зеркало из https://github.com/mozilla/gecko-dev.git
Bug 1428392 - Remove AudioOutputObserver, and feed the reverse-stream directly to the AEC. r=pehrsons
MozReview-Commit-ID: EVpH5LUOZ6g --HG-- extra : rebase_source : d172cd10dffee89ac48b9ec6b6435dcf0c0581d4
This commit is contained in:
Paul Adenot
Родитель
0b9f42aad6
Коммит
e3c5088847
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@ -30,9 +30,6 @@
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#include "VideoFrameContainer.h"
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#include "mozilla/AbstractThread.h"
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#include "mozilla/Unused.h"
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#ifdef MOZ_WEBRTC
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#include "AudioOutputObserver.h"
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#endif
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#include "mtransport/runnable_utils.h"
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#include "VideoUtils.h"
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@ -3569,9 +3566,6 @@ MediaStreamGraphImpl::MediaStreamGraphImpl(GraphDriverType aDriverRequested,
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, mLatencyLog(AsyncLatencyLogger::Get())
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, mAbstractMainThread(aMainThread)
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, mThreadPool(GetMediaThreadPool(MediaThreadType::MSG_CONTROL))
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#ifdef MOZ_WEBRTC
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, mFarendObserverRef(nullptr)
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#endif
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, mSelfRef(this)
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, mOutputChannels(std::min<uint32_t>(8, CubebUtils::MaxNumberOfChannels()))
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#ifdef DEBUG
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@ -32,9 +32,6 @@ class ShutdownTicket;
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template <typename T>
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class LinkedList;
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#ifdef MOZ_WEBRTC
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class AudioOutputObserver;
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#endif
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/**
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* A per-stream update message passed from the media graph thread to the
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@ -818,9 +815,6 @@ public:
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AudioMixer mMixer;
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const RefPtr<AbstractThread> mAbstractMainThread;
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RefPtr<SharedThreadPool> mThreadPool;
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#ifdef MOZ_WEBRTC
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RefPtr<AudioOutputObserver> mFarendObserverRef;
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#endif
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// used to limit graph shutdown time
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// Only accessed on the main thread.
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@ -1,4 +1,3 @@
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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
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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 file,
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* You can obtain one at http://mozilla.org/MPL/2.0/. */
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@ -29,9 +28,6 @@
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#include <algorithm>
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#include "DOMMediaStream.h"
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#include "GeckoProfiler.h"
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#ifdef MOZ_WEBRTC
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#include "AudioOutputObserver.h"
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#endif
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using namespace mozilla::layers;
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using namespace mozilla::dom;
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@ -1,61 +0,0 @@
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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 file,
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* You can obtain one at http://mozilla.org/MPL/2.0/. */
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#ifndef AUDIOOUTPUTOBSERVER_H_
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#define AUDIOOUTPUTOBSERVER_H_
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#include "mozilla/StaticPtr.h"
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#include "nsAutoPtr.h"
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#include "AudioMixer.h"
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#include "MediaData.h"
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namespace webrtc {
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class SingleRwFifo;
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}
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namespace mozilla {
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typedef struct FarEndAudioChunk_ {
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size_t mSamples;
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bool mOverrun;
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AudioDataValue mData[1]; // variable-length
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} FarEndAudioChunk;
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// This class is used to packetize and send the mixed audio from an MSG, in
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// float, to the AEC module of WebRTC.org.
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class AudioOutputObserver
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{
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public:
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AudioOutputObserver();
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NS_INLINE_DECL_THREADSAFE_REFCOUNTING(AudioOutputObserver);
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void Clear();
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void InsertFarEnd(const AudioDataValue *aBuffer, uint32_t aFrames, bool aOverran,
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int aFreq, int aChannels);
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uint32_t PlayoutFrequency() { return mPlayoutFreq; }
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uint32_t PlayoutChannels() { return mPlayoutChannels; }
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FarEndAudioChunk *Pop();
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uint32_t Size();
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private:
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virtual ~AudioOutputObserver();
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uint32_t mPlayoutFreq;
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uint32_t mPlayoutChannels;
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nsAutoPtr<webrtc::SingleRwFifo> mPlayoutFifo;
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uint32_t mChunkSize;
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// chunking to 10ms support
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FarEndAudioChunk *mSaved; // can't be nsAutoPtr since we need to use free(), not delete
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uint32_t mSamplesSaved;
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AlignedAudioBuffer mDownmixBuffer;
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};
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extern StaticRefPtr<AudioOutputObserver> gFarendObserver;
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}
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#endif
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@ -59,7 +59,6 @@
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#include "webrtc/modules/video_capture/video_capture_defines.h"
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#include "NullTransport.h"
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#include "AudioOutputObserver.h"
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namespace mozilla {
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@ -522,10 +521,10 @@ private:
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static int sChannelsOpen;
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const UniquePtr<webrtc::AudioProcessing> mAudioProcessing;
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const RefPtr<AudioOutputObserver> mAudioOutputObserver;
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// accessed from the GraphDriver thread except for deletion
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// accessed from the GraphDriver thread except for deletion.
