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import FIREFOX_52_6_0esr_RELEASE from mozilla-esr52 hg repo
This commit is contained in:
Roy Tam
commit
dcd9973243
150858 changed files with 23884658 additions and 0 deletions
393
dom/media/AudioConverter.cpp
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393
dom/media/AudioConverter.cpp
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/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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/* vim: set ts=8 sts=2 et sw=2 tw=80: */
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this
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* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
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#include "AudioConverter.h"
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#include <string.h>
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#include <speex/speex_resampler.h>
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#include <cmath>
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/*
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* Parts derived from MythTV AudioConvert Class
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* Created by Jean-Yves Avenard.
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*
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* Copyright (C) Bubblestuff Pty Ltd 2013
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* Copyright (C) foobum@gmail.com 2010
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*/
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namespace mozilla {
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AudioConverter::AudioConverter(const AudioConfig& aIn, const AudioConfig& aOut)
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: mIn(aIn)
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, mOut(aOut)
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, mResampler(nullptr)
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{
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MOZ_DIAGNOSTIC_ASSERT(aIn.Format() == aOut.Format() &&
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aIn.Interleaved() == aOut.Interleaved(),
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"No format or rate conversion is supported at this stage");
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MOZ_DIAGNOSTIC_ASSERT(aOut.Channels() <= 2 ||
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aIn.Channels() == aOut.Channels(),
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"Only down/upmixing to mono or stereo is supported at this stage");
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MOZ_DIAGNOSTIC_ASSERT(aOut.Interleaved(), "planar audio format not supported");
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mIn.Layout().MappingTable(mOut.Layout(), mChannelOrderMap);
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if (aIn.Rate() != aOut.Rate()) {
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RecreateResampler();
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}
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}
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AudioConverter::~AudioConverter()
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{
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if (mResampler) {
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speex_resampler_destroy(mResampler);
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mResampler = nullptr;
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}
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}
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bool
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AudioConverter::CanWorkInPlace() const
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{
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bool needDownmix = mIn.Channels() > mOut.Channels();
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bool needUpmix = mIn.Channels() < mOut.Channels();
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bool canDownmixInPlace =
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mIn.Channels() * AudioConfig::SampleSize(mIn.Format()) >=
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mOut.Channels() * AudioConfig::SampleSize(mOut.Format());
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bool needResample = mIn.Rate() != mOut.Rate();
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bool canResampleInPlace = mIn.Rate() >= mOut.Rate();
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// We should be able to work in place if 1s of audio input takes less space
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// than 1s of audio output. However, as we downmix before resampling we can't
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// perform any upsampling in place (e.g. if incoming rate >= outgoing rate)
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return !needUpmix && (!needDownmix || canDownmixInPlace) &&
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(!needResample || canResampleInPlace);
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}
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size_t
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AudioConverter::ProcessInternal(void* aOut, const void* aIn, size_t aFrames)
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{
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if (mIn.Channels() > mOut.Channels()) {
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return DownmixAudio(aOut, aIn, aFrames);
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} else if (mIn.Channels() < mOut.Channels()) {
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return UpmixAudio(aOut, aIn, aFrames);
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} else if (mIn.Layout() != mOut.Layout() && CanReorderAudio()) {
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ReOrderInterleavedChannels(aOut, aIn, aFrames);
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} else if (aIn != aOut) {
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memmove(aOut, aIn, FramesOutToBytes(aFrames));
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}
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return aFrames;
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}
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// Reorder interleaved channels.
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// Can work in place (e.g aOut == aIn).
