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Merge "AudioEffect: Convert aux buffer format from q4.27 to float"

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gugelfrei
Andy Hung 7 years ago committed by Android (Google) Code Review
parent 61e916dd6b 116a49881b
commit d466435670
5 changed files with 97 additions and 45 deletions
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3
media/libaudioclient/include/media/AudioMixer.h
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@ -33,6 +33,9 @@
// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
// This must match frameworks/av/services/audioflinger/Configuration.h
#define FLOAT_AUX
namespace android {
// ----------------------------------------------------------------------------

105
media/libaudioprocessing/AudioMixer.cpp
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@ -71,12 +71,20 @@ T max(const T& x, const T& y) {
// Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
// original code will be used for stereo sinks, the new mixer for multichannel.
static const bool kUseNewMixer = true;
static constexpr bool kUseNewMixer = true;
// Set kUseFloat to true to allow floating input into the mixer engine.
// If kUseNewMixer is false, this is ignored or may be overridden internally
// because of downmix/upmix support.
static const bool kUseFloat = true;
static constexpr bool kUseFloat = true;
#ifdef FLOAT_AUX
using TYPE_AUX = float;
static_assert(kUseNewMixer && kUseFloat,
"kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
#else
using TYPE_AUX = int32_t; // q4.27
#endif
// Set to default copy buffer size in frames for input processing.
static const size_t kCopyBufferFrameCount = 256;
@ -861,16 +869,25 @@ inline void AudioMixer::track_t::adjustVolumeRamp(bool aux, bool useFloat)
}
}
}
/* TODO: aux is always integer regardless of output buffer type */
if (aux) {
if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) ||
((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) {
#ifdef FLOAT_AUX
if (useFloat) {
if ((mAuxInc > 0.f && mPrevAuxLevel + mAuxInc >= mAuxLevel) ||
(mAuxInc < 0.f && mPrevAuxLevel + mAuxInc <= mAuxLevel)) {
auxInc = 0;
prevAuxLevel = auxLevel << 16;
mAuxInc = 0.f;
mPrevAuxLevel = mAuxLevel;
}
} else
#endif
if ((auxInc > 0 && ((prevAuxLevel + auxInc) >> 16) >= auxLevel) ||
(auxInc < 0 && ((prevAuxLevel + auxInc) >> 16) <= auxLevel)) {
auxInc = 0;
prevAuxLevel = auxLevel << 16;
mAuxInc = 0.;
mAuxInc = 0.f;
mPrevAuxLevel = mAuxLevel;
} else {
//ALOGV("aux ramp: %d %d %d", auxLevel << 16, prevAuxLevel, auxInc);
}
}
}
@ -1694,7 +1711,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
* TO: int32_t (Q4.27) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TA: int32_t (Q4.27)
* TA: int32_t (Q4.27) or float
*/
template <int MIXTYPE,
typename TO, typename TI, typename TV, typename TA, typename TAV>
@ -1738,7 +1755,7 @@ static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount,
/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
* TO: int32_t (Q4.27) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TA: int32_t (Q4.27)
* TA: int32_t (Q4.27) or float
*/
template <int MIXTYPE,
typename TO, typename TI, typename TV, typename TA, typename TAV>
@ -1778,7 +1795,7 @@ static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
* ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
* TO: int32_t (Q4.27) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TA: int32_t (Q4.27)
* TA: int32_t (Q4.27) or float
*/
template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
typename TO, typename TI, typename TA>
@ -1788,13 +1805,25 @@ void AudioMixer::volumeMix(TO *out, size_t outFrames,
if (USEFLOATVOL) {
if (ramp) {
volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
t->mPrevVolume, t->mVolumeInc, &t->prevAuxLevel, t->auxInc);
t->mPrevVolume, t->mVolumeInc,
#ifdef FLOAT_AUX
&t->mPrevAuxLevel, t->mAuxInc
#else
&t->prevAuxLevel, t->auxInc
#endif
);
if (ADJUSTVOL) {
t->adjustVolumeRamp(aux != NULL, true);
}
} else {
volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
t->mVolume, t->auxLevel);
t->mVolume,
#ifdef FLOAT_AUX
t->mAuxLevel
#else
t->auxLevel
#endif
);
}
} else {
if (ramp) {
@ -1851,7 +1880,7 @@ void AudioMixer::process_NoResampleOneTrack(state_t* state)
}
const size_t outFrames = b.frameCount;
volumeMix<MIXTYPE, is_same<TI, float>::value, false> (
volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, false /* ADJUSTVOL */> (
out, outFrames, in, aux, ramp, t);
out += outFrames * channels;
@ -1874,7 +1903,7 @@ void AudioMixer::process_NoResampleOneTrack(state_t* state)
* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
