mirror of
https://github.com/mozilla/gecko-dev.git
synced 2024-11-25 05:41:12 +00:00
7ec9b68802
No need to first downmix to stereo and then mono. Differential Revision: https://phabricator.services.mozilla.com/D27216 --HG-- extra : moz-landing-system : lando
466 lines
16 KiB
C++
466 lines
16 KiB
C++
/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
|
|
/* vim: set ts=8 sts=2 et sw=2 tw=80: */
|
|
/* This Source Code Form is subject to the terms of the Mozilla Public
|
|
* License, v. 2.0. If a copy of the MPL was not distributed with this
|
|
* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
|
|
|
|
#include "AudioConverter.h"
|
|
#include <speex/speex_resampler.h>
|
|
#include <string.h>
|
|
#include <cmath>
|
|
|
|
/*
|
|
* Parts derived from MythTV AudioConvert Class
|
|
* Created by Jean-Yves Avenard.
|
|
*
|
|
* Copyright (C) Bubblestuff Pty Ltd 2013
|
|
* Copyright (C) foobum@gmail.com 2010
|
|
*/
|
|
|
|
namespace mozilla {
|
|
|
|
AudioConverter::AudioConverter(const AudioConfig& aIn, const AudioConfig& aOut)
|
|
: mIn(aIn), mOut(aOut), mResampler(nullptr) {
|
|
MOZ_DIAGNOSTIC_ASSERT(
|
|
aIn.Format() == aOut.Format() && aIn.Interleaved() == aOut.Interleaved(),
|
|
"No format or rate conversion is supported at this stage");
|
|
MOZ_DIAGNOSTIC_ASSERT(
|
|
aOut.Channels() <= 2 || aIn.Channels() == aOut.Channels(),
|
|
"Only down/upmixing to mono or stereo is supported at this stage");
|
|
MOZ_DIAGNOSTIC_ASSERT(aOut.Interleaved(),
|
|
"planar audio format not supported");
|
|
mIn.Layout().MappingTable(mOut.Layout(), &mChannelOrderMap);
|
|
if (aIn.Rate() != aOut.Rate()) {
|
|
RecreateResampler();
|
|
}
|
|
}
|
|
|
|
AudioConverter::~AudioConverter() {
|
|
if (mResampler) {
|
|
speex_resampler_destroy(mResampler);
|
|
mResampler = nullptr;
|
|
}
|
|
}
|
|
|
|
bool AudioConverter::CanWorkInPlace() const {
|
|
bool needDownmix = mIn.Channels() > mOut.Channels();
|
|
bool needUpmix = mIn.Channels() < mOut.Channels();
|
|
bool canDownmixInPlace =
|
|
mIn.Channels() * AudioConfig::SampleSize(mIn.Format()) >=
|
|
mOut.Channels() * AudioConfig::SampleSize(mOut.Format());
|
|
bool needResample = mIn.Rate() != mOut.Rate();
|
|
bool canResampleInPlace = mIn.Rate() >= mOut.Rate();
|
|
// We should be able to work in place if 1s of audio input takes less space
|
|
// than 1s of audio output. However, as we downmix before resampling we can't
|
|
// perform any upsampling in place (e.g. if incoming rate >= outgoing rate)
|
|
return !needUpmix && (!needDownmix || canDownmixInPlace) &&
|
|
(!needResample || canResampleInPlace);
|
|
}
|
|
|
|
size_t AudioConverter::ProcessInternal(void* aOut, const void* aIn,
|
|
size_t aFrames) {
|
|
if (!aFrames) {
|
|
return 0;
|
|
}
|
|
if (mIn.Channels() > mOut.Channels()) {
|
|
return DownmixAudio(aOut, aIn, aFrames);
|
|
} else if (mIn.Channels() < mOut.Channels()) {
|
|
return UpmixAudio(aOut, aIn, aFrames);
|
|
} else if (mIn.Layout() != mOut.Layout() && CanReorderAudio()) {
|
|
ReOrderInterleavedChannels(aOut, aIn, aFrames);
|
|
} else if (aIn != aOut) {
|
|
memmove(aOut, aIn, FramesOutToBytes(aFrames));
|
|
}
|
|
return aFrames;
|
|
}
|
|
|
|
// Reorder interleaved channels.
