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/* -*- 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(CanConvert(aIn, aOut),
"The conversion is 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::CanConvert(const AudioConfig& aIn,
const AudioConfig& aOut) {
if (aIn.Format() != aOut.Format() ||
aIn.Interleaved() != aOut.Interleaved()) {
NS_WARNING("No format conversion is supported at this stage");
return false;
}
if (aIn.Channels() != aOut.Channels() && aOut.Channels() > 2) {
NS_WARNING(
"Only down/upmixing to mono or stereo is supported at this stage");
return false;
}
if (!aOut.Interleaved()) {
NS_WARNING("planar audio format not supported");
return false;
}
return true;
}
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