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AudioEffect: clean up format conversion in EffectModule::process()

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Editing for clarity, no change in operation.

Test: Solo Tester
Bug: 69984406
Change-Id: I683fd9b83417c3d1c6d63339537d03c55625ac13
gugelfrei
Andy Hung 7 years ago
parent 14093dd03c
commit fa69ca3012
1 changed files with 79 additions and 84 deletions
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163
services/audioflinger/Effects.cpp
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@ -296,6 +296,43 @@ void AudioFlinger::EffectModule::process()
const bool auxType =
(mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY;
// safeInputOutputSampleCount is 0 if the channel count between input and output
// buffers do not match. This prevents automatic accumulation or copying between the
// input and output effect buffers without an intermediary effect process.
// TODO: consider implementing channel conversion.
const size_t safeInputOutputSampleCount =
inChannelCount != outChannelCount ? 0
: outChannelCount * std::min(
mConfig.inputCfg.buffer.frameCount,
mConfig.outputCfg.buffer.frameCount);
const auto accumulateInputToOutput = [this, safeInputOutputSampleCount]() {
#ifdef FLOAT_EFFECT_CHAIN
accumulate_float(
mConfig.outputCfg.buffer.f32,
mConfig.inputCfg.buffer.f32,
safeInputOutputSampleCount);
#else
accumulate_i16(
mConfig.outputCfg.buffer.s16,
mConfig.inputCfg.buffer.s16,
safeInputOutputSampleCount);
#endif
};
const auto copyInputToOutput = [this, safeInputOutputSampleCount]() {
#ifdef FLOAT_EFFECT_CHAIN
memcpy(
mConfig.outputCfg.buffer.f32,
mConfig.inputCfg.buffer.f32,
safeInputOutputSampleCount * sizeof(*mConfig.outputCfg.buffer.f32));
#else
memcpy(
mConfig.outputCfg.buffer.s16,
mConfig.inputCfg.buffer.s16,
safeInputOutputSampleCount * sizeof(*mConfig.outputCfg.buffer.s16));
#endif
};
if (isProcessEnabled()) {
int ret;
if (isProcessImplemented()) {
@ -308,97 +345,69 @@ void AudioFlinger::EffectModule::process()
static_assert(sizeof(float) <= sizeof(int32_t),
"in-place conversion requires sizeof(float) <= sizeof(int32_t)");
const int32_t * const p32 = mConfig.inputCfg.buffer.s32;
float * const pFloat = mConfig.inputCfg.buffer.f32;
memcpy_to_float_from_q4_27(pFloat, p32, mConfig.inputCfg.buffer.frameCount);
} else {
memcpy_to_i16_from_q4_27(mConfig.inputCfg.buffer.s16,
memcpy_to_float_from_q4_27(
mConfig.inputCfg.buffer.f32,
mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.frameCount);
}
#else
memcpy_to_i16_from_q4_27(mConfig.inputCfg.buffer.s16,
mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.frameCount);
} else
#endif
{
memcpy_to_i16_from_q4_27(
mConfig.inputCfg.buffer.s16,
mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.frameCount);
}
}
#ifdef FLOAT_EFFECT_CHAIN
if (mSupportsFloat) {
ret = mEffectInterface->process();
} else {
{ // convert input to int16_t as effect doesn't support float.
if (!auxType) {
if (mInConversionBuffer.get() == nullptr) {
ALOGW("%s: mInConversionBuffer is null, bypassing", __func__);
goto data_bypass;
}
const float * const pIn = mInBuffer->audioBuffer()->f32;
int16_t * const pIn16 = mInConversionBuffer->audioBuffer()->s16;
memcpy_to_i16_from_float(
pIn16, pIn, inChannelCount * mConfig.inputCfg.buffer.frameCount);
if (!mSupportsFloat) { // convert input to int16_t as effect doesn't support float.
if (!auxType) {
if (mInConversionBuffer.get() == nullptr) {
ALOGW("%s: mInConversionBuffer is null, bypassing", __func__);
goto data_bypass;
}
if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
if (mOutConversionBuffer.get() == nullptr) {
ALOGW("%s: mOutConversionBuffer is null, bypassing", __func__);
goto data_bypass;
}
int16_t * const pOut16 = mOutConversionBuffer->audioBuffer()->s16;