|
|
// Can work in place (e.g aOut == aIn).
|
|
template <class AudioDataType>
|
|
void _ReOrderInterleavedChannels(AudioDataType* aOut, const AudioDataType* aIn,
|
|
uint32_t aFrames, uint32_t aChannels,
|
|
const uint8_t* aChannelOrderMap) {
|
|
MOZ_DIAGNOSTIC_ASSERT(aChannels <= AudioConfig::ChannelLayout::MAX_CHANNELS);
|
|
AudioDataType val[AudioConfig::ChannelLayout::MAX_CHANNELS];
|
|
for (uint32_t i = 0; i < aFrames; i++) {
|
|
for (uint32_t j = 0; j < aChannels; j++) {
|
|
val[j] = aIn[aChannelOrderMap[j]];
|
|
}
|
|
for (uint32_t j = 0; j < aChannels; j++) {
|
|
aOut[j] = val[j];
|
|
}
|
|
aOut += aChannels;
|
|
aIn += aChannels;
|
|
}
|
|
}
|
|
|
|
void AudioConverter::ReOrderInterleavedChannels(void* aOut, const void* aIn,
|
|
size_t aFrames) const {
|
|
MOZ_DIAGNOSTIC_ASSERT(mIn.Channels() == mOut.Channels());
|
|
MOZ_DIAGNOSTIC_ASSERT(CanReorderAudio());
|
|
|
|
if (mChannelOrderMap.IsEmpty() || mOut.Channels() == 1 ||
|
|
mOut.Layout() == mIn.Layout()) {
|
|
// If channel count is 1, planar and non-planar formats are the same or
|
|
// there's nothing to reorder, or if we don't know how to re-order.
|
|
if (aOut != aIn) {
|
|
memmove(aOut, aIn, FramesOutToBytes(aFrames));
|
|
}
|
|
return;
|
|
}
|
|
|
|
uint32_t bits = AudioConfig::FormatToBits(mOut.Format());
|
|
switch (bits) {
|
|
case 8:
|
|
_ReOrderInterleavedChannels((uint8_t*)aOut, (const uint8_t*)aIn, aFrames,
|
|
mIn.Channels(), mChannelOrderMap.Elements());
|
|
break;
|
|
case 16:
|
|
_ReOrderInterleavedChannels((int16_t*)aOut, (const int16_t*)aIn, aFrames,
|
|
mIn.Channels(), mChannelOrderMap.Elements());
|
|
break;
|
|
default:
|
|
MOZ_DIAGNOSTIC_ASSERT(AudioConfig::SampleSize(mOut.Format()) == 4);
|
|
_ReOrderInterleavedChannels((int32_t*)aOut, (const int32_t*)aIn, aFrames,
|
|
mIn.Channels(), mChannelOrderMap.Elements());
|
|
break;
|
|
}
|
|
}
|
|
|
|
static inline int16_t clipTo15(int32_t aX) {
|
|
return aX < -32768 ? -32768 : aX <= 32767 ? aX : 32767;
|
|
}
|
|
|
|
template <typename TYPE>
|
|
static void dumbUpDownMix(TYPE* aOut, int32_t aOutChannels, const TYPE* aIn,
|
|
int32_t aInChannels, int32_t aFrames) {
|
|
if (aIn == aOut) {
|
|
return;
|
|
}
|
|
int32_t commonChannels = std::min(aInChannels, aOutChannels);
|
|
|
|
for (int32_t i = 0; i < aFrames; i++) {
|
|
for (int32_t j = 0; j < commonChannels; j++) {
|
|
aOut[i * aOutChannels + j] = aIn[i * aInChannels + j];
|
|
}
|
|
for (int32_t j = 0; j < aInChannels - aOutChannels; j++) {
|
|
aOut[i * aOutChannels + j] = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
size_t AudioConverter::DownmixAudio(void* aOut, const void* aIn,
|
|
size_t aFrames) const {
|
|
MOZ_DIAGNOSTIC_ASSERT(mIn.Format() == AudioConfig::FORMAT_S16 ||
|
|
mIn.Format() == AudioConfig::FORMAT_FLT);
|
|
MOZ_DIAGNOSTIC_ASSERT(mIn.Channels() >= mOut.Channels());
|
|
MOZ_DIAGNOSTIC_ASSERT(mOut.Layout() == AudioConfig::ChannelLayout(2) ||
|
|
mOut.Layout() == AudioConfig::ChannelLayout(1));
|
|
|
|
uint32_t inChannels = mIn.Channels();
|
|
uint32_t outChannels = mOut.Channels();
|
|
|
|
if (inChannels == outChannels) {
|
|
if (aOut != aIn) {
|
|
memmove(aOut, aIn, FramesOutToBytes(aFrames));
|
|
}
|
|
return aFrames;
|
|
}
|
|
|
|
if (!mIn.Layout().IsValid() || !mOut.Layout().IsValid()) {
|
|
// Dumb copy dropping extra channels.