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nsAutoPtr<AudioPacketizer<AudioDataValue, float>> mPacketizerInput;
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nsAutoPtr<AudioPacketizer<AudioDataValue, float>> mPacketizerOutput;
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// mMonitor protects mSources[] and mPrinicpalIds[] access/changes, and
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// transitions of mState from kStarted to kStopped (which are combined with
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@ -557,9 +556,12 @@ private:
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// To only update microphone when needed, we keep track of previous settings.
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MediaEnginePrefs mLastPrefs;
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// Stores the mixed audio output for the reverse-stream of the AEC.
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AlignedFloatBuffer mOutputBuffer;
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AlignedFloatBuffer mInputBuffer;
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AlignedFloatBuffer mDeinterleavedBuffer;
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AlignedAudioBuffer mInputDownmixBuffer;
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AlignedFloatBuffer mInputDownmixBuffer;
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};
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class MediaEngineWebRTC : public MediaEngine
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@ -1,3 +1,4 @@
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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
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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 file,
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* You can obtain one at http://mozilla.org/MPL/2.0/. */
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@ -53,134 +54,6 @@ NS_IMPL_ISUPPORTS0(MediaEngineWebRTCAudioCaptureSource)
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int MediaEngineWebRTCMicrophoneSource::sChannelsOpen = 0;
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AudioOutputObserver::AudioOutputObserver()
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: mPlayoutFreq(0)
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, mPlayoutChannels(0)
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, mChunkSize(0)
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, mSaved(nullptr)
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, mSamplesSaved(0)
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, mDownmixBuffer(MAX_SAMPLING_FREQ * MAX_CHANNELS / 100)
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{
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// Buffers of 10ms chunks
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mPlayoutFifo = new SingleRwFifo(MAX_AEC_FIFO_DEPTH/10);
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}
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AudioOutputObserver::~AudioOutputObserver()
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{
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Clear();
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free(mSaved);
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mSaved = nullptr;
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}
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void
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AudioOutputObserver::Clear()
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{
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while (mPlayoutFifo->size() > 0) {
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free(mPlayoutFifo->Pop());
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}
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// we'd like to touch mSaved here, but we can't if we might still be getting callbacks
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}
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FarEndAudioChunk *
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AudioOutputObserver::Pop()
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{
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return (FarEndAudioChunk *) mPlayoutFifo->Pop();
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}
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uint32_t
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AudioOutputObserver::Size()
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{
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return mPlayoutFifo->size();
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}
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// static
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void
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AudioOutputObserver::InsertFarEnd(const AudioDataValue *aBuffer, uint32_t aFrames, bool aOverran,
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int aFreq, int aChannels)
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{
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// Prepare for downmix if needed
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int channels = aChannels;
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if (aChannels > MAX_CHANNELS) {
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channels = MAX_CHANNELS;
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}
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if (mPlayoutChannels != 0) {
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if (mPlayoutChannels != static_cast<uint32_t>(channels)) {
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MOZ_CRASH();
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}
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} else {
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MOZ_ASSERT(channels <= MAX_CHANNELS);
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mPlayoutChannels = static_cast<uint32_t>(channels);
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}
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if (mPlayoutFreq != 0) {
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if (mPlayoutFreq != static_cast<uint32_t>(aFreq)) {
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MOZ_CRASH();
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}
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} else {
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MOZ_ASSERT(aFreq <= MAX_SAMPLING_FREQ);
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MOZ_ASSERT(!(aFreq % 100), "Sampling rate for far end data should be multiple of 100.");
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mPlayoutFreq = aFreq;
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mChunkSize = aFreq/100; // 10ms
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}
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#ifdef LOG_FAREND_INSERTION
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static FILE *fp = fopen("insertfarend.pcm","wb");
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#endif
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if (mSaved) {
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// flag overrun as soon as possible, and only once
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mSaved->mOverrun = aOverran;
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aOverran = false;
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}
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// Rechunk to 10ms.
