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[media] Rewrite AudioConverter::DownmixAudio

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- Structure the code better
- Directly downmix to mono from multichannel
This commit is contained in:
wolfbeast 2019-05-25 17:44:44 +02:00 • committed by Roy Tam
commit 450910d668
1 changed files with 62 additions and 46 deletions
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108
dom/media/AudioConverter.cpp
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@ -5,8 +5,8 @@
* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
#include "AudioConverter.h"
#include <string.h>
#include <speex/speex_resampler.h>
#include <string.h>
#include <cmath>
/*
@ -140,24 +140,28 @@ static inline int16_t clipTo15(int32_t aX)
size_t
AudioConverter::DownmixAudio(void* aOut, const void* aIn, size_t aFrames) const
{
MOZ_ASSERT(mIn.Format() == AudioConfig::FORMAT_S16 ||
mIn.Format() == AudioConfig::FORMAT_FLT);
MOZ_ASSERT(mIn.Channels() >= mOut.Channels());
MOZ_ASSERT(mIn.Layout() == AudioConfig::ChannelLayout(mIn.Channels()),
"Can only downmix input data in SMPTE layout");
MOZ_ASSERT(mOut.Layout() == AudioConfig::ChannelLayout(2) ||
mOut.Layout() == AudioConfig::ChannelLayout(1));
MOZ_DIAGNOSTIC_ASSERT(mIn.Format() == AudioConfig::FORMAT_S16 ||
mIn.Format() == AudioConfig::FORMAT_FLT);
MOZ_DIAGNOSTIC_ASSERT(mIn.Channels() >= mOut.Channels());
MOZ_DIAGNOSTIC_ASSERT(
mIn.Layout() == AudioConfig::ChannelLayout(mIn.Channels()),
"Can only downmix input data in SMPTE layout");
MOZ_DIAGNOSTIC_ASSERT(mOut.Layout() == AudioConfig::ChannelLayout(2) ||
mOut.Layout() == AudioConfig::ChannelLayout(1),
"Can only downmix to stereo or mono");
uint32_t channels = mIn.Channels();
uint32_t inChannels = mIn.Channels();
uint32_t outChannels = mOut.Channels();
if (channels == 1 && mOut.Channels() == 1) {
if (inChannels == outChannels) {
// Number of channels is equal; no processing needed, just move data.
if (aOut != aIn) {
memmove(aOut, aIn, FramesOutToBytes(aFrames));
}
return aFrames;
}
if (channels > 2) {
if (inChannels > 2) {
if (mIn.Format() == AudioConfig::FORMAT_FLT) {
// Downmix matrix. Per-row normalization 1 for rows 3,4 and 2 for rows 5-8.
static const float dmatrix[6][8][2]= {
@ -174,12 +178,18 @@ AudioConverter::DownmixAudio(void* aOut, const void* aIn, size_t aFrames) const
for (uint32_t i = 0; i < aFrames; i++) {
float sampL = 0.0;
float sampR = 0.0;
for (uint32_t j = 0; j < channels; j++) {
sampL += in[i*mIn.Channels()+j]*dmatrix[mIn.Channels()-3][j][0];
sampR += in[i*mIn.Channels()+j]*dmatrix[mIn.Channels()-3][j][1];
for (uint32_t j = 0; j < inChannels; j++) {
sampL += in[i * inChannels + j] * dmatrix[inChannels - 3][j][0];
sampR += in[i * inChannels + j] * dmatrix[inChannels - 3][j][1];
}
if (outChannels == 2) {
// Stereo
*out++ = sampL;
*out++ = sampR;
} else {
// Mono
*out++ = (sampL + sampR) * 0.5;
}
*out++ = sampL;
*out++ = sampR;
}
} else if (mIn.Format() == AudioConfig::FORMAT_S16) {
// Downmix matrix. Per-row normalization 1 for rows 3,4 and 2 for rows 5-8.
@ -198,45 +208,51 @@ AudioConverter::DownmixAudio(void* aOut, const void* aIn, size_t aFrames) const
for (uint32_t i = 0; i < aFrames; i++) {
int32_t sampL = 0;
int32_t sampR = 0;
for (uint32_t j = 0; j < channels; j++) {
sampL+=in[i*channels+j]*dmatrix[channels-3][j][0];
sampR+=in[i*channels+j]*dmatrix[channels-3][j][1];
for (uint32_t j = 0; j < inChannels; j++) {
sampL += in[i * inChannels + j] * dmatrix[inChannels - 3][j][0];
sampR += in[i * inChannels + j] * dmatrix[inChannels - 3][j][1];
}
sampL = clipTo15((sampL + 8192) >> 14);
sampR = clipTo15((sampR + 8192) >> 14);
if (outChannels == 2) {
// Stereo
*out++ = sampL;
*out++ = sampR;
} else {
// Mono
*out++ = (sampL + sampR) * 0.5;
}
*out++ = clipTo15((sampL + 8192)>>14);
*out++ = clipTo15((sampR + 8192)>>14);
}
} else {
MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
}
// If we are to continue downmixing to mono, start working on the output
// buffer.
aIn = aOut;
channels = 2;
return aFrames;
}
if (mOut.Channels() == 1) {
if (mIn.Format() == AudioConfig::FORMAT_FLT) {
const float* in = static_cast<const float*>(aIn);
float* out = static_cast<float*>(aOut);
for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
float sample = 0.0;
// The sample of the buffer would be interleaved.
sample = (in[fIdx*channels] + in[fIdx*channels + 1]) * 0.5;
*out++ = sample;
}
} else if (mIn.Format() == AudioConfig::FORMAT_S16) {
const int16_t* in = static_cast<const int16_t*>(aIn);
int16_t* out = static_cast<int16_t*>(aOut);
for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
int32_t sample = 0.0;
// The sample of the buffer would be interleaved.
sample = (in[fIdx*channels] + in[fIdx*channels + 1]) * 0.5;
*out++ = sample;
}
} else {
MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
// If we get here, we're doing a stereo -> mono conversion.
MOZ_DIAGNOSTIC_ASSERT(inChannels == 2 && outChannels == 1);
if (mIn.Format() == AudioConfig::FORMAT_FLT) {
const float* in = static_cast<const float*>(aIn);
float* out = static_cast<float*>(aOut);
for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
float sample = 0.0;
// The sample of the buffer would be interleaved.
sample = (in[fIdx * inChannels] + in[fIdx * inChannels + 1]) * 0.5;
*out++ = sample;
}
} else if (mIn.Format() == AudioConfig::FORMAT_S16) {
const int16_t* in = static_cast<const int16_t*>(aIn);
int16_t* out = static_cast<int16_t*>(aOut);
for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
int32_t sample = 0.0;
// The sample of the buffer would be interleaved.
sample = (in[fIdx * inChannels] + in[fIdx * inChannels + 1]) * 0.5;
*out++ = sample;
}
} else {
MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
}
return aFrames;
}