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template <class AudioDataType>
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void
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_ReOrderInterleavedChannels(AudioDataType* aOut, const AudioDataType* aIn,
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uint32_t aFrames, uint32_t aChannels,
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const uint8_t* aChannelOrderMap)
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{
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MOZ_DIAGNOSTIC_ASSERT(aChannels <= MAX_AUDIO_CHANNELS);
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AudioDataType val[MAX_AUDIO_CHANNELS];
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for (uint32_t i = 0; i < aFrames; i++) {
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for (uint32_t j = 0; j < aChannels; j++) {
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val[j] = aIn[aChannelOrderMap[j]];
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}
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for (uint32_t j = 0; j < aChannels; j++) {
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aOut[j] = val[j];
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}
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aOut += aChannels;
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aIn += aChannels;
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}
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}
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void
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AudioConverter::ReOrderInterleavedChannels(void* aOut, const void* aIn,
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size_t aFrames) const
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{
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MOZ_DIAGNOSTIC_ASSERT(mIn.Channels() == mOut.Channels());
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if (mOut.Channels() == 1 || mOut.Layout() == mIn.Layout()) {
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// If channel count is 1, planar and non-planar formats are the same and
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// there's nothing to reorder.
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if (aOut != aIn) {
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memmove(aOut, aIn, FramesOutToBytes(aFrames));
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}
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return;
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}
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uint32_t bits = AudioConfig::FormatToBits(mOut.Format());
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switch (bits) {
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case 8:
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_ReOrderInterleavedChannels((uint8_t*)aOut, (const uint8_t*)aIn,
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aFrames, mIn.Channels(), mChannelOrderMap);
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break;
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case 16:
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_ReOrderInterleavedChannels((int16_t*)aOut,(const int16_t*)aIn,
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aFrames, mIn.Channels(), mChannelOrderMap);
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break;
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default:
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MOZ_DIAGNOSTIC_ASSERT(AudioConfig::SampleSize(mOut.Format()) == 4);
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_ReOrderInterleavedChannels((int32_t*)aOut,(const int32_t*)aIn,
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aFrames, mIn.Channels(), mChannelOrderMap);
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break;
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}
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}
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static inline int16_t clipTo15(int32_t aX)
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{
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return aX < -32768 ? -32768 : aX <= 32767 ? aX : 32767;
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}
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size_t
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AudioConverter::DownmixAudio(void* aOut, const void* aIn, size_t aFrames) const
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{
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MOZ_ASSERT(mIn.Format() == AudioConfig::FORMAT_S16 ||
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mIn.Format() == AudioConfig::FORMAT_FLT);
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MOZ_ASSERT(mIn.Channels() >= mOut.Channels());
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MOZ_ASSERT(mIn.Layout() == AudioConfig::ChannelLayout(mIn.Channels()),
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"Can only downmix input data in SMPTE layout");
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MOZ_ASSERT(mOut.Layout() == AudioConfig::ChannelLayout(2) ||
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mOut.Layout() == AudioConfig::ChannelLayout(1));
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uint32_t channels = mIn.Channels();
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if (channels == 1 && mOut.Channels() == 1) {
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if (aOut != aIn) {
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memmove(aOut, aIn, FramesOutToBytes(aFrames));
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}
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return aFrames;
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}
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if (channels > 2) {
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if (mIn.Format() == AudioConfig::FORMAT_FLT) {
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// Downmix matrix. Per-row normalization 1 for rows 3,4 and 2 for rows 5-8.
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static const float dmatrix[6][8][2]= {
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/*3*/{{0.5858f,0},{0,0.5858f},{0.4142f,0.4142f}},
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/*4*/{{0.4226f,0},{0,0.4226f},{0.366f, 0.2114f},{0.2114f,0.366f}},
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/*5*/{{0.6510f,0},{0,0.6510f},{0.4600f,0.4600f},{0.5636f,0.3254f},{0.3254f,0.5636f}},
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/*6*/{{0.5290f,0},{0,0.5290f},{0.3741f,0.3741f},{0.3741f,0.3741f},{0.4582f,0.2645f},{0.2645f,0.4582f}},
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/*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}},
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/*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}},
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};
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// Re-write the buffer with downmixed data
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const float* in = static_cast<const float*>(aIn);
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float* out = static_cast<float*>(aOut);
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for (uint32_t i = 0; i < aFrames; i++) {
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float sampL = 0.0;
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float sampR = 0.0;
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for (uint32_t j = 0; j < channels; j++) {
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sampL += in[i*mIn.Channels()+j]*dmatrix[mIn.Channels()-3][j][0];
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sampR += in[i*mIn.Channels()+j]*dmatrix[mIn.Channels()-3][j][1];
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}
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*out++ = sampL;
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*out++ = sampR;
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}
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} else if (mIn.Format() == AudioConfig::FORMAT_S16) {
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// Downmix matrix. Per-row normalization 1 for rows 3,4 and 2 for rows 5-8.
