mirror of
https://github.com/roytam1/palemoon27.git
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c1485b21e1
- 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)
247 lines
8.6 KiB
C++
247 lines
8.6 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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/* vim:set ts=2 sw=2 sts=2 et cindent: */
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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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#ifndef MOZILLA_AUDIOCHANNELFORMAT_H_
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#define MOZILLA_AUDIOCHANNELFORMAT_H_
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#include <stdint.h>
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#include "nsTArrayForwardDeclare.h"
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#include "AudioSampleFormat.h"
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#include "nsTArray.h"
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namespace mozilla {
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/*
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* This file provides utilities for upmixing and downmixing channels.
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*
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* The channel layouts, upmixing and downmixing are consistent with the
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* Web Audio spec.
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*
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* Channel layouts for up to 6 channels:
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* mono { M }
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* stereo { L, R }
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* { L, R, C }
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* quad { L, R, SL, SR }
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* { L, R, C, SL, SR }
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* 5.1 { L, R, C, LFE, SL, SR }
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*
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* Only 1, 2, 4 and 6 are currently defined in Web Audio.
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*/
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enum {
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SURROUND_L,
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SURROUND_R,
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SURROUND_C,
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SURROUND_LFE,
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SURROUND_SL,
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SURROUND_SR
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};
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const uint32_t CUSTOM_CHANNEL_LAYOUTS = 6;
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// This is defined by some Windows SDK header.
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#undef IGNORE
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const int IGNORE = CUSTOM_CHANNEL_LAYOUTS;
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const float IGNORE_F = 0.0f;
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const int gMixingMatrixIndexByChannels[CUSTOM_CHANNEL_LAYOUTS - 1] =
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{ 0, 5, 9, 12, 14 };
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/**
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* Return a channel count whose channel layout includes all the channels from
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* aChannels1 and aChannels2.
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*/
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uint32_t
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GetAudioChannelsSuperset(uint32_t aChannels1, uint32_t aChannels2);
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/**
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* DownMixMatrix represents a conversion matrix efficiently by exploiting the
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* fact that each input channel contributes to at most one output channel,
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* except possibly for the C input channel in layouts that have one. Also,
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* every input channel is multiplied by the same coefficient for every output
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* channel it contributes to.
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*/
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struct DownMixMatrix {
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// Every input channel c is copied to output channel mInputDestination[c]
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// after multiplying by mInputCoefficient[c].
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uint8_t mInputDestination[CUSTOM_CHANNEL_LAYOUTS];
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// If not IGNORE, then the C channel is copied to this output channel after
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// multiplying by its coefficient.
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uint8_t mCExtraDestination;
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float mInputCoefficient[CUSTOM_CHANNEL_LAYOUTS];
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};
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static const DownMixMatrix
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gDownMixMatrices[CUSTOM_CHANNEL_LAYOUTS*(CUSTOM_CHANNEL_LAYOUTS - 1)/2] =
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{
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// Downmixes to mono
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{ { 0, 0 }, IGNORE, { 0.5f, 0.5f } },
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{ { 0, IGNORE, IGNORE }, IGNORE, { 1.0f, IGNORE_F, IGNORE_F } },
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{ { 0, 0, 0, 0 }, IGNORE, { 0.25f, 0.25f, 0.25f, 0.25f } },
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{ { 0, IGNORE, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, IGNORE_F, IGNORE_F, IGNORE_F, IGNORE_F } },
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{ { 0, 0, 0, IGNORE, 0, 0 }, IGNORE, { 0.7071f, 0.7071f, 1.0f, IGNORE_F, 0.5f, 0.5f } },
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// Downmixes to stereo
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{ { 0, 1, IGNORE }, IGNORE, { 1.0f, 1.0f, IGNORE_F } },
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{ { 0, 1, 0, 1 }, IGNORE, { 0.5f, 0.5f, 0.5f, 0.5f } },
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{ { 0, 1, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, IGNORE_F, IGNORE_F, IGNORE_F } },
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{ { 0, 1, 0, IGNORE, 0, 1 }, 1, { 1.0f, 1.0f, 0.7071f, IGNORE_F, 0.7071f, 0.7071f } },
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// Downmixes to 3-channel
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{ { 0, 1, 2, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F } },
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{ { 0, 1, 2, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F, IGNORE_F } },
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{ { 0, 1, 2, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F, IGNORE_F, IGNORE_F } },
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// Downmixes to quad
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{ { 0, 1, 2, 3, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, 1.0f, IGNORE_F } },
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{ { 0, 1, 0, IGNORE, 2, 3 }, 1, { 1.0f, 1.0f, 0.7071f, IGNORE_F, 1.0f, 1.0f } },
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// Downmixes to 5-channel
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{ { 0, 1, 2, 3, 4, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, IGNORE_F } }
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};
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/**
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* Given an array of input channels, downmix to aOutputChannelCount, and copy
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* the results to the channel buffers in aOutputChannels. Don't call this with
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* input count <= output count.
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*/
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template<typename T>
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void AudioChannelsDownMix(const nsTArray<const T*>& aChannelArray,
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T** aOutputChannels,
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uint32_t aOutputChannelCount,
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uint32_t aDuration)
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{
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uint32_t inputChannelCount = aChannelArray.Length();
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const T* const* inputChannels = aChannelArray.Elements();
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NS_ASSERTION(inputChannelCount > aOutputChannelCount, "Nothing to do");
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if (inputChannelCount > 6) {
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// Just drop the unknown channels.