* TO: int32_t (Q4.27) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TA: int32_t (Q4.27)
* TA: int32_t (Q4.27) or float
*/
template <int MIXTYPE, typename TO, typename TI, typename TA>
void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* temp, TA* aux)
@ -1890,7 +1919,7 @@ void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO*
memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(TO));
t->resampler->resample((int32_t*)temp, outFrameCount, t->bufferProvider);
volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
out, outFrameCount, temp, aux, ramp, t);
} else { // constant volume gain
@ -1905,7 +1934,7 @@ void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO*
* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
* TO: int32_t (Q4.27) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TA: int32_t (Q4.27)
* TA: int32_t (Q4.27) or float
*/
template <int MIXTYPE, typename TO, typename TI, typename TA>
void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount,
@ -1914,7 +1943,7 @@ void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount,
ALOGVV("track__NoResample\n");
const TI *in = static_cast<const TI *>(t->in);
volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
out, frameCount, in, aux, t->needsRamp(), t);
// MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
@ -1990,11 +2019,11 @@ AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount
case TRACKTYPE_RESAMPLE:
switch (mixerInFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
return (AudioMixer::hook_t)
track__Resample<MIXTYPE_MULTI, float /*TO*/, float /*TI*/, int32_t /*TA*/>;
return (AudioMixer::hook_t)track__Resample<
MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
case AUDIO_FORMAT_PCM_16_BIT:
return (AudioMixer::hook_t)\
track__Resample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
return (AudioMixer::hook_t)track__Resample<
MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
default:
LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
break;
@ -2003,11 +2032,11 @@ AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount
case TRACKTYPE_NORESAMPLEMONO:
switch (mixerInFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
return (AudioMixer::hook_t)
track__NoResample<MIXTYPE_MONOEXPAND, float, float, int32_t>;
return (AudioMixer::hook_t)track__NoResample<
MIXTYPE_MONOEXPAND, float /*TO*/, float /*TI*/, TYPE_AUX>;
case AUDIO_FORMAT_PCM_16_BIT:
return (AudioMixer::hook_t)
track__NoResample<MIXTYPE_MONOEXPAND, int32_t, int16_t, int32_t>;
return (AudioMixer::hook_t)track__NoResample<
MIXTYPE_MONOEXPAND, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
default:
LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
break;
@ -2016,11 +2045,11 @@ AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount
case TRACKTYPE_NORESAMPLE:
switch (mixerInFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
return (AudioMixer::hook_t)
track__NoResample<MIXTYPE_MULTI, float, float, int32_t>;
return (AudioMixer::hook_t)track__NoResample<
MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
case AUDIO_FORMAT_PCM_16_BIT:
return (AudioMixer::hook_t)
track__NoResample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
return (AudioMixer::hook_t)track__NoResample<
MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
default:
LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
break;
@ -2055,11 +2084,11 @@ AudioMixer::process_hook_t AudioMixer::getProcessHook(int processType, uint32_t
case AUDIO_FORMAT_PCM_FLOAT:
switch (mixerOutFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
float /*TO*/, float /*TI*/, int32_t /*TA*/>;
return process_NoResampleOneTrack<
MIXTYPE_MULTI_SAVEONLY, float /*TO*/, float /*TI*/, TYPE_AUX>;
case AUDIO_FORMAT_PCM_16_BIT:
return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
int16_t, float, int32_t>;
return process_NoResampleOneTrack<
MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, float /*TI*/, TYPE_AUX>;
default:
LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
break;
@ -2068,11 +2097,11 @@ AudioMixer::process_hook_t AudioMixer::getProcessHook(int processType, uint32_t
case AUDIO_FORMAT_PCM_16_BIT:
switch (mixerOutFormat) {
case AUDIO_FORMAT_PCM_FLOAT:
return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
float, int16_t, int32_t>;
return process_NoResampleOneTrack<
MIXTYPE_MULTI_SAVEONLY, float /*TO*/, int16_t /*TI*/, TYPE_AUX>;
case AUDIO_FORMAT_PCM_16_BIT:
return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
int16_t, int16_t, int32_t>;
return process_NoResampleOneTrack<
MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
default:
LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
break;

