Files
palemoon27/dom/media/AudioChannelFormat.h
T
roytam1 c1485b21e1 import changes from `dev' branch of rmottola/Arctic-Fox:
- bug 1179662 rename AudioNode::Stream() to GetStream() as it may return null r=padenot (d2d7e5f90)
-  bug 1197043 remove unnecessary aSampleRate parameter for AudioNodeStream creation r=padenot (3733ceb56)
- bug 1197043 rename Add/RemoveStream to Add/RemoveStreamGraphThread r=padenot (f648b8251)
- bug 1197043 introduce MediaStreamGraph::AddStream() r=padenot (ac021d4b2)
- bug 1197043 move AudioNodeStream creation to stream class r=padenot (a90a05910)
- bug 1197043 use flags to distinguish between external streams and events r=padenot (024dd96f1)
- Bug 1176300 - Move libsoundtouch to lgpllibs; r=glandium (99a546adf)
- Bug 1176300 - Add lgpllibs library to build system; r=glandium (bb4d07670)
- Bug 1176300 - Update libsoundtouch to patched r222; r=padenot (hg rev 8c32c900cb48)
- Bug 1176300 - Add soundtouch factory functions for DLL memory handling on windows; r=padenot (hg rev 84a1ffbc2db8)
- Bug 901633 - Part 1 - Implement a generic audio packetizer. r=jesup (a38c2d70b)
- Bug 901633 - Part 2 - Make AudioChannelFormat and AudioSegment more generic. r=roc (556b7349f)
- Bug 901633 - Part 3 - Fix TrackEncoder to use the new AudioChunk methods. r=jesup (56e018f83)
- Bug 901633 - Part 4 - Update AudioNodeStream to use new chunk methods. r=roc (9df19b894)
- Bug 901633 - Part 6 - Update DelayBuffer to use the new AudioChunk methods. r=karlt (6d5684334)
- Bug 901633 - Part 7 - Update AudioNodeExternalInputStream to use the new AudioChunk methods. r=karlt (caa3afa01)
- Bug 1155089: Fix hazard analysis bustage on a CLOSED TREE. r=bustage (8d23ccf39)
- Bug 1166183 - Back out the direct listener removal landed by mistake in bug 1141781. r=jesup (745d683d4)
- Bug 1166183 - Reset PipelineListener's flag after ReplaceTrack(). r=bwc (2fb38ca01)
- Bug 1170059 - Fix -Wunreachable-code clang warnings in webrtc/signaling. r=jesup (0d99b30ca)
- Bug 1139144 - Remove unused empty() definition from databuffer.h. r=mt (2fef64e3c)
- Bug 822129: don't alloc/free on every packet send in MediaPipeline r=bwc (ddfdc9455)
- Bug 1172397 - Check for Conduit/Type mismatch on every frame. r=jesup, r=bwc (a399a3336)
- Bug 1137169 - Uninitialised value uses related to mozilla::dom::WebAdioUtils::SpeexResamplerProcess. r=rjesup. (ce14ac278)
- Bug 901633 - Part 5 - Make MediaPipeline downmix and properly convert audio for webrtc.org code. r=jesup (89138b5d5)
- Bug 901633 - Part 8 - Use our new generic packetizer in the MediaPipeline so that we can packetize stereo easily. r=jesup (fb5d075b6)
- Bug 901633 - Part 9 - Make the necessary changes to VoEExternalMediaImpl::ExternalRecordingInsertData so that it the number of channels is forwarded down the webrtc.org code. r=jesup (d5d7dd4ca)
- Bug 901633 - Part 10 - Change the receiving side of the MediaPipeline so that it can detect and handle stereo. r=jesup (a73c1520f)
- Bug 901633 - Part 11 - Add an API in webrtc.org's output mixer to get the output channel count. r=jesup (4b396b85e)
- Bug 901633 - Part 12 - Add a function to deinterleave and convert an audio buffer. r=jesup (47ce3c7a5)