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// The AnalyzeReverseStream() and WebRtcAec_BufferFarend() functions insist on 10ms
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// samples per call. Annoying...
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while (aFrames) {
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if (!mSaved) {
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mSaved = (FarEndAudioChunk *) moz_xmalloc(sizeof(FarEndAudioChunk) +
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(mChunkSize * channels - 1)*sizeof(AudioDataValue));
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mSaved->mSamples = mChunkSize;
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mSaved->mOverrun = aOverran;
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aOverran = false;
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}
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uint32_t to_copy = mChunkSize - mSamplesSaved;
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if (to_copy > aFrames) {
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to_copy = aFrames;
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}
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AudioDataValue* dest = &(mSaved->mData[mSamplesSaved * channels]);
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if (aChannels > MAX_CHANNELS) {
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AudioConverter converter(AudioConfig(aChannels, 0), AudioConfig(channels, 0));
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converter.Process(mDownmixBuffer, aBuffer, to_copy);
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ConvertAudioSamples(mDownmixBuffer.Data(), dest, to_copy * channels);
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} else {
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ConvertAudioSamples(aBuffer, dest, to_copy * channels);
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}
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#ifdef LOG_FAREND_INSERTION
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if (fp) {
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fwrite(&(mSaved->mData[mSamplesSaved * aChannels]), to_copy * aChannels, sizeof(AudioDataValue), fp);
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}
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#endif
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aFrames -= to_copy;
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mSamplesSaved += to_copy;
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aBuffer += to_copy * aChannels;
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if (mSamplesSaved >= mChunkSize) {
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int free_slots = mPlayoutFifo->capacity() - mPlayoutFifo->size();
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if (free_slots <= 0) {
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// XXX We should flag an overrun for the reader. We can't drop data from it due to
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// thread safety issues.
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break;
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} else {
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mPlayoutFifo->Push((int8_t *) mSaved); // takes ownership
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mSaved = nullptr;
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mSamplesSaved = 0;
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}
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}
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}
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}
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MediaEngineWebRTCMicrophoneSource::MediaEngineWebRTCMicrophoneSource(
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mozilla::AudioInput* aAudioInput,
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int aIndex,
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@ -191,7 +64,6 @@ MediaEngineWebRTCMicrophoneSource::MediaEngineWebRTCMicrophoneSource(
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: MediaEngineAudioSource(kReleased)
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, mAudioInput(aAudioInput)
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, mAudioProcessing(AudioProcessing::Create())
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, mAudioOutputObserver(new AudioOutputObserver())
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, mMonitor("WebRTCMic.Monitor")
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, mCapIndex(aIndex)
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, mDelayAgnostic(aDelayAgnostic)
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@ -633,8 +505,6 @@ MediaEngineWebRTCMicrophoneSource::Start(SourceMediaStream *aStream,
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// Make sure logger starts before capture
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AsyncLatencyLogger::Get(true);
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mAudioOutputObserver->Clear();
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mAudioInput->StartRecording(aStream, mListener);
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return NS_OK;
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@ -697,9 +567,88 @@ MediaEngineWebRTCMicrophoneSource::NotifyOutputData(MediaStreamGraph* aGraph,
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TrackRate aRate,
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uint32_t aChannels)
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{
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if (!PassThrough()) {
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mAudioOutputObserver->InsertFarEnd(aBuffer, aFrames, false,
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aRate, aChannels);
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if (!mPacketizerOutput ||
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mPacketizerOutput->PacketSize() != aRate/100u ||
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mPacketizerOutput->Channels() != aChannels) {
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// It's ok to drop the audio still in the packetizer here: if this changes,
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// we changed devices or something.
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mPacketizerOutput =
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new AudioPacketizer<AudioDataValue, float>(aRate/100, aChannels);
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}
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mPacketizerOutput->Input(aBuffer, aFrames);
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while (mPacketizerOutput->PacketsAvailable()) {
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uint32_t samplesPerPacket = mPacketizerOutput->PacketSize() *
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mPacketizerOutput->Channels();
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if (mOutputBuffer.Length() < samplesPerPacket) {
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mOutputBuffer.SetLength(samplesPerPacket);
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}
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if (mDeinterleavedBuffer.Length() < samplesPerPacket) {
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mDeinterleavedBuffer.SetLength(samplesPerPacket);
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}
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float* packet = mOutputBuffer.Data();
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mPacketizerOutput->Output(packet);
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AutoTArray<float*, MAX_CHANNELS> deinterleavedPacketDataChannelPointers;
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float* interleavedFarend = nullptr;
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uint32_t channelCountFarend = 0;
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uint32_t framesPerPacketFarend = 0;
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// Downmix from aChannels to MAX_CHANNELS if needed. We always have floats
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// here, the packetized performed the conversion.