11
media/libaudioprocessing/AudioMixerOps.h
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@ -188,13 +188,13 @@ inline void MixAccum(TO *auxaccum, TI value) {
template<>
inline void MixAccum<float, int16_t>(float *auxaccum, int16_t value) {
static const float norm = 1. / (1 << 15);
static constexpr float norm = 1. / (1 << 15);
*auxaccum += norm * value;
}
template<>
inline void MixAccum<float, int32_t>(float *auxaccum, int32_t value) {
static const float norm = 1. / (1 << 27);
static constexpr float norm = 1. / (1 << 27);
*auxaccum += norm * value;
}
@ -238,6 +238,7 @@ enum {
* NCHAN represents number of input and output channels.
* TO: int32_t (Q4.27) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TA: int32_t (Q4.27) or float
* TV: int32_t (U4.28) or int16_t (U4.12) or float
* vol: represents a volume array.
*
@ -247,7 +248,8 @@ enum {
* Single input channel. NCHAN represents number of output channels.
* TO: int32_t (Q4.27) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TV: int32_t (U4.28) or int16_t (U4.12) or float
* TA: int32_t (Q4.27) or float
* TV/TAV: int32_t (U4.28) or int16_t (U4.12) or float
* Input channel count is 1.
* vol: represents volume array.
*
@ -257,7 +259,8 @@ enum {
* NCHAN represents number of input and output channels.
* TO: int16_t (Q.15) or float
* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
* TV: int32_t (U4.28) or int16_t (U4.12) or float
* TA: int32_t (Q4.27) or float
* TV/TAV: int32_t (U4.28) or int16_t (U4.12) or float
* vol: represents a volume array.
*
* MIXTYPE_MULTI_SAVEONLY does not accumulate into the out pointer.

5
services/audioflinger/Configuration.h
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@ -44,4 +44,9 @@
// define FLOAT_EFFECT_CHAIN to request float effects (falls back to int16_t if unavailable)
#define FLOAT_EFFECT_CHAIN
#ifdef FLOAT_EFFECT_CHAIN
// define FLOAT_AUX to process aux effect buffers in float (FLOAT_EFFECT_CHAIN must be defined)
#define FLOAT_AUX
#endif
#endif // ANDROID_AUDIOFLINGER_CONFIGURATION_H

18
services/audioflinger/Effects.cpp
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@ -342,9 +342,9 @@ void AudioFlinger::EffectModule::process()
if (isProcessImplemented()) {
if (auxType) {
// We overwrite the aux input buffer here and clear after processing.
// Note that aux input buffers are format q4_27.
#ifdef FLOAT_EFFECT_CHAIN
if (mSupportsFloat) {
#ifndef FLOAT_AUX
// Do in-place float conversion for auxiliary effect input buffer.
static_assert(sizeof(float) <= sizeof(int32_t),
"in-place conversion requires sizeof(float) <= sizeof(int32_t)");
@ -353,13 +353,21 @@ void AudioFlinger::EffectModule::process()
mConfig.inputCfg.buffer.f32,
mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.frameCount);
#endif // !FLOAT_AUX
} else
#endif
#endif // FLOAT_EFFECT_CHAIN
{
#ifdef FLOAT_AUX
memcpy_to_i16_from_float(
mConfig.inputCfg.buffer.s16,
mConfig.inputCfg.buffer.f32,
mConfig.inputCfg.buffer.frameCount);
#else
memcpy_to_i16_from_q4_27(
mConfig.inputCfg.buffer.s16,
mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.frameCount);
#endif
}
}
#ifdef FLOAT_EFFECT_CHAIN
@ -419,9 +427,13 @@ void AudioFlinger::EffectModule::process()
// clear auxiliary effect input buffer for next accumulation
if (auxType) {
// input always q4_27 regardless of FLOAT_EFFECT_CHAIN.
#ifdef FLOAT_AUX
const size_t size =
mConfig.inputCfg.buffer.frameCount * inChannelCount * sizeof(float);
#else
const size_t size =
mConfig.inputCfg.buffer.frameCount * inChannelCount * sizeof(int32_t);
#endif
memset(mConfig.inputCfg.buffer.raw, 0, size);
}
} else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&

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