- Bug 901633 - Part 13 - Teach the resampler at the input of the MSG to dynamically change its channel count if needed. r=jesup (120c8d037)
- Bug 901633 - Part 14 - Add testing for our audio processing functions. r=jesup (5aa95b82e)
- Bug 901633 - Part 15 - Remove an allocation on the sending side, out of the packetizer. r=jesup (df8aed252)
- Bug 901633 - Part 16 - Remove another allocation in the sending side r=jesup (1e2fc8bca)
- Bug 1196408 - Make sure we only report a corrupt/slow video frame once. r=cpearce (5bae2f17a)
- bug 1162364 report telemetry on WMFMediaDataDecoder errors r=cpearce,f=vladan,bsmedberg (e217618ef)
- Bug 1193864 - Fixed dom/media/platforms/wmf/ compilation on mingw. r=cpearce (4e8c0ecd7)
- Bug 1141139: Enable low latency decoding on Windows. r=cpearce (9e0a36e27)
- Bug 1193547 - Fallback to software decoding explicitly if the GPU doesn't support decoding the current resolution in hardware. r=cpearce,jya (7fbab8784)
- Bug 1196417 - Make video software fallback only affect the current video instead of the entire browser. r=cpearce (3e83f0677)
2021-10-12 10:01:23 +08:00

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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim:set ts=2 sw=2 sts=2 et cindent: */
/* 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/. */
#ifndef MOZILLA_AUDIOCHANNELFORMAT_H_
#define MOZILLA_AUDIOCHANNELFORMAT_H_
#include <stdint.h>
#include "nsTArrayForwardDeclare.h"
#include "AudioSampleFormat.h"
#include "nsTArray.h"
namespace mozilla {
/*
* This file provides utilities for upmixing and downmixing channels.
*
* The channel layouts, upmixing and downmixing are consistent with the
* Web Audio spec.
*
* Channel layouts for up to 6 channels:
* mono { M }
* stereo { L, R }
* { L, R, C }
* quad { L, R, SL, SR }
* { L, R, C, SL, SR }
* 5.1 { L, R, C, LFE, SL, SR }
*
* Only 1, 2, 4 and 6 are currently defined in Web Audio.
*/
enum {
SURROUND_L,
SURROUND_R,
SURROUND_C,
SURROUND_LFE,
SURROUND_SL,
SURROUND_SR
};
const uint32_t CUSTOM_CHANNEL_LAYOUTS = 6;
// This is defined by some Windows SDK header.
#undef IGNORE
const int IGNORE = CUSTOM_CHANNEL_LAYOUTS;
const float IGNORE_F = 0.0f;
const int gMixingMatrixIndexByChannels[CUSTOM_CHANNEL_LAYOUTS - 1] =
{ 0, 5, 9, 12, 14 };
/**
* Return a channel count whose channel layout includes all the channels from
* aChannels1 and aChannels2.
*/
uint32_t
GetAudioChannelsSuperset(uint32_t aChannels1, uint32_t aChannels2);
/**
* DownMixMatrix represents a conversion matrix efficiently by exploiting the
* fact that each input channel contributes to at most one output channel,
* except possibly for the C input channel in layouts that have one. Also,
* every input channel is multiplied by the same coefficient for every output
* channel it contributes to.
*/
struct DownMixMatrix {
// Every input channel c is copied to output channel mInputDestination[c]
// after multiplying by mInputCoefficient[c].
uint8_t mInputDestination[CUSTOM_CHANNEL_LAYOUTS];
// If not IGNORE, then the C channel is copied to this output channel after
// multiplying by its coefficient.
