1 files changed, 303 insertions, 0 deletions
diff --git a/dom/media/webrtc/libwebrtcglue/AudioConduit.h b/dom/media/webrtc/libwebrtcglue/AudioConduit.h
new file mode 100644
index 0000000000..e8de331e12
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/AudioConduit.h
@@ -0,0 +1,303 @@
+/* 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 AUDIO_SESSION_H_
+#define AUDIO_SESSION_H_
+
+#include "mozilla/Attributes.h"
+#include "mozilla/ReentrantMonitor.h"
+#include "mozilla/RWLock.h"
+#include "mozilla/StateMirroring.h"
+#include "mozilla/TimeStamp.h"
+
+#include "MediaConduitInterface.h"
+#include "common/MediaEngineWrapper.h"
+
+/**
+ * This file hosts several structures identifying different aspects of a RTP
+ * Session.
+ */
+namespace mozilla {
+
+struct DtmfEvent;
+
+/**
+ * Concrete class for Audio session. Hooks up
+ *  - media-source and target to external transport
+ */
+class WebrtcAudioConduit : public AudioSessionConduit,
+                           public webrtc::RtcpEventObserver {
+ public:
+  Maybe<int> ActiveSendPayloadType() const override;
+  Maybe<int> ActiveRecvPayloadType() const override;
+
+  void OnRtpReceived(webrtc::RtpPacketReceived&& aPacket,
+                     webrtc::RTPHeader&& aHeader);
+  void OnRtcpReceived(MediaPacket&& aPacket);
+
+  void OnRtcpBye() override;
+  void OnRtcpTimeout() override;
+
+  void SetTransportActive(bool aActive) override;
+
+  MediaEventSourceExc<MediaPacket>& SenderRtpSendEvent() override {
+    return mSenderRtpSendEvent;
+  }
+  MediaEventSourceExc<MediaPacket>& SenderRtcpSendEvent() override {
+    return mSenderRtcpSendEvent;
+  }
+  MediaEventSourceExc<MediaPacket>& ReceiverRtcpSendEvent() override {
+    return mReceiverRtcpSendEvent;
+  }
+  void ConnectReceiverRtpEvent(
+      MediaEventSourceExc<webrtc::RtpPacketReceived, webrtc::RTPHeader>& aEvent)
+      override {
+    mReceiverRtpEventListener =
+        aEvent.Connect(mCallThread, this, &WebrtcAudioConduit::OnRtpReceived);
+  }
+  void ConnectReceiverRtcpEvent(
+      MediaEventSourceExc<MediaPacket>& aEvent) override {
+    mReceiverRtcpEventListener =
+        aEvent.Connect(mCallThread, this, &WebrtcAudioConduit::OnRtcpReceived);
+  }
+  void ConnectSenderRtcpEvent(
+      MediaEventSourceExc<MediaPacket>& aEvent) override {
+    mSenderRtcpEventListener =
+        aEvent.Connect(mCallThread, this, &WebrtcAudioConduit::OnRtcpReceived);
+  }
+
+  Maybe<uint16_t> RtpSendBaseSeqFor(uint32_t aSsrc) const override;
+
+  const dom::RTCStatsTimestampMaker& GetTimestampMaker() const override;
+
+  void StopTransmitting();
+  void StartTransmitting();
+  void StopReceiving();
+  void StartReceiving();
+
+  /**
+   * Function to deliver externally captured audio sample for encoding and
+   * transport
+   * @param frame [in]: AudioFrame in upstream's format for forwarding to the
+   *                    send stream. Ownership is passed along.
+   * NOTE: ConfigureSendMediaCodec() SHOULD be called before this function can
+   * be invoked. This ensures the inserted audio-samples can be transmitted by
+   * the conduit.
+   */
+  MediaConduitErrorCode SendAudioFrame(
+      std::unique_ptr<webrtc::AudioFrame> frame) override;
+
+  /**
+   * Function to grab a decoded audio-sample from the media engine for
+   * rendering / playout of length 10 milliseconds.
+   *
+   * @param samplingFreqHz [in]: Frequency of the sampling for playback in
+   *                             Hertz (16000, 32000,..)