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if (aChannels > MAX_CHANNELS) {
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AudioConverter converter(AudioConfig(aChannels, 0, AudioConfig::FORMAT_FLT),
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AudioConfig(MAX_CHANNELS, 0, AudioConfig::FORMAT_FLT));
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framesPerPacketFarend = mPacketizerOutput->PacketSize();
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framesPerPacketFarend =
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converter.Process(mInputDownmixBuffer,
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packet,
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framesPerPacketFarend);
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interleavedFarend = mInputDownmixBuffer.Data();
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channelCountFarend = MAX_CHANNELS;
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deinterleavedPacketDataChannelPointers.SetLength(MAX_CHANNELS);
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} else {
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interleavedFarend = packet;
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channelCountFarend = aChannels;
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framesPerPacketFarend = mPacketizerOutput->PacketSize();
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deinterleavedPacketDataChannelPointers.SetLength(aChannels);
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}
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MOZ_ASSERT(interleavedFarend &&
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(channelCountFarend == 1 || channelCountFarend == 2) &&
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framesPerPacketFarend);
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if (mInputBuffer.Length() < framesPerPacketFarend * channelCountFarend) {
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mInputBuffer.SetLength(framesPerPacketFarend * channelCountFarend);
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}
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size_t offset = 0;
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for (size_t i = 0; i < deinterleavedPacketDataChannelPointers.Length(); ++i) {
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deinterleavedPacketDataChannelPointers[i] = mInputBuffer.Data() + offset;
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offset += framesPerPacketFarend;
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}
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// Deinterleave, prepare a channel pointers array, with enough storage for
|
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// the frames.
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DeinterleaveAndConvertBuffer(interleavedFarend,
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framesPerPacketFarend,
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channelCountFarend,
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deinterleavedPacketDataChannelPointers.Elements());
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// Having the same config for input and output means we potentially save
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// some CPU.
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StreamConfig inputConfig(aRate, channelCountFarend, false);
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StreamConfig outputConfig = inputConfig;
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// Passing the same pointers here saves a copy inside this function.
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DebugOnly<int> err =
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mAudioProcessing->ProcessReverseStream(deinterleavedPacketDataChannelPointers.Elements(),
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inputConfig,
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outputConfig,
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deinterleavedPacketDataChannelPointers.Elements());
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MOZ_ASSERT(!err, "Could not process the reverse stream.");
|
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}
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}
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|
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@ -722,94 +671,11 @@ MediaEngineWebRTCMicrophoneSource::PacketizeAndProcess(MediaStreamGraph* aGraph,
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new AudioPacketizer<AudioDataValue, float>(aRate/100, aChannels);
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}
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// On initial capture, throw away all far-end data except the most recent sample
|
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// since it's already irrelevant and we want to keep avoid confusing the AEC far-end
|
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// input code with "old" audio.
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// On initial capture, throw away all far-end data except the most recent
|
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// sample since it's already irrelevant and we want to avoid confusing the AEC
|
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// far-end input code with "old" audio.
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if (!mStarted) {
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mStarted = true;
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while (mAudioOutputObserver->Size() > 1) {
|
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free(mAudioOutputObserver->Pop()); // only call if size() > 0
|
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}
|
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}
|
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|
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// Feed the far-end audio data (speakers) to the feedback input of the AEC.
|
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while (mAudioOutputObserver->Size() > 0) {
|
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// Bug 1414837: This will call `free()`, and we should remove it.
|
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// Pop gives ownership.