|
|
if (mIn.Format() == AudioConfig::FORMAT_FLT) {
|
|
dumbUpDownMix(static_cast<float*>(aOut), outChannels,
|
|
static_cast<const float*>(aIn), inChannels, aFrames);
|
|
} else if (mIn.Format() == AudioConfig::FORMAT_S16) {
|
|
dumbUpDownMix(static_cast<int16_t*>(aOut), outChannels,
|
|
static_cast<const int16_t*>(aIn), inChannels, aFrames);
|
|
} else {
|
|
MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
|
|
}
|
|
return aFrames;
|
|
}
|
|
|
|
MOZ_ASSERT(
|
|
mIn.Layout() == AudioConfig::ChannelLayout::SMPTEDefault(mIn.Layout()),
|
|
"Can only downmix input data in SMPTE layout");
|
|
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] = {
|
|
/*3*/ {{0.5858f, 0}, {0, 0.5858f}, {0.4142f, 0.4142f}},
|
|
/*4*/
|
|
{{0.4226f, 0}, {0, 0.4226f}, {0.366f, 0.2114f}, {0.2114f, 0.366f}},
|
|
/*5*/
|
|
{{0.6510f, 0},
|
|
{0, 0.6510f},
|
|
{0.4600f, 0.4600f},
|
|
{0.5636f, 0.3254f},
|
|
{0.3254f, 0.5636f}},
|
|
/*6*/
|
|
{{0.5290f, 0},
|
|
{0, 0.5290f},
|
|
{0.3741f, 0.3741f},
|
|
{0.3741f, 0.3741f},
|
|
{0.4582f, 0.2645f},
|
|
{0.2645f, 0.4582f}},
|
|
/*7*/
|
|
{{0.4553f, 0},
|
|
{0, 0.4553f},
|
|
{0.3220f, 0.3220f},
|
|
{0.3220f, 0.3220f},
|
|
{0.2788f, 0.2788f},
|
|
{0.3943f, 0.2277f},
|
|
{0.2277f, 0.3943f}},
|
|
/*8*/
|
|
{{0.3886f, 0},
|
|
{0, 0.3886f},
|
|
{0.2748f, 0.2748f},
|
|
{0.2748f, 0.2748f},
|
|
{0.3366f, 0.1943f},
|
|
{0.1943f, 0.3366f},
|
|
{0.3366f, 0.1943f},
|
|
{0.1943f, 0.3366f}},
|
|
};
|
|
// Re-write the buffer with downmixed data
|
|
const float* in = static_cast<const float*>(aIn);
|
|
float* out = static_cast<float*>(aOut);
|
|
for (uint32_t i = 0; i < aFrames; i++) {
|
|
float sampL = 0.0;
|
|
float sampR = 0.0;
|
|
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) {
|
|
*out++ = sampL;
|
|
*out++ = sampR;
|
|
} else {
|
|
*out++ = (sampL + sampR) * 0.5;
|
|
}
|
|
}
|
|
} else if (mIn.Format() == AudioConfig::FORMAT_S16) {
|
|
// Downmix matrix. Per-row normalization 1 for rows 3,4 and 2 for rows
|
|
// 5-8. Coefficients in Q14.