+   * @param frame [in/out]: Pointer to an AudioFrame to which audio data will be
+   *                        copied
+   * NOTE: This function should be invoked every 10 milliseconds for the best
+   *       performance
+   * NOTE: ConfigureRecvMediaCodec() SHOULD be called before this function can
+   *       be invoked
+   * This ensures the decoded samples are ready for reading and playout is
+   * enabled.
+   */
+  MediaConduitErrorCode GetAudioFrame(int32_t samplingFreqHz,
+                                      webrtc::AudioFrame* frame) override;
+
+  bool SendRtp(const uint8_t* aData, size_t aLength,
+               const webrtc::PacketOptions& aOptions) override;
+  bool SendSenderRtcp(const uint8_t* aData, size_t aLength) override;
+  bool SendReceiverRtcp(const uint8_t* aData, size_t aLength) override;
+
+  bool HasCodecPluginID(uint64_t aPluginID) const override { return false; }
+
+  void SetJitterBufferTarget(DOMHighResTimeStamp aTargetMs) override;
+
+  void DeliverPacket(rtc::CopyOnWriteBuffer packet, PacketType type) override;
+
+  RefPtr<GenericPromise> Shutdown() override;
+
+  WebrtcAudioConduit(RefPtr<WebrtcCallWrapper> aCall,
+                     nsCOMPtr<nsISerialEventTarget> aStsThread);
+
+  virtual ~WebrtcAudioConduit();
+
+  // Call thread.
+  void InitControl(AudioConduitControlInterface* aControl) override;
+
+  // Handle a DTMF event from mControl.mOnDtmfEventListener.
+  void OnDtmfEvent(const DtmfEvent& aEvent);
+
+  // Called when a parameter in mControl has changed. Call thread.
+  void OnControlConfigChange();
+
+  Ssrcs GetLocalSSRCs() const override;
+  Maybe<Ssrc> GetRemoteSSRC() const override;
+
+  void DisableSsrcChanges() override {
+    MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+    mAllowSsrcChange = false;
+  }
+
+ private:
+  /**
+   * Override the remote ssrc configured on mRecvStreamConfig.
+   *
+   * Recreates and restarts the recv stream if needed. The overriden value is
+   * overwritten the next time the mControl.mRemoteSsrc mirror changes value.
+   *
+   * Call thread only.
+   */
+  bool OverrideRemoteSSRC(uint32_t aSsrc);
+
+ public:
+  void UnsetRemoteSSRC(uint32_t aSsrc) override {}
+
+  Maybe<webrtc::AudioReceiveStreamInterface::Stats> GetReceiverStats()
+      const override;
+  Maybe<webrtc::AudioSendStream::Stats> GetSenderStats() const override;
+  Maybe<webrtc::CallBasicStats> GetCallStats() const override;
+
+  bool IsSamplingFreqSupported(int freq) const override;
+
+  MediaEventSource<void>& RtcpByeEvent() override { return mRtcpByeEvent; }
+  MediaEventSource<void>& RtcpTimeoutEvent() override {
+    return mRtcpTimeoutEvent;
+  }
+  MediaEventSource<void>& RtpPacketEvent() override { return mRtpPacketEvent; }
+
+  std::vector<webrtc::RtpSource> GetUpstreamRtpSources() const override;
+
+ private:
+  WebrtcAudioConduit(const WebrtcAudioConduit& other) = delete;
+  void operator=(const WebrtcAudioConduit& other) = delete;
+
+  // Generate block size in sample length for a given sampling frequency
+  unsigned int GetNum10msSamplesForFrequency(int samplingFreqHz) const;
+
+  // Checks the codec to be applied
+  static MediaConduitErrorCode ValidateCodecConfig(
+      const AudioCodecConfig& codecInfo, bool send);
+  /**
+   * Of all extensions in aExtensions, returns a list of supported extensions.