|
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nsAutoPtr<FarEndAudioChunk> buffer(mAudioOutputObserver->Pop()); // only call if size() > 0
|
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if (!buffer) {
|
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continue;
|
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}
|
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AudioDataValue* packetDataPointer = buffer->mData;
|
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AutoTArray<float*, MAX_CHANNELS> deinterleavedPacketDataChannelPointers;
|
||||
AudioDataValue* interleavedFarend = nullptr;
|
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uint32_t channelCountFarend = 0;
|
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uint32_t framesPerPacketFarend = 0;
|
||||
|
||||
// Downmix from aChannels to MAX_CHANNELS if needed
|
||||
if (mAudioOutputObserver->PlayoutChannels() > MAX_CHANNELS) {
|
||||
AudioConverter converter(AudioConfig(aChannels, 0, AudioConfig::FORMAT_DEFAULT),
|
||||
AudioConfig(MAX_CHANNELS, 0, AudioConfig::FORMAT_DEFAULT));
|
||||
framesPerPacketFarend =
|
||||
buffer->mSamples;
|
||||
framesPerPacketFarend =
|
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converter.Process(mInputDownmixBuffer,
|
||||
packetDataPointer,
|
||||
framesPerPacketFarend);
|
||||
interleavedFarend = mInputDownmixBuffer.Data();
|
||||
channelCountFarend = MAX_CHANNELS;
|
||||
deinterleavedPacketDataChannelPointers.SetLength(MAX_CHANNELS);
|
||||
} else {
|
||||
uint32_t outputChannels = mAudioOutputObserver->PlayoutChannels();
|
||||
interleavedFarend = packetDataPointer;
|
||||
channelCountFarend = outputChannels;
|
||||
framesPerPacketFarend = buffer->mSamples;
|
||||
deinterleavedPacketDataChannelPointers.SetLength(outputChannels);
|
||||
}
|
||||
|
||||
MOZ_ASSERT(interleavedFarend &&
|
||||
(channelCountFarend == 1 || channelCountFarend == 2) &&
|
||||
framesPerPacketFarend);
|
||||
|
||||
if (mInputBuffer.Length() < framesPerPacketFarend * channelCountFarend) {
|
||||
mInputBuffer.SetLength(framesPerPacketFarend * channelCountFarend);
|
||||
}
|
||||
|
||||
offset = 0;
|
||||
for (size_t i = 0; i < deinterleavedPacketDataChannelPointers.Length(); ++i) {
|
||||
deinterleavedPacketDataChannelPointers[i] = mInputBuffer.Data() + offset;
|
||||
offset += framesPerPacketFarend;
|
||||
}
|
||||
|
||||
// Deinterleave, prepare a channel pointers array, with enough storage for
|
||||
// the frames.
|
||||
//
|
||||
// If this is a platform that uses s16 for audio input and output,
|
||||
// convert to floats, the APM API we use only accepts floats.
|
||||
DeinterleaveAndConvertBuffer(interleavedFarend,
|
||||
framesPerPacketFarend,
|
||||
channelCountFarend,
|
||||
deinterleavedPacketDataChannelPointers.Elements());
|
||||
|
||||
// Having the same config for input and output means we potentially save
|
||||
// some CPU.
|
||||
StreamConfig inputConfig(mAudioOutputObserver->PlayoutFrequency(),
|
||||
channelCountFarend,
|
||||
false /* we don't use typing detection*/);
|
||||
StreamConfig outputConfig = inputConfig;
|
||||
|
||||
// Passing the same pointers here saves a copy inside this function.
|
||||
int err =
|
||||
mAudioProcessing->ProcessReverseStream(deinterleavedPacketDataChannelPointers.Elements(),
|
||||
inputConfig,
|
||||
outputConfig,
|
||||
deinterleavedPacketDataChannelPointers.Elements());
|
||||
|
||||
if (err) {
|
||||
MOZ_LOG(GetMediaManagerLog(), LogLevel::Error,
|
||||
("error in audio ProcessReverseStream(): %d", err));
|
||||
return;
|
||||
}
|
||||
}
|
||||
|
||||
// Packetize our input data into 10ms chunks, deinterleave into planar channel
|
||||
|
|
|
|||
|
|
@ -22,8 +22,7 @@ EXPORTS += [
|
|||
]
|
||||
|
||||
if CONFIG['MOZ_WEBRTC']:
|
||||
EXPORTS += ['AudioOutputObserver.h',
|
||||
'MediaEngineRemoteVideoSource.h',
|
||||
EXPORTS += ['MediaEngineRemoteVideoSource.h',
|
||||
'MediaEngineWebRTC.h']
|
||||
EXPORTS.mozilla.dom += [ 'RTCIdentityProviderRegistrar.h' ]
|
||||
UNIFIED_SOURCES += [
|
||||
|
|
|
|||
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