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for (uint32_t o = 0; o < aOutputChannelCount; ++o) {
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PodCopy(aOutputChannels[o], inputChannels[o], aDuration);
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}
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return;
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}
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// Ignore unknown channels, they're just dropped.
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inputChannelCount = std::min<uint32_t>(6, inputChannelCount);
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const DownMixMatrix& m = gDownMixMatrices[
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gMixingMatrixIndexByChannels[aOutputChannelCount - 1] +
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inputChannelCount - aOutputChannelCount - 1];
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// This is slow, but general. We can define custom code for special
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// cases later.
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for (uint32_t s = 0; s < aDuration; ++s) {
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// Reserve an extra junk channel at the end for the cases where we
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// want an input channel to contribute to nothing
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T outputChannels[CUSTOM_CHANNEL_LAYOUTS + 1] = {0};
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for (uint32_t c = 0; c < inputChannelCount; ++c) {
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outputChannels[m.mInputDestination[c]] +=
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m.mInputCoefficient[c]*(static_cast<const T*>(inputChannels[c]))[s];
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}
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// Utilize the fact that in every layout, C is the third channel.
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if (m.mCExtraDestination != IGNORE) {
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outputChannels[m.mCExtraDestination] +=
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m.mInputCoefficient[SURROUND_C]*(static_cast<const T*>(inputChannels[SURROUND_C]))[s];
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}
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for (uint32_t c = 0; c < aOutputChannelCount; ++c) {
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aOutputChannels[c][s] = outputChannels[c];
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}
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}
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}
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/**
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* UpMixMatrix represents a conversion matrix by exploiting the fact that
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* each output channel comes from at most one input channel.
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*/
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struct UpMixMatrix {
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uint8_t mInputDestination[CUSTOM_CHANNEL_LAYOUTS];
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};
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static const UpMixMatrix
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gUpMixMatrices[CUSTOM_CHANNEL_LAYOUTS*(CUSTOM_CHANNEL_LAYOUTS - 1)/2] =
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{
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// Upmixes from mono
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{ { 0, 0 } },
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{ { 0, IGNORE, IGNORE } },
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{ { 0, 0, IGNORE, IGNORE } },
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{ { 0, IGNORE, IGNORE, IGNORE, IGNORE } },
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{ { IGNORE, IGNORE, 0, IGNORE, IGNORE, IGNORE } },
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// Upmixes from stereo
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{ { 0, 1, IGNORE } },
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{ { 0, 1, IGNORE, IGNORE } },
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{ { 0, 1, IGNORE, IGNORE, IGNORE } },
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{ { 0, 1, IGNORE, IGNORE, IGNORE, IGNORE } },
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// Upmixes from 3-channel
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{ { 0, 1, 2, IGNORE } },
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{ { 0, 1, 2, IGNORE, IGNORE } },
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{ { 0, 1, 2, IGNORE, IGNORE, IGNORE } },
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// Upmixes from quad
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{ { 0, 1, 2, 3, IGNORE } },
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{ { 0, 1, IGNORE, IGNORE, 2, 3 } },
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// Upmixes from 5-channel
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{ { 0, 1, 2, 3, 4, IGNORE } }
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};
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/**
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* Given an array of input channel data, and an output channel count,
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* replaces the array with an array of upmixed channels.
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* This shuffles the array and may set some channel buffers to aZeroChannel.
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* Don't call this with input count >= output count.
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* This may return *more* channels than requested. In that case, downmixing
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* is required to to get to aOutputChannelCount. (This is how we handle
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* odd cases like 3 -> 4 upmixing.)
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* If aChannelArray.Length() was the input to one of a series of
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* GetAudioChannelsSuperset calls resulting in aOutputChannelCount,
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* no downmixing will be required.
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*/
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template<typename T>
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void
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AudioChannelsUpMix(nsTArray<const T*>* aChannelArray,
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uint32_t aOutputChannelCount,
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const T* aZeroChannel)
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{
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uint32_t inputChannelCount = aChannelArray->Length();
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uint32_t outputChannelCount =
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GetAudioChannelsSuperset(aOutputChannelCount, inputChannelCount);
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NS_ASSERTION(outputChannelCount > inputChannelCount,
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"No up-mix needed");
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MOZ_ASSERT(inputChannelCount > 0, "Bad number of channels");
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MOZ_ASSERT(outputChannelCount > 0, "Bad number of channels");
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aChannelArray->SetLength(outputChannelCount);
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if (inputChannelCount < CUSTOM_CHANNEL_LAYOUTS &&
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outputChannelCount <= CUSTOM_CHANNEL_LAYOUTS) {
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const UpMixMatrix& m = gUpMixMatrices[
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gMixingMatrixIndexByChannels[inputChannelCount - 1] +
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outputChannelCount - inputChannelCount - 1];
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const T* outputChannels[CUSTOM_CHANNEL_LAYOUTS];
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for (uint32_t i = 0; i < outputChannelCount; ++i) {
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uint8_t channelIndex = m.mInputDestination[i];
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if (channelIndex == IGNORE) {
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outputChannels[i] = aZeroChannel;
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} else {
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outputChannels[i] = aChannelArray->ElementAt(channelIndex);
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}
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}
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for (uint32_t i = 0; i < outputChannelCount; ++i) {
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aChannelArray->ElementAt(i) = outputChannels[i];
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}
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return;
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}
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for (uint32_t i = inputChannelCount; i < outputChannelCount; ++i) {
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aChannelArray->ElementAt(i) = aZeroChannel;
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}
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}
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} // namespace mozilla
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#endif /* MOZILLA_AUDIOCHANNELFORMAT_H_ */
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