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// Coefficients in Q14.
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static const int16_t dmatrix[6][8][2]= {
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/*3*/{{9598, 0},{0, 9598},{6786,6786}},
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/*4*/{{6925, 0},{0, 6925},{5997,3462},{3462,5997}},
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/*5*/{{10663,0},{0, 10663},{7540,7540},{9234,5331},{5331,9234}},
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/*6*/{{8668, 0},{0, 8668},{6129,6129},{6129,6129},{7507,4335},{4335,7507}},
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/*7*/{{7459, 0},{0, 7459},{5275,5275},{5275,5275},{4568,4568},{6460,3731},{3731,6460}},
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/*8*/{{6368, 0},{0, 6368},{4502,4502},{4502,4502},{5514,3184},{3184,5514},{5514,3184},{3184,5514}}
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};
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// Re-write the buffer with downmixed data
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const int16_t* in = static_cast<const int16_t*>(aIn);
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int16_t* out = static_cast<int16_t*>(aOut);
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for (uint32_t i = 0; i < aFrames; i++) {
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int32_t sampL = 0;
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int32_t sampR = 0;
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for (uint32_t j = 0; j < channels; j++) {
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sampL+=in[i*channels+j]*dmatrix[channels-3][j][0];
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sampR+=in[i*channels+j]*dmatrix[channels-3][j][1];
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}
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*out++ = clipTo15((sampL + 8192)>>14);
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*out++ = clipTo15((sampR + 8192)>>14);
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}
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} else {
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MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
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}
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// If we are to continue downmixing to mono, start working on the output
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// buffer.
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aIn = aOut;
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channels = 2;
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}
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if (mOut.Channels() == 1) {
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if (mIn.Format() == AudioConfig::FORMAT_FLT) {
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const float* in = static_cast<const float*>(aIn);
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float* out = static_cast<float*>(aOut);
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for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
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float sample = 0.0;
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// The sample of the buffer would be interleaved.
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sample = (in[fIdx*channels] + in[fIdx*channels + 1]) * 0.5;
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*out++ = sample;
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}
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} else if (mIn.Format() == AudioConfig::FORMAT_S16) {
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const int16_t* in = static_cast<const int16_t*>(aIn);
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int16_t* out = static_cast<int16_t*>(aOut);
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for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
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int32_t sample = 0.0;
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// The sample of the buffer would be interleaved.
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sample = (in[fIdx*channels] + in[fIdx*channels + 1]) * 0.5;
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*out++ = sample;
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}
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} else {
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MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
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}
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}
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return aFrames;
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}
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size_t
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AudioConverter::ResampleAudio(void* aOut, const void* aIn, size_t aFrames)
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{
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if (!mResampler) {
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return 0;
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}
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uint32_t outframes = ResampleRecipientFrames(aFrames);
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uint32_t inframes = aFrames;
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int error;
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if (mOut.Format() == AudioConfig::FORMAT_FLT) {
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const float* in = reinterpret_cast<const float*>(aIn);
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float* out = reinterpret_cast<float*>(aOut);
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error =
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speex_resampler_process_interleaved_float(mResampler, in, &inframes,
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out, &outframes);
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} else if (mOut.Format() == AudioConfig::FORMAT_S16) {
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const int16_t* in = reinterpret_cast<const int16_t*>(aIn);
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int16_t* out = reinterpret_cast<int16_t*>(aOut);
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error =
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speex_resampler_process_interleaved_int(mResampler, in, &inframes,
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out, &outframes);
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} else {
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MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
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error = RESAMPLER_ERR_ALLOC_FAILED;
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}
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MOZ_ASSERT(error == RESAMPLER_ERR_SUCCESS);
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if (error != RESAMPLER_ERR_SUCCESS) {
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speex_resampler_destroy(mResampler);
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mResampler = nullptr;
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return 0;
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}
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MOZ_ASSERT(inframes == aFrames, "Some frames will be dropped");
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return outframes;
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}
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void
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AudioConverter::RecreateResampler()
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{
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if (mResampler) {
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speex_resampler_destroy(mResampler);
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}
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int error;
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mResampler = speex_resampler_init(mOut.Channels(),
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mIn.Rate(),
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mOut.Rate(),
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SPEEX_RESAMPLER_QUALITY_DEFAULT,
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&error);
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if (error == RESAMPLER_ERR_SUCCESS) {
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speex_resampler_skip_zeros(mResampler);
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} else {
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NS_WARNING("Failed to initialize resampler.");
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mResampler = nullptr;
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}
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}
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size_t
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AudioConverter::DrainResampler(void* aOut)
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{
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if (!mResampler) {
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return 0;
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}
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int frames = speex_resampler_get_input_latency(mResampler);
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AlignedByteBuffer buffer(FramesOutToBytes(frames));
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if (!buffer) {
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// OOM
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return 0;
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}
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frames = ResampleAudio(aOut, buffer.Data(), frames);
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// Tore down the resampler as it's easier than handling follow-up.
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RecreateResampler();
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return frames;
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}
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size_t
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AudioConverter::UpmixAudio(void* aOut, const void* aIn, size_t aFrames) const
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{
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MOZ_ASSERT(mIn.Format() == AudioConfig::FORMAT_S16 ||
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mIn.Format() == AudioConfig::FORMAT_FLT);
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MOZ_ASSERT(mIn.Channels() < mOut.Channels());
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MOZ_ASSERT(mIn.Channels() == 1, "Can only upmix mono for now");
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MOZ_ASSERT(mOut.Channels() == 2, "Can only upmix to stereo for now");
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if (mOut.Channels() != 2) {
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return 0;
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}
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// Upmix mono to stereo.
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// This is a very dumb mono to stereo upmixing, power levels are preserved
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// following the calculation: left = right = -3dB*mono.
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if (mIn.Format() == AudioConfig::FORMAT_FLT) {
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const float m3db = std::sqrt(0.5); // -3dB = sqrt(1/2)
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const float* in = static_cast<const float*>(aIn);
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float* out = static_cast<float*>(aOut);
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for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
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float sample = in[fIdx] * m3db;
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// The samples of the buffer would be interleaved.
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*out++ = sample;
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*out++ = sample;
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}
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} else if (mIn.Format() == AudioConfig::FORMAT_S16) {
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const int16_t* in = static_cast<const int16_t*>(aIn);
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int16_t* out = static_cast<int16_t*>(aOut);
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for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
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int16_t sample = ((int32_t)in[fIdx] * 11585) >> 14; // close enough to i*sqrt(0.5)
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// The samples of the buffer would be interleaved.
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*out++ = sample;
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*out++ = sample;
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}
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} else {
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MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
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}
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return aFrames;
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}
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size_t
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AudioConverter::ResampleRecipientFrames(size_t aFrames) const
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{
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if (!aFrames && mIn.Rate() != mOut.Rate()) {
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// The resampler will be drained, account for frames currently buffered
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// in the resampler.
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if (!mResampler) {
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return 0;
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}
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return speex_resampler_get_output_latency(mResampler);
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} else {
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return (uint64_t)aFrames * mOut.Rate() / mIn.Rate() + 1;
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}
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}
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size_t
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AudioConverter::FramesOutToSamples(size_t aFrames) const
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{
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return aFrames * mOut.Channels();
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}
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size_t
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AudioConverter::SamplesInToFrames(size_t aSamples) const
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{
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return aSamples / mIn.Channels();
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}
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size_t
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AudioConverter::FramesOutToBytes(size_t aFrames) const
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{
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return FramesOutToSamples(aFrames) * AudioConfig::SampleSize(mOut.Format());
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}
|
||||
} // namespace mozilla
|
||||
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