uint8_t mCExtraDestination;
float mInputCoefficient[CUSTOM_CHANNEL_LAYOUTS];
};
static const DownMixMatrix
gDownMixMatrices[CUSTOM_CHANNEL_LAYOUTS*(CUSTOM_CHANNEL_LAYOUTS - 1)/2] =
{
// Downmixes to mono
{ { 0, 0 }, IGNORE, { 0.5f, 0.5f } },
{ { 0, IGNORE, IGNORE }, IGNORE, { 1.0f, IGNORE_F, IGNORE_F } },
{ { 0, 0, 0, 0 }, IGNORE, { 0.25f, 0.25f, 0.25f, 0.25f } },
{ { 0, IGNORE, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, IGNORE_F, IGNORE_F, IGNORE_F, IGNORE_F } },
{ { 0, 0, 0, IGNORE, 0, 0 }, IGNORE, { 0.7071f, 0.7071f, 1.0f, IGNORE_F, 0.5f, 0.5f } },
// Downmixes to stereo
{ { 0, 1, IGNORE }, IGNORE, { 1.0f, 1.0f, IGNORE_F } },
{ { 0, 1, 0, 1 }, IGNORE, { 0.5f, 0.5f, 0.5f, 0.5f } },
{ { 0, 1, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, IGNORE_F, IGNORE_F, IGNORE_F } },
{ { 0, 1, 0, IGNORE, 0, 1 }, 1, { 1.0f, 1.0f, 0.7071f, IGNORE_F, 0.7071f, 0.7071f } },
// Downmixes to 3-channel
{ { 0, 1, 2, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F } },
{ { 0, 1, 2, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F, IGNORE_F } },
{ { 0, 1, 2, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F, IGNORE_F, IGNORE_F } },
// Downmixes to quad
{ { 0, 1, 2, 3, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, 1.0f, IGNORE_F } },
{ { 0, 1, 0, IGNORE, 2, 3 }, 1, { 1.0f, 1.0f, 0.7071f, IGNORE_F, 1.0f, 1.0f } },
// Downmixes to 5-channel
{ { 0, 1, 2, 3, 4, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, IGNORE_F } }
};
/**
* Given an array of input channels, downmix to aOutputChannelCount, and copy
* the results to the channel buffers in aOutputChannels. Don't call this with
* input count <= output count.
*/
template<typename T>
void AudioChannelsDownMix(const nsTArray<const T*>& aChannelArray,
T** aOutputChannels,
uint32_t aOutputChannelCount,
uint32_t aDuration)
{
uint32_t inputChannelCount = aChannelArray.Length();
const T* const* inputChannels = aChannelArray.Elements();
NS_ASSERTION(inputChannelCount > aOutputChannelCount, "Nothing to do");
if (inputChannelCount > 6) {
// Just drop the unknown channels.
for (uint32_t o = 0; o < aOutputChannelCount; ++o) {
PodCopy(aOutputChannels[o], inputChannels[o], aDuration);
}
return;
}
// Ignore unknown channels, they're just dropped.
inputChannelCount = std::min<uint32_t>(6, inputChannelCount);
const DownMixMatrix& m = gDownMixMatrices[
gMixingMatrixIndexByChannels[aOutputChannelCount - 1] +
inputChannelCount - aOutputChannelCount - 1];
// This is slow, but general. We can define custom code for special
// cases later.
for (uint32_t s = 0; s < aDuration; ++s) {
// Reserve an extra junk channel at the end for the cases where we
// want an input channel to contribute to nothing
T outputChannels[CUSTOM_CHANNEL_LAYOUTS + 1] = {0};
for (uint32_t c = 0; c < inputChannelCount; ++c) {
outputChannels[m.mInputDestination[c]] +=
m.mInputCoefficient[c]*(static_cast<const T*>(inputChannels[c]))[s];
}
// Utilize the fact that in every layout, C is the third channel.
if (m.mCExtraDestination != IGNORE) {
outputChannels[m.mCExtraDestination] +=
m.mInputCoefficient[SURROUND_C]*(static_cast<const T*>(inputChannels[SURROUND_C]))[s];
}
for (uint32_t c = 0; c < aOutputChannelCount; ++c) {
aOutputChannels[c][s] = outputChannels[c];
}
}
}
/**
* UpMixMatrix represents a conversion matrix by exploiting the fact that
* each output channel comes from at most one input channel.
*/
struct UpMixMatrix {
uint8_t mInputDestination[CUSTOM_CHANNEL_LAYOUTS];
};
static const UpMixMatrix
gUpMixMatrices[CUSTOM_CHANNEL_LAYOUTS*(CUSTOM_CHANNEL_LAYOUTS - 1)/2] =
{
// Upmixes from mono
{ { 0, 0 } },
{ { 0, IGNORE, IGNORE } },
{ { 0, 0, IGNORE, IGNORE } },
{ { 0, IGNORE, IGNORE, IGNORE, IGNORE } },
{ { IGNORE, IGNORE, 0, IGNORE, IGNORE, IGNORE } },
// Upmixes from stereo
{ { 0, 1, IGNORE } },
{ { 0, 1, IGNORE, IGNORE } },
{ { 0, 1, IGNORE, IGNORE, IGNORE } },
{ { 0, 1, IGNORE, IGNORE, IGNORE, IGNORE } },
// Upmixes from 3-channel
{ { 0, 1, 2, IGNORE } },
{ { 0, 1, 2, IGNORE, IGNORE } },
{ { 0, 1, 2, IGNORE, IGNORE, IGNORE } },
// Upmixes from quad
{ { 0, 1, 2, 3, IGNORE } },
{ { 0, 1, IGNORE, IGNORE, 2, 3 } },
// Upmixes from 5-channel
{ { 0, 1, 2, 3, 4, IGNORE } }
};
/**
* Given an array of input channel data, and an output channel count,
* replaces the array with an array of upmixed channels.
* This shuffles the array and may set some channel buffers to aZeroChannel.
* Don't call this with input count >= output count.
* This may return *more* channels than requested. In that case, downmixing
* is required to to get to aOutputChannelCount. (This is how we handle
* odd cases like 3 -> 4 upmixing.)
* If aChannelArray.Length() was the input to one of a series of
* GetAudioChannelsSuperset calls resulting in aOutputChannelCount,
* no downmixing will be required.
*/
template<typename T>
void
AudioChannelsUpMix(nsTArray<const T*>* aChannelArray,
uint32_t aOutputChannelCount,
const T* aZeroChannel)
{
uint32_t inputChannelCount = aChannelArray->Length();
uint32_t outputChannelCount =
GetAudioChannelsSuperset(aOutputChannelCount, inputChannelCount);
NS_ASSERTION(outputChannelCount > inputChannelCount,
"No up-mix needed");
MOZ_ASSERT(inputChannelCount > 0, "Bad number of channels");
MOZ_ASSERT(outputChannelCount > 0, "Bad number of channels");
aChannelArray->SetLength(outputChannelCount);
if (inputChannelCount < CUSTOM_CHANNEL_LAYOUTS &&
outputChannelCount <= CUSTOM_CHANNEL_LAYOUTS) {
const UpMixMatrix& m = gUpMixMatrices[
gMixingMatrixIndexByChannels[inputChannelCount - 1] +
outputChannelCount - inputChannelCount - 1];
const T* outputChannels[CUSTOM_CHANNEL_LAYOUTS];
for (uint32_t i = 0; i < outputChannelCount; ++i) {
uint8_t channelIndex = m.mInputDestination[i];
if (channelIndex == IGNORE) {
outputChannels[i] = aZeroChannel;
} else {
outputChannels[i] = aChannelArray->ElementAt(channelIndex);
}
}
for (uint32_t i = 0; i < outputChannelCount; ++i) {
aChannelArray->ElementAt(i) = outputChannels[i];
}
return;
}
for (uint32_t i = inputChannelCount; i < outputChannelCount; ++i) {
aChannelArray->ElementAt(i) = aZeroChannel;
}
}
} // namespace mozilla
#endif /* MOZILLA_AUDIOCHANNELFORMAT_H_ */