|
|
static const int16_t dmatrix[6][8][2] = {
|
|
/*3*/ {{9598, 0}, {0, 9598}, {6786, 6786}},
|
|
/*4*/ {{6925, 0}, {0, 6925}, {5997, 3462}, {3462, 5997}},
|
|
/*5*/
|
|
{{10663, 0}, {0, 10663}, {7540, 7540}, {9234, 5331}, {5331, 9234}},
|
|
/*6*/
|
|
{{8668, 0},
|
|
{0, 8668},
|
|
{6129, 6129},
|
|
{6129, 6129},
|
|
{7507, 4335},
|
|
{4335, 7507}},
|
|
/*7*/
|
|
{{7459, 0},
|
|
{0, 7459},
|
|
{5275, 5275},
|
|
{5275, 5275},
|
|
{4568, 4568},
|
|
{6460, 3731},
|
|
{3731, 6460}},
|
|
/*8*/
|
|
{{6368, 0},
|
|
{0, 6368},
|
|
{4502, 4502},
|
|
{4502, 4502},
|
|
{5514, 3184},
|
|
{3184, 5514},
|
|
{5514, 3184},
|
|
{3184, 5514}}};
|
|
// Re-write the buffer with downmixed data
|
|
const int16_t* in = static_cast<const int16_t*>(aIn);
|
|
int16_t* out = static_cast<int16_t*>(aOut);
|
|
for (uint32_t i = 0; i < aFrames; i++) {
|
|
int32_t sampL = 0;
|
|
int32_t sampR = 0;
|
|
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) {
|
|
*out++ = sampL;
|
|
*out++ = sampR;
|
|
} else {
|
|
*out++ = (sampL + sampR) * 0.5;
|
|
}
|
|
}
|
|
} else {
|
|
MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
|
|
}
|
|
return aFrames;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
size_t AudioConverter::ResampleAudio(void* aOut, const void* aIn,
|
|
size_t aFrames) {
|
|
if (!mResampler) {
|
|
return 0;
|
|
}
|
|
uint32_t outframes = ResampleRecipientFrames(aFrames);
|
|
uint32_t inframes = aFrames;
|
|
|
|
int error;
|
|
if (mOut.Format() == AudioConfig::FORMAT_FLT) {
|
|
const float* in = reinterpret_cast<const float*>(aIn);
|
|
float* out = reinterpret_cast<float*>(aOut);
|
|
error = speex_resampler_process_interleaved_float(mResampler, in, &inframes,
|
|
out, &outframes);
|
|
} else if (mOut.Format() == AudioConfig::FORMAT_S16) {
|
|
const int16_t* in = reinterpret_cast<const int16_t*>(aIn);
|
|
int16_t* out = reinterpret_cast<int16_t*>(aOut);
|
|
error = speex_resampler_process_interleaved_int(mResampler, in, &inframes,
|
|
out, &outframes);
|
|
} else {
|
|
MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
|
|
error = RESAMPLER_ERR_ALLOC_FAILED;
|
|
}
|
|
MOZ_ASSERT(error == RESAMPLER_ERR_SUCCESS);
|
|
if (error != RESAMPLER_ERR_SUCCESS) {
|
|
speex_resampler_destroy(mResampler);
|
|
mResampler = nullptr;
|
|
return 0;
|
|
}
|
|
MOZ_ASSERT(inframes == aFrames, "Some frames will be dropped");
|
|
return outframes;
|
|
}
|
|
|
|
void AudioConverter::RecreateResampler() {
|
|
if (mResampler) {
|
|
speex_resampler_destroy(mResampler);
|
|
}
|
|
int error;
|
|
mResampler = speex_resampler_init(mOut.Channels(), mIn.Rate(), mOut.Rate(),
|
|
SPEEX_RESAMPLER_QUALITY_DEFAULT, &error);
|
|
|
|
if (error == RESAMPLER_ERR_SUCCESS) {
|
|
speex_resampler_skip_zeros(mResampler);
|
|
} else {
|
|
NS_WARNING("Failed to initialize resampler.");
|
|
mResampler = nullptr;
|
|
}
|
|
}
|
|
|
|
size_t AudioConverter::DrainResampler(void* aOut) {
|
|
if (!mResampler) {
|
|
return 0;
|
|
}
|
|
int frames = speex_resampler_get_input_latency(mResampler);
|
|
AlignedByteBuffer buffer(FramesOutToBytes(frames));
|
|
if (!buffer) {
|
|
// OOM
|
|
return 0;
|
|
}
|
|
frames = ResampleAudio(aOut, buffer.Data(), frames);
|
|
// Tore down the resampler as it's easier than handling follow-up.