+   */
+  static RtpExtList FilterExtensions(
+      MediaSessionConduitLocalDirection aDirection,
+      const RtpExtList& aExtensions);
+  static webrtc::SdpAudioFormat CodecConfigToLibwebrtcFormat(
+      const AudioCodecConfig& aConfig);
+
+  void CreateSendStream();
+  void DeleteSendStream();
+  void CreateRecvStream();
+  void DeleteRecvStream();
+
+  // Are SSRC changes without signaling allowed or not.
+  // Call thread only.
+  bool mAllowSsrcChange = true;
+
+  // Const so can be accessed on any thread. Most methods are called on the Call
+  // thread.
+  const RefPtr<WebrtcCallWrapper> mCall;
+
+  // Set up in the ctor and then not touched. Called through by the streams on
+  // any thread.
+  WebrtcSendTransport mSendTransport;
+  WebrtcReceiveTransport mRecvTransport;
+
+  // Accessed only on the Call thread.
+  webrtc::AudioReceiveStreamInterface::Config mRecvStreamConfig;
+
+  // Written only on the Call thread. Guarded by mLock, except for reads on the
+  // Call thread.
+  webrtc::AudioReceiveStreamInterface* mRecvStream;
+
+  // Accessed only on the Call thread.
+  webrtc::AudioSendStream::Config mSendStreamConfig;
+
+  // Written only on the Call thread. Guarded by mLock, except for reads on the
+  // Call thread.
+  webrtc::AudioSendStream* mSendStream;
+
+  // If true => mSendStream started and not stopped
+  // Written only on the Call thread.
+  Atomic<bool> mSendStreamRunning;
+  // If true => mRecvStream started and not stopped
+  // Written only on the Call thread.
+  Atomic<bool> mRecvStreamRunning;
+
+  // Accessed only on the Call thread.
+  bool mDtmfEnabled;
+
+  mutable RWLock mLock MOZ_UNANNOTATED;
+
+  // Call worker thread. All access to mCall->Call() happens here.
+  const RefPtr<AbstractThread> mCallThread;
+
+  // Socket transport service thread. Any thread.
+  const nsCOMPtr<nsISerialEventTarget> mStsThread;
+
+  // Target jitter buffer to be applied to the receive stream in milliseconds.
+  uint16_t mJitterBufferTargetMs = 0;
+
+  struct Control {
+    // Mirrors and events that map to AudioConduitControlInterface for control.
+    // Call thread only.
+    Mirror<bool> mReceiving;
+    Mirror<bool> mTransmitting;
+    Mirror<Ssrcs> mLocalSsrcs;
+    Mirror<std::string> mLocalCname;
+    Mirror<std::string> mMid;
+    Mirror<Ssrc> mRemoteSsrc;
+    Mirror<std::string> mSyncGroup;
+    Mirror<RtpExtList> mLocalRecvRtpExtensions;
+    Mirror<RtpExtList> mLocalSendRtpExtensions;
+    Mirror<Maybe<AudioCodecConfig>> mSendCodec;
+    Mirror<std::vector<AudioCodecConfig>> mRecvCodecs;
+    MediaEventListener mOnDtmfEventListener;
+
+    // For caching mRemoteSsrc, since another caller may change the remote ssrc
+    // in the stream config directly.
+    Ssrc mConfiguredRemoteSsrc = 0;
+    // For tracking changes to mSendCodec.
+    Maybe<AudioCodecConfig> mConfiguredSendCodec;
+    // For tracking changes to mRecvCodecs.
+    std::vector<AudioCodecConfig> mConfiguredRecvCodecs;
+
+    Control() = delete;
+    explicit Control(const RefPtr<AbstractThread>& aCallThread);
+  } mControl;
+
+  // WatchManager allowing Mirrors to trigger functions that will update the
+  // webrtc.org configuration.
+  WatchManager<WebrtcAudioConduit> mWatchManager;
+
+  // Accessed from mStsThread. Last successfully polled RTT
+  Maybe<DOMHighResTimeStamp> mRttSec;
+
+  // Call thread only. ssrc -> base_seq
+  std::map<uint32_t, uint16_t> mRtpSendBaseSeqs;
+  // libwebrtc network thread only. ssrc -> base_seq.
+  // To track changes needed to mRtpSendBaseSeqs.
+  std::map<uint32_t, uint16_t> mRtