Adding upstream version 86.0.1.upstream/86.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 339 insertions, 0 deletions

diff --git a/dom/media/webrtc/MediaEngineWebRTCAudio.h b/dom/media/webrtc/MediaEngineWebRTCAudio.h
new file mode 100644
index 0000000000..3536c3df79
--- /dev/null
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.h
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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 "MediaEngineWebRTC.h"
+#include "MediaTrackListener.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+
+namespace mozilla {
+
+class AudioInputProcessing;
+class AudioInputTrack;
+
+// 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:
+  MediaEngineWebRTCMicrophoneSource(RefPtr<AudioDeviceInfo> aInfo,
+                                    const nsString& aDeviceName,
+                                    const nsCString& aDeviceUUID,
+                                    const nsString& aDeviceGroup,
+                                    uint32_t aMaxChannelCount,
+                                    bool aDelayAgnostic, bool aExtendedFilter);
+
+  nsString GetName() const override;
+  nsCString GetUUID() const override;
+  nsString GetGroupId() const override;
+
+  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;
+
+  dom::MediaSourceEnum GetMediaSource() const override {
+    return dom::MediaSourceEnum::Microphone;
+  }
+
+  nsresult TakePhoto(MediaEnginePhotoCallback* aCallback) override {
+    return NS_ERROR_NOT_IMPLEMENTED;
+  }
+
+  void Shutdown() override;
+
+ 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);
+
+  /**
+   * Sent the AudioProcessingModule parameter for a given processing algorithm.
+   */
+  void UpdateAECSettings(bool aEnable, bool aUseAecMobile,
+                         webrtc::EchoCancellation::SuppressionLevel aLevel,
+                         webrtc::EchoControlMobile::RoutingMode aRoutingMode);
+  void UpdateAGCSettings(bool aEnable, webrtc::GainControl::Mode aMode);
+  void UpdateHPFSettings(bool aEnable);
+  void UpdateNSSettings(bool aEnable, webrtc::NoiseSuppression::Level aLevel);
+  void UpdateAPMExtraOptions(bool aExtendedFilter, bool aDelayAgnostic);
+
+  PrincipalHandle mPrincipal = PRINCIPAL_HANDLE_NONE;
+
+  const RefPtr<AudioDeviceInfo> mDeviceInfo;
+  const bool mDelayAgnostic;
+  const bool mExtendedFilter;
+  const nsString mDeviceName;
+  const nsCString mDeviceUUID;
+  const nsString mDeviceGroup;
+
+  // 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 for the APM's various processing stages.
+  MediaEnginePrefs mCurrentPrefs;
+
+  // The AudioInputTrack used to inteface with the MediaTrackGraph. Set in
+  // SetTrack as part of the initialization, and nulled in ::Deallocate.
+  RefPtr<AudioInputTrack> mTrack;
+
+  // See note at the top of this class.
+  RefPtr<AudioInputProcessing> mInputProcessing;
+};
+
+// 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:
+  AudioInputProcessing(uint32_t aMaxChannelCount,
+                       const PrincipalHandle& aPrincipalHandle);
+
+  void Pull(MediaTrackGraphImpl* aGraph, GraphTime aFrom, GraphTime aTo,
+            GraphTime aTrackEnd, AudioSegment* aSegment,
+            bool aLastPullThisIteration, bool* aEnded);
+
+  void NotifyOutputData(MediaTrackGraphImpl* aGraph, AudioDataValue* aBuffer,
+                        size_t aFrames, TrackRate aRate,
+                        uint32_t aChannels) override;
+  void NotifyInputStopped(MediaTrackGraphImpl* aGraph) override;
+  void NotifyInputData(MediaTrackGraphImpl* aGraph,
+                       const AudioDataValue* aBuffer, size_t aFrames,
+                       TrackRate aRate, uint32_t aChannels,
+                       uint32_t aAlreadyBuffered) override;
+  bool IsVoiceInput(MediaTrackGraphImpl* 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();
+  void Stop();
+
+  void DeviceChanged(MediaTrackGraphImpl* aGraph) override;
+
+  uint32_t RequestedInputChannelCount(MediaTrackGraphImpl*) override {
+    return GetRequestedInputChannelCount();
+  }
+
+  void Disconnect(MediaTrackGraphImpl* aGraph) override;
+
+  template <typename T>
+  void InsertInGraph(MediaTrackGraphImpl* aGraph, const T* aBuffer,
+                     size_t aFrames, uint32_t aChannels);
+
+  void PacketizeAndProcess(MediaTrackGraphImpl* aGraph,
+                           const AudioDataValue* aBuffer, size_t aFrames,
+                           TrackRate aRate, uint32_t aChannels);
+
+  void SetPassThrough(MediaTrackGraphImpl* aGraph, bool aPassThrough);
+  uint32_t GetRequestedInputChannelCount();
+  void SetRequestedInputChannelCount(MediaTrackGraphImpl* aGraph,
+                                     uint32_t aRequestedInputChannelCount);
+  // This is true when all processing is disabled, we can skip
+  // packetization, resampling and other processing passes.