|
|
RecreateResampler();
|
|
return frames;
|
|
}
|
|
|
|
size_t AudioConverter::UpmixAudio(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.Channels() == 1, "Can only upmix mono for now");
|
|
MOZ_ASSERT(mOut.Channels() == 2, "Can only upmix to stereo for now");
|
|
|
|
if (!mIn.Layout().IsValid() || !mOut.Layout().IsValid() ||
|
|
mOut.Channels() != 2) {
|
|
// Dumb copy the channels and insert silence for the extra channels.
|
|
if (mIn.Format() == AudioConfig::FORMAT_FLT) {
|
|
dumbUpDownMix(static_cast<float*>(aOut), mOut.Channels(),
|
|
static_cast<const float*>(aIn), mIn.Channels(), aFrames);
|
|
} else if (mIn.Format() == AudioConfig::FORMAT_S16) {
|
|
dumbUpDownMix(static_cast<int16_t*>(aOut), mOut.Channels(),
|
|
static_cast<const int16_t*>(aIn), mIn.Channels(), aFrames);
|
|
} else {
|
|
MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
|
|
}
|
|
return aFrames;
|
|
}
|
|
|
|
// Upmix mono to stereo.
|
|
// This is a very dumb mono to stereo upmixing, power levels are preserved
|
|
// following the calculation: left = right = -3dB*mono.
|
|
if (mIn.Format() == AudioConfig::FORMAT_FLT) {
|
|
const float m3db = std::sqrt(0.5); // -3dB = sqrt(1/2)
|
|
const float* in = static_cast<const float*>(aIn);
|
|
float* out = static_cast<float*>(aOut);
|
|
for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
|
|
float sample = in[fIdx] * m3db;
|
|
// The samples of the buffer would be interleaved.
|
|
*out++ = sample;
|
|
*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) {
|
|
int16_t sample =
|
|
((int32_t)in[fIdx] * 11585) >> 14; // close enough to i*sqrt(0.5)
|
|
// The samples of the buffer would be interleaved.
|
|
*out++ = sample;
|
|
*out++ = sample;
|
|
}
|
|
} else {
|
|
MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
|
|
}
|
|
|
|
return aFrames;
|
|
}
|
|
|
|
size_t AudioConverter::ResampleRecipientFrames(size_t aFrames) const {
|
|
if (!aFrames && mIn.Rate() != mOut.Rate()) {
|
|
if (!mResampler) {
|
|
return 0;
|
|
}
|
|
// We drain by pushing in get_input_latency() samples of 0
|
|
aFrames = speex_resampler_get_input_latency(mResampler);
|
|
}
|
|
return (uint64_t)aFrames * mOut.Rate() / mIn.Rate() + 1;
|
|
}
|
|
|
|
size_t AudioConverter::FramesOutToSamples(size_t aFrames) const {
|
|
return aFrames * mOut.Channels();
|
|
}
|
|
|
|
size_t AudioConverter::SamplesInToFrames(size_t aSamples) const {
|
|
return aSamples / mIn.Channels();
|
|
}
|
|
|
|
size_t AudioConverter::FramesOutToBytes(size_t aFrames) const {
|
|
return FramesOutToSamples(aFrames) * AudioConfig::SampleSize(mOut.Format());
|
|
}
|
|
} // namespace mozilla
|