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+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* 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 MediaEngineWebRTCAudio_h
+#define MediaEngineWebRTCAudio_h
+
+#include "AudioPacketizer.h"
+#include "AudioSegment.h"
+#include "AudioDeviceInfo.h"
+#include "DeviceInputTrack.h"
+#include "MediaEngineWebRTC.h"
+#include "MediaEnginePrefs.h"
+#include "MediaTrackListener.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace mozilla {
+
+class AudioInputProcessing;
+class AudioProcessingTrack;
+
+// This class is created and used exclusively on the Media Manager thread, with
+// exactly two exceptions:
+// - Pull is always called on the MTG thread. It only ever uses
+// mInputProcessing. mInputProcessing is set, then a message is sent first to
+// the main thread and then the MTG thread so that it can be used as part of
+// the graph processing. On destruction, similarly, a message is sent to the
+// graph so that it stops using it, and then it is deleted.
+// - mSettings is created on the MediaManager thread is always ever accessed on
+// the Main Thread. It is const.
+class MediaEngineWebRTCMicrophoneSource : public MediaEngineSource {
+ public:
+ explicit MediaEngineWebRTCMicrophoneSource(const MediaDevice* aMediaDevice);
+
+ nsresult Allocate(const dom::MediaTrackConstraints& aConstraints,
+ const MediaEnginePrefs& aPrefs, uint64_t aWindowID,
+ const char** aOutBadConstraint) override;
+ nsresult Deallocate() override;
+ void SetTrack(const RefPtr<MediaTrack>& aTrack,
+ const PrincipalHandle& aPrincipal) override;
+ nsresult Start() override;
+ nsresult Stop() override;
+ nsresult Reconfigure(const dom::MediaTrackConstraints& aConstraints,
+ const MediaEnginePrefs& aPrefs,
+ const char** aOutBadConstraint) override;
+
+ /**
+ * Assigns the current settings of the capture to aOutSettings.
+ * Main thread only.
+ */
+ void GetSettings(dom::MediaTrackSettings& aOutSettings) const override;
+
+ nsresult TakePhoto(MediaEnginePhotoCallback* aCallback) override {
+ return NS_ERROR_NOT_IMPLEMENTED;
+ }
+
+ protected:
+ ~MediaEngineWebRTCMicrophoneSource() = default;
+
+ private:
+ /**
+ * From a set of constraints and about:config preferences, output the correct
+ * set of preferences that can be sent to AudioInputProcessing.
+ *
+ * This can fail if the number of channels requested is zero, negative, or
+ * more than the device supports.
+ */
+ nsresult EvaluateSettings(const NormalizedConstraints& aConstraintsUpdate,
+ const MediaEnginePrefs& aInPrefs,
+ MediaEnginePrefs* aOutPrefs,
+ const char** aOutBadConstraint);
+ /**
+ * From settings output by EvaluateSettings, send those settings to the
+ * AudioInputProcessing instance and the main thread (for use in GetSettings).
+ */
+ void ApplySettings(const MediaEnginePrefs& aPrefs);
+
+ PrincipalHandle mPrincipal = PRINCIPAL_HANDLE_NONE;
+
+ const RefPtr<AudioDeviceInfo> mDeviceInfo;
+
+ // The maximum number of channels that this device supports.
+ const uint32_t mDeviceMaxChannelCount;
+ // The current settings for the underlying device.
+ // Constructed on the MediaManager thread, and then only ever accessed on the
+ // main thread.
+ const nsMainThreadPtrHandle<media::Refcountable<dom::MediaTrackSettings>>
+ mSettings;
+
+ // Current state of the resource for this source.
+ MediaEngineSourceState mState;
+
+ // The current preferences that will be forwarded to mAudioProcessingConfig
+ // below.
+ MediaEnginePrefs mCurrentPrefs;
+
+ // The AudioProcessingTrack used to inteface with the MediaTrackGraph. Set in
+ // SetTrack as part of the initialization, and nulled in ::Deallocate.
+ RefPtr<AudioProcessingTrack> mTrack;
+
+ // See note at the top of this class.
+ RefPtr<AudioInputProcessing> mInputProcessing;
+
+ // Copy of the config currently applied to AudioProcessing through
+ // mInputProcessing.
+ webrtc::AudioProcessing::Config mAudioProcessingConfig;
+};
+
+// This class is created on the MediaManager thread, and then exclusively used
+// on the MTG thread.