+  bool PassThrough(MediaTrackGraphImpl* aGraphImpl) const;
+
+  // This allow changing the APM options, enabling or disabling processing
+  // steps.
+  void UpdateAECSettings(bool aEnable, bool aUseAecMobile,
+                         webrtc::EchoCancellation::SuppressionLevel aLevel,
+                         webrtc::EchoControlMobile::RoutingMode aRoutingMode);
+  void UpdateAGCSettings(bool aEnable, webrtc::GainControl::Mode aMode);
+  void UpdateHPFSettings(bool aEnable);
+  void UpdateNSSettings(bool aEnable, webrtc::NoiseSuppression::Level aLevel);
+  void UpdateAPMExtraOptions(bool aExtendedFilter, bool aDelayAgnostic);
+
+  void End();
+
+ private:
+  ~AudioInputProcessing() = default;
+  // 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.
+  UniquePtr<AudioPacketizer<AudioDataValue, float>> mPacketizerInput;
+  // Packetizer to be able to feed 10ms packets to the output side of
+  // mAudioProcessing. Not used if the processing is bypassed.
+  UniquePtr<AudioPacketizer<AudioDataValue, float>> mPacketizerOutput;
+  // The number of channels asked for by content, after clamping to the range of
+  // legal channel count for this particular device. This is the number of
+  // channels of the input buffer passed as parameter in NotifyInputData.
+  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;
+  // Stores the mixed audio output for the reverse-stream of the AEC (the
+  // speaker data).
+  AlignedFloatBuffer mOutputBuffer;
+  // 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;
+  // Set to false by Start(). Becomes true after the first time we append real
+  // audio frames from the audio callback.
+  bool mLiveFramesAppended;
+  // Once live frames have been appended, this is the number of frames appended
+  // as pre-buffer for that data, to avoid underruns. Buffering in the track
+  // might be needed because of the AUDIO_BLOCK interval at which we run the
+  // graph, the packetizer keeping some input data. Care must be taken when
+  // turning on and off the packetizer.
+  TrackTime mLiveBufferingAppended;
+  // Principal for the data that flows through this class.
+  const PrincipalHandle mPrincipal;
+  // 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 AudioInputTrack.
+  bool mEnded;
+};
+
+// MediaTrack subclass tailored for MediaEngineWebRTCMicrophoneSource.
+class AudioInputTrack : public ProcessedMediaTrack {
+  // Only accessed on the graph thread.
+  RefPtr<AudioInputProcessing> mInputProcessing;
+
+  // Only accessed on the main thread. Used for bookkeeping on main thread, such
+  // that CloseAudioInput can be idempotent.
+  // XXX Should really be a CubebUtils::AudioDeviceID, but they aren't
+  // copyable (opaque pointers)
+  RefPtr<AudioDataListener> mInputListener;
+
+  explicit AudioInputTrack(TrackRate aSampleRate)
+      : ProcessedMediaTrack(aSampleRate, MediaSegment::AUDIO,
+                            new AudioSegment()) {}
+
+  ~AudioInputTrack() = default;
+
+ public:
+  // Main Thread API
+  // Users of audio inputs go through the track so it can track when the
+  // last track referencing an input goes away, so it can close the cubeb
+  // input. Main thread only.
+  nsresult OpenAudioInput(CubebUtils::AudioDeviceID aId,
+                          AudioDataListener* aListener);
+  void CloseAudioInput(Maybe<CubebUtils::AudioDeviceID>& aId);
+  void Destroy() override;
+  void SetInputProcessing(RefPtr<AudioInputProcessing> aInputProcessing);
+  static AudioInputTrack* 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();
+  }
+
+  // Any thread
+  AudioInputTrack* AsAudioInputTrack() override { return this; }
+
+ private:
+  // Graph thread API
+  void SetInputProcessingImpl(RefPtr<AudioInputProcessing> aInputProcessing);
+};
+
+class MediaEngineWebRTCAudioCaptureSource : public MediaEngineSource {
+ public:
+  explicit MediaEngineWebRTCAudioCaptureSource(const char* aUuid) {}
+  nsString GetName() const override;
+  nsCString GetUUID() const override;
+  nsString GetGroupId() const override;
+  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;
+
+  dom::MediaSourceEnum GetMediaSource() const override {
+    return dom::MediaSourceEnum::AudioCapture;
+  }
+
+  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