const float * const pOut = mOutBuffer->audioBuffer()->f32;
memcpy_to_i16_from_float(
pOut16,
pOut,
outChannelCount * mConfig.outputCfg.buffer.frameCount);
memcpy_to_i16_from_float(
mInConversionBuffer->audioBuffer()->s16,
mInBuffer->audioBuffer()->f32,
inChannelCount * mConfig.inputCfg.buffer.frameCount);
}
if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
if (mOutConversionBuffer.get() == nullptr) {
ALOGW("%s: mOutConversionBuffer is null, bypassing", __func__);
goto data_bypass;
}
memcpy_to_i16_from_float(
mOutConversionBuffer->audioBuffer()->s16,
mOutBuffer->audioBuffer()->f32,
outChannelCount * mConfig.outputCfg.buffer.frameCount);
}
}
#endif
ret = mEffectInterface->process();
ret = mEffectInterface->process();
{ // convert output back to float.
const int16_t * const pOut16 = mOutConversionBuffer->audioBuffer()->s16;
float * const pOut = mOutBuffer->audioBuffer()->f32;
memcpy_to_float_from_i16(
pOut, pOut16, outChannelCount * mConfig.outputCfg.buffer.frameCount);
}
#ifdef FLOAT_EFFECT_CHAIN
if (!mSupportsFloat) { // convert output int16_t back to float.
memcpy_to_float_from_i16(
mOutBuffer->audioBuffer()->f32,
mOutConversionBuffer->audioBuffer()->s16,
outChannelCount * mConfig.outputCfg.buffer.frameCount);
}
#else
ret = mEffectInterface->process();
#endif
} else {
#ifdef FLOAT_EFFECT_CHAIN
data_bypass:
#endif
if (!auxType /* aux effects do not require data bypass */
&& mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw
&& inChannelCount == outChannelCount) {
const size_t sampleCount = std::min(
mConfig.inputCfg.buffer.frameCount,
mConfig.outputCfg.buffer.frameCount) * outChannelCount;
#ifdef FLOAT_EFFECT_CHAIN
const float * const in = mConfig.inputCfg.buffer.f32;
float * const out = mConfig.outputCfg.buffer.f32;
&& mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
accumulate_float(out, in, sampleCount);
accumulateInputToOutput();
} else {
memcpy(mConfig.outputCfg.buffer.f32, mConfig.inputCfg.buffer.f32,
sampleCount * sizeof(*mConfig.outputCfg.buffer.f32));
copyInputToOutput();
}
#else
const int16_t * const in = mConfig.inputCfg.buffer.s16;
int16_t * const out = mConfig.outputCfg.buffer.s16;
if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
accumulate_i16(out, in, sampleCount);
} else {
memcpy(mConfig.outputCfg.buffer.s16, mConfig.inputCfg.buffer.s16,
sampleCount * sizeof(*mConfig.outputCfg.buffer.s16));
}
#endif
}
ret = -ENODATA;
}
// force transition to IDLE state when engine is ready
if (mState == STOPPED && ret == -ENODATA) {
mDisableWaitCnt = 1;
@ -417,21 +426,8 @@ void AudioFlinger::EffectModule::process()
// If an insert effect is idle and input buffer is different from output buffer,
// accumulate input onto output
sp<EffectChain> chain = mChain.promote();
if (chain != 0
&& chain->activeTrackCnt() != 0
&& inChannelCount == outChannelCount) {
const size_t sampleCount = std::min(
mConfig.inputCfg.buffer.frameCount,
mConfig.outputCfg.buffer.frameCount) * outChannelCount;
#ifdef FLOAT_EFFECT_CHAIN
const float * const in = mConfig.inputCfg.buffer.f32;
float * const out = mConfig.outputCfg.buffer.f32;
accumulate_float(out, in, sampleCount);
#else
const int16_t * const in = mConfig.inputCfg.buffer.s16;
int16_t * const out = mConfig.outputCfg.buffer.s16;
accumulate_i16(out, in, sampleCount);
#endif
if (chain.get() != nullptr && chain->activeTrackCnt() != 0) {
accumulateInputToOutput();
}
}
}
@ -927,7 +923,6 @@ void AudioFlinger::EffectModule::setInBuffer(const sp<EffectBufferHalInterface>&
(void)EffectBufferHalInterface::allocate(size, &mInConversionBuffer);
}
if (mInConversionBuffer.get() != nullptr) {
// FIXME: confirm buffer has enough size.
mInConversionBuffer->setFrameCount(inFrameCount);
mEffectInterface->setInBuffer(mInConversionBuffer);
} else if (size > 0) {

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