+// All communication is done via message passing using MTG ControlMessages
+class AudioInputProcessing : public AudioDataListener {
+ public:
+ explicit AudioInputProcessing(uint32_t aMaxChannelCount);
+ void Process(MediaTrackGraph* aGraph, GraphTime aFrom, GraphTime aTo,
+ AudioSegment* aInput, AudioSegment* aOutput);
+
+ void ProcessOutputData(MediaTrackGraph* aGraph, const AudioChunk& aChunk);
+ bool IsVoiceInput(MediaTrackGraph* aGraph) const override {
+ // If we're passing data directly without AEC or any other process, this
+ // means that all voice-processing has been disabled intentionaly. In this
+ // case, consider that the device is not used for voice input.
+ return !PassThrough(aGraph);
+ }
+
+ void Start(MediaTrackGraph* aGraph);
+ void Stop(MediaTrackGraph* aGraph);
+
+ void DeviceChanged(MediaTrackGraph* aGraph) override;
+
+ uint32_t RequestedInputChannelCount(MediaTrackGraph*) override {
+ return GetRequestedInputChannelCount();
+ }
+
+ void Disconnect(MediaTrackGraph* aGraph) override;
+
+ void PacketizeAndProcess(MediaTrackGraph* aGraph,
+ const AudioSegment& aSegment);
+
+ void SetPassThrough(MediaTrackGraph* aGraph, bool aPassThrough);
+ uint32_t GetRequestedInputChannelCount();
+ void SetRequestedInputChannelCount(MediaTrackGraph* aGraph,
+ CubebUtils::AudioDeviceID aDeviceId,
+ uint32_t aRequestedInputChannelCount);
+ // This is true when all processing is disabled, we can skip
+ // packetization, resampling and other processing passes.
+ bool PassThrough(MediaTrackGraph* aGraph) const;
+
+ // This allow changing the APM options, enabling or disabling processing
+ // steps. The config gets applied the next time we're about to process input
+ // data.
+ void ApplyConfig(MediaTrackGraph* aGraph,
+ const webrtc::AudioProcessing::Config& aConfig);
+
+ void End();
+
+ TrackTime NumBufferedFrames(MediaTrackGraph* aGraph) const;
+
+ // The packet size contains samples in 10ms. The unit of aRate is hz.
+ static uint32_t GetPacketSize(TrackRate aRate) {
+ return webrtc::AudioProcessing::GetFrameSize(aRate);
+ }
+
+ bool IsEnded() const { return mEnded; }
+
+ private:
+ ~AudioInputProcessing() = default;
+ void EnsureAudioProcessing(MediaTrackGraph* aGraph, uint32_t aChannels);
+ void ResetAudioProcessing(MediaTrackGraph* aGraph);
+ PrincipalHandle GetCheckedPrincipal(const AudioSegment& aSegment);
+ // This implements the processing algoritm to apply to the input (e.g. a
+ // microphone). If all algorithms are disabled, this class in not used. This
+ // class only accepts audio chunks of 10ms. It has two inputs and one output:
+ // it is fed the speaker data and the microphone data. It outputs processed
+ // input data.
+ const UniquePtr<webrtc::AudioProcessing> mAudioProcessing;
+ // Packetizer to be able to feed 10ms packets to the input side of
+ // mAudioProcessing. Not used if the processing is bypassed.
+ Maybe<AudioPacketizer<AudioDataValue, float>> mPacketizerInput;
+ // The number of channels asked for by content, after clamping to the range of
+ // legal channel count for this particular device.
+ uint32_t mRequestedInputChannelCount;
+ // mSkipProcessing is true if none of the processing passes are enabled,
+ // because of prefs or constraints. This allows simply copying the audio into
+ // the MTG, skipping resampling and the whole webrtc.org code.
+ bool mSkipProcessing;
+ // Buffer for up to one 10ms packet of planar mixed audio output for the
+ // reverse-stream (speaker data) of mAudioProcessing AEC.
+ // Length is packet size * channel count, regardless of how many frames are
+ // buffered. Not used if the processing is bypassed.
+ AlignedFloatBuffer mOutputBuffer;
+ // Number of channels into which mOutputBuffer is divided.
+ uint32_t mOutputBufferChannelCount = 0;
+ // Number of frames buffered in mOutputBuffer for the reverse stream.
+ uint32_t mOutputBufferFrameCount = 0;
+ // Stores the input audio, to be processed by the APM.
+ AlignedFloatBuffer mInputBuffer;
+ // Stores the deinterleaved microphone audio
+ AlignedFloatBuffer mDeinterleavedBuffer;
+ // Stores the mixed down input audio
+ AlignedFloatBuffer mInputDownmixBuffer;
+ // Stores data waiting to be pulled.
+ AudioSegment mSegment;
+ // Whether or not this MediaEngine is enabled. If it's not enabled, it
+ // operates in "pull" mode, and we append silence only, releasing the audio
+ // input track.
+ bool mEnabled;
+ // Whether or not we've ended and removed the AudioProcessingTrack.
+ bool mEnded;
+ // When processing is enabled, the number of packets received by this
+ // instance, to implement periodic logging.
+ uint64_t mPacketCount;
+ // Temporary descriptor for a slice of an AudioChunk parameter passed to
+ // ProcessOutputData(). This is a member rather than on the stack so that
+ // any memory allocated for its mChannelData pointer array is not
+ // reallocated on each iteration.
+ AudioChunk mSubChunk;
+ // A storage holding the interleaved audio data converted the AudioSegment.
+ // This will be used as an input parameter for PacketizeAndProcess. This
+ // should be removed once bug 1729041 is done.
+ AutoTArray<AudioDataValue,
+ SilentChannel::AUDIO_PROCESSING_FRAMES * GUESS_AUDIO_CHANNELS>
+ mInterleavedBuffer;
+ // Tracks the pending frames with paired principals piled up in packetizer.
+ std::deque<std::pair<TrackTime, PrincipalHandle>> mChunksInPacketizer;
+};
+
+// MediaTrack subclass tailored for MediaEngineWebRTCMicrophoneSource.
+class AudioProcessingTrack : public DeviceInputConsumerTrack {
+ // Only accessed on the graph thread.
+ RefPtr<AudioInputProcessing> mInputProcessing;
+
+ explicit AudioProcessingTrack(TrackRate aSampleRate)
+ : DeviceInputConsumerTrack(aSampleRate) {}
+
+ ~AudioProcessingTrack() = default;
+
+ public:
+ // Main Thread API
+ void Destroy() override;
+ void SetInputProcessing(RefPtr<AudioInputProcessing> aInputProcessing);
+ static AudioProcessingTrack* Create(MediaTrackGraph* aGraph);
+
+ // Graph Thread API
+ void DestroyImpl() override;
+ void ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags) override;
+ uint32_t NumberOfChannels() const override {
+ MOZ_DIAGNOSTIC_ASSERT(
+ mInputProcessing,
+ "Must set mInputProcessing before exposing to content");
+ return mInputProcessing->GetRequestedInputChannelCount();
+ }
+ // Pass the graph's mixed audio output to mInputProcessing for processing as
+ // the reverse stream.
+ void NotifyOutputData(MediaTrackGraph* aGraph, const AudioChunk& aChunk);
+
+ // Any thread
+ AudioProcessingTrack* AsAudioProcessingTrack() override { return this; }
+
+ private:
+ // Graph thread API
+ void SetInputProcessingImpl(RefPtr<AudioInputProcessing> aInputProcessing);
+};
+
+class MediaEngineWebRTCAudioCaptureSource : public MediaEngineSource {
+ public:
+ explicit MediaEngineWebRTCAudioCaptureSource(const MediaDevice* aMediaDevice);
+ static nsString GetUUID();
+ static nsString GetGroupId();
+ nsresult Allocate(const dom::MediaTrackConstraints& aConstraints,
+ const MediaEnginePrefs& aPrefs, uint64_t aWindowID,
+ const char** aOutBadConstraint) override {
+ // Nothing to do here, everything is managed in MediaManager.cpp
+ return NS_OK;
+ }
+ nsresult Deallocate() override {
+ // Nothing to do here, everything is managed in MediaManager.cpp
+ return NS_OK;
+ }
+ void SetTrack(const RefPtr<MediaTrack>& aTrack,
+ const PrincipalHandle& aPrincipal) override;
+ nsresult Start() override;
+ nsresult Stop() override;
+ nsresult Reconfigure(const dom::MediaTrackConstraints& aConstraints,
+ const MediaEnginePrefs& aPrefs,
+ const char** aOutBadConstraint) override;
+
+ nsresult TakePhoto(MediaEnginePhotoCallback* aCallback) override {
+ return NS_ERROR_NOT_IMPLEMENTED;
+ }
+
+ void GetSettings(dom::MediaTrackSettings& aOutSettings) const override;
+
+ protected:
+ virtual ~MediaEngineWebRTCAudioCaptureSource() = default;
+};
+
+} // end namespace mozilla
+
+#endif // MediaEngineWebRTCAudio_h