ }
+ } else {
+ if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+ t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
+ memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
+ t->resampler->resample(temp, outFrameCount, t->bufferProvider);
+ volumeRampStereo(t, out, outFrameCount, temp, aux);
+ }
- // constant gain
- else {
- t->resampler->setVolume(t->volume[0], t->volume[1]);
- t->resampler->resample(out, outFrameCount, t->bufferProvider);
+ // 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::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)
+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];
@@ -514,98 +586,238 @@ void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, i
//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);
+ if UNLIKELY(aux != NULL) {
+ int32_t va = t->prevAuxLevel;
+ const int32_t vaInc = t->auxInc;
+ int32_t l;
+ int32_t r;
+ do {
+ l = (*temp++ >> 12);
+ r = (*temp++ >> 12);
+ *out++ += (vl >> 16) * l;
+ *out++ += (vr >> 16) * r;
+ *aux++ += (va >> 17) * (l + r);
+ vl += vlInc;
+ vr += vrInc;
+ va += vaInc;
+ } while (--frameCount);
+ t->prevAuxLevel = va;
+ } else {
+ do {
+ *out++ += (vl >> 16) * (*temp++ >> 12);
+ *out++ += (vr >> 16) * (*temp++ >> 12);
+ vl += vlInc;
+ vr += vrInc;
+ } while (--frameCount);
+ }
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
- t->adjustVolumeRamp();
+ t->adjustVolumeRamp((aux != NULL));
}
-void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp)
+void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
{
- int16_t const *in = static_cast<int16_t const *>(t->in);
-
- // 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);
+ const int16_t vl = t->volume[0];
+ const int16_t vr = t->volume[1];
+ if UNLIKELY(aux != NULL) {
+ const int16_t va = (int16_t)t->auxLevel;
do {
- *out++ += (vl >> 16) * (int32_t) *in++;
- *out++ += (vr >> 16) * (int32_t) *in++;
- vl += vlInc;
- vr += vrInc;
+ 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);
-
- t->prevVolume[0] = vl;
- t->prevVolume[1] = vr;
- t->adjustVolumeRamp();
- }
-
- // constant gain
- else {
- const uint32_t vrl = t->volumeRL;
+ } else {
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]);
+ int16_t l = (int16_t)(*temp++ >> 12);
+ int16_t r = (int16_t)(*temp++ >> 12);
+ out[0] = mulAdd(l, vl, out[0]);
+ out[1] = mulAdd(r, vr, out[1]);
out += 2;
} while (--frameCount);
}
+}
+
+void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+{
+ int16_t const *in = static_cast<int16_t const *>(t->in);
+
+ if UNLIKELY(aux != NULL) {
+ int32_t l;
+ int32_t r;
+ // ramp gain
+ if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
+ int32_t vl = t->prevVolume[0];
+ int32_t vr = t->prevVolume[1];
+ int32_t va = t->prevAuxLevel;
+ const int32_t vlInc = t->volumeInc[0];
+ const int32_t vrInc = t->volumeInc[1];
+ const int32_t vaInc = t->auxInc;
+ // LOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+ // t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
+ // (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+ do {
+ l = (int32_t)*in++;
+ r = (int32_t)*in++;
+ *out++ += (vl >> 16) * l;
+ *out++ += (vr >> 16) * r;
+ *aux++ += (va >> 17) * (l + r);
+ vl += vlInc;
+ vr += vrInc;
+ va += vaInc;
+ } while (--frameCount);
+
+ t->prevVolume[0] = vl;
+ t->prevVolume[1] = vr;
+ t->prevAuxLevel = va;
+ t->adjustVolumeRamp(true);
+ }
+
+ // constant gain
+ else {
+ const uint32_t vrl = t->volumeRL;
+ const int16_t va = (int16_t)t->auxLevel;
+ do {
+ uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+ int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
+ in += 2;
+ out[0] = mulAddRL(1, rl, vrl, out[0]);
+ out[1] = mulAddRL(0, rl, vrl, out[1]);
+ out += 2;
+ aux[0] = mulAdd(a, va, aux[0]);
+ aux++;
+ } while (--frameCount);
+ }
+ } else {
+ // ramp gain
+ if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+ int32_t vl = t->prevVolume[0];
+ int32_t vr = t->prevVolume[1];
+ const int32_t vlInc = t->volumeInc[0];
+ const int32_t vrInc = t->volumeInc[1];
+
+ // LOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+ // t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
+ // (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+ do {
+ *out++ += (vl >> 16) * (int32_t) *in++;
+ *out++ += (vr >> 16) * (int32_t) *in++;
+ vl += vlInc;
+ vr += vrInc;
+ } while (--frameCount);
+
+ t->prevVolume[0] = vl;
+ t->prevVolume[1] = vr;
+ t->adjustVolumeRamp(false);
+ }
+
+ // constant gain
+ else {
+ const uint32_t vrl = t->volumeRL;
+ do {
+ uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+ in += 2;
+ out[0] = mulAddRL(1, rl, vrl, out[0]);
+ out[1] = mulAddRL(0, rl, vrl, out[1]);
+ out += 2;
+ } while (--frameCount);
+ }
+ }
t->in = in;
}
-void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp)
+void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
{
int16_t const *in = static_cast<int16_t const *>(t->in);
- // 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];
+ if UNLIKELY(aux != NULL) {
+ // ramp gain
+ if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
+ int32_t vl = t->prevVolume[0];
+ int32_t vr = t->prevVolume[1];
+ int32_t va = t->prevAuxLevel;
+ const int32_t vlInc = t->volumeInc[0];
+ const int32_t vrInc = t->volumeInc[1];
+ const int32_t vaInc = t->auxInc;
- // LOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
- // (vl + vlInc*frameCount)/65536.0f, frameCount);
+ // 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);
+ do {
+ int32_t l = *in++;
+ *out++ += (vl >> 16) * l;
+ *out++ += (vr >> 16) * l;
+ *aux++ += (va >> 16) * l;
+ vl += vlInc;
+ vr += vrInc;
+ va += vaInc;
+ } while (--frameCount);
+
+ t->prevVolume[0] = vl;
+ t->prevVolume[1] = vr;
+ t->prevAuxLevel = va;
+ t->adjustVolumeRamp(true);
+ }
+ // constant gain
+ else {
+ const int16_t vl = t->volume[0];
+ const int16_t vr = t->volume[1];
+ const int16_t va = (int16_t)t->auxLevel;
+ do {
+ int16_t l = *in++;
+ out[0] = mulAdd(l, vl, out[0]);
+ out[1] = mulAdd(l, vr, out[1]);
+ out += 2;
+ aux[0] = mulAdd(l, va, aux[0]);
+ aux++;
+ } while (--frameCount);
+ }
+ } else {
+ // ramp gain
+ if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+ int32_t vl = t->prevVolume[0];
+ int32_t vr = t->prevVolume[1];
+ const int32_t vlInc = t->volumeInc[0];
+ const int32_t vrInc = t->volumeInc[1];
+
+ // LOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+ // t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
+ // (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+ do {
+ int32_t l = *in++;
+ *out++ += (vl >> 16) * l;
+ *out++ += (vr >> 16) * l;
+ vl += vlInc;
+ vr += vrInc;
+ } while (--frameCount);
+
+ t->prevVolume[0] = vl;
+ t->prevVolume[1] = vr;
+ t->adjustVolumeRamp(false);
+ }
+ // constant gain
+ else {
+ const int16_t vl = t->volume[0];
+ const int16_t vr = t->volume[1];
+ do {
+ int16_t l = *in++;
+ out[0] = mulAdd(l, vl, out[0]);
+ out[1] = mulAdd(l, vr, out[1]);
+ out += 2;
+ } while (--frameCount);
+ }
}
t->in = in;
}
@@ -624,37 +836,56 @@ void AudioMixer::ditherAndClamp(int32_t* out, int32_t const *sums, size_t c)
}
// no-op case
-void AudioMixer::process__nop(state_t* state, void* output)
+void AudioMixer::process__nop(state_t* state)
{
- // 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);
+ uint32_t e0 = state->enabledTracks;
+ size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
+ while (e0) {
+ // process by group of tracks with same output buffer to
+ // avoid multiple memset() on same buffer
+ uint32_t e1 = e0, e2 = e0;
+ int i = 31 - __builtin_clz(e1);
+ track_t& t1 = state->tracks[i];
+ e2 &= ~(1<<i);
+ while (e2) {
+ i = 31 - __builtin_clz(e2);
+ e2 &= ~(1<<i);
+ track_t& t2 = state->tracks[i];
+ if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
+ e1 &= ~(1<<i);
+ }
+ }
+ e0 &= ~(e1);
+
+ memset(t1.mainBuffer, 0, bufSize);
+
+ while (e1) {
+ i = 31 - __builtin_clz(e1);
+ e1 &= ~(1<<i);
+ t1 = state->tracks[i];
+ size_t outFrames = state->frameCount;
+ while (outFrames) {
+ t1.buffer.frameCount = outFrames;
+ t1.bufferProvider->getNextBuffer(&t1.buffer);
+ if (!t1.buffer.raw) break;
+ outFrames -= t1.buffer.frameCount;
+ t1.bufferProvider->releaseBuffer(&t1.buffer);
+ }
}
}
}
// generic code without resampling
-void AudioMixer::process__genericNoResampling(state_t* state, void* output)
+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 en = enabledTracks;
- while (en) {
- const int i = 31 - __builtin_clz(en);
- en &= ~(1<<i);
+ 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);
@@ -666,110 +897,156 @@ void AudioMixer::process__genericNoResampling(state_t* state, void* output)
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;
+ e0 = enabledTracks;
+ while (e0) {
+ // process by group of tracks with same output buffer to
+ // optimize cache use
+ uint32_t e1 = e0, e2 = e0;
+ int j = 31 - __builtin_clz(e1);
+ track_t& t1 = state->tracks[j];
+ e2 &= ~(1<<j);
+ while (e2) {
+ j = 31 - __builtin_clz(e2);
+ e2 &= ~(1<<j);
+ track_t& t2 = state->tracks[j];
+ if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
+ e1 &= ~(1<<j);
+ }
+ }
+ e0 &= ~(e1);
+ // this assumes output 16 bits stereo, no resampling
+ int32_t *out = t1.mainBuffer;
+ size_t numFrames = 0;
+ do {
+ memset(outTemp, 0, sizeof(outTemp));
+ e2 = e1;
+ while (e2) {
+ const int i = 31 - __builtin_clz(e2);
+ e2 &= ~(1<<i);
+ track_t& t = state->tracks[i];
+ size_t outFrames = BLOCKSIZE;
+ int32_t *aux = NULL;
+ if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
+ aux = t.auxBuffer + numFrames;
}
- 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;
+ while (outFrames) {
+ size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
+ if (inFrames) {
+ (t.hook)(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux);
+ t.frameCount -= inFrames;
+ outFrames -= inFrames;
+ if UNLIKELY(aux != NULL) {
+ aux += inFrames;
+ }
}
- t.frameCount = t.buffer.frameCount;
- }
+ 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);
-
+ ditherAndClamp(out, outTemp, BLOCKSIZE);
+ out += BLOCKSIZE;
+ numFrames += BLOCKSIZE;
+ } while (numFrames < state->frameCount);
+ }
// release each track's buffer
- en = enabledTracks;
- while (en) {
- const int i = 31 - __builtin_clz(en);
- en &= ~(1<<i);
+ 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, void* output)
+
+ // generic code with resampling
+void AudioMixer::process__genericResampling(state_t* state)
{
int32_t* const outTemp = state->outputTemp;
const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
memset(outTemp, 0, size);
- 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];
+ uint32_t e0 = state->enabledTracks;
+ while (e0) {
+ // process by group of tracks with same output buffer
+ // to optimize cache use
+ uint32_t e1 = e0, e2 = e0;
+ int j = 31 - __builtin_clz(e1);
+ track_t& t1 = state->tracks[j];
+ e2 &= ~(1<<j);
+ while (e2) {
+ j = 31 - __builtin_clz(e2);
+ e2 &= ~(1<<j);
+ track_t& t2 = state->tracks[j];
+ if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
+ e1 &= ~(1<<j);
+ }
+ }
+ e0 &= ~(e1);
+ int32_t *out = t1.mainBuffer;
+ while (e1) {
+ const int i = 31 - __builtin_clz(e1);
+ e1 &= ~(1<<i);
+ track_t& t = state->tracks[i];
+ int32_t *aux = NULL;
+ if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
+ aux = t.auxBuffer;
+ }
- // this is a little goofy, on the resampling case we don't
- // acquire/release the buffers because it's done by
- // the resampler.
- if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
- (t.hook)(&t, outTemp, numFrames, state->resampleTemp);
- } else {
+ // 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 = 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);
+ size_t outFrames = 0;
+
+ while (outFrames < numFrames) {
+ t.buffer.frameCount = numFrames - outFrames;
+ t.bufferProvider->getNextBuffer(&t.buffer);
+ t.in = t.buffer.raw;
+ // t.in == NULL can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (t.in == NULL) break;
+
+ if UNLIKELY(aux != NULL) {
+ aux += outFrames;
+ }
+ (t.hook)(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux);
+ outFrames += t.buffer.frameCount;
+ t.bufferProvider->releaseBuffer(&t.buffer);
+ }
}
}
+ ditherAndClamp(out, outTemp, numFrames);
}
-
- 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)
+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 = static_cast<int32_t*>(output);
+
+ 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;
@@ -787,7 +1064,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, void*
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.
@@ -816,7 +1093,9 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, void*
}
// 2 tracks is also a common case
-void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, void* output)
+// 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;
@@ -829,24 +1108,25 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, void
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);
+
+ //FIXME: only works if two tracks use same buffer
+ int32_t* out = t0.mainBuffer;
size_t numFrames = state->frameCount;
int16_t const *buff = NULL;
-
+
while (numFrames) {
-
+
if (frameCount0 == 0) {
b0.frameCount = numFrames;
t0.bufferProvider->getNextBuffer(&b0);
@@ -875,13 +1155,13 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, void
}
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++;
@@ -896,17 +1176,17 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, void
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;
+ delete [] buff;
}
}
generated by cgit v1.1 at 2024-09-19 18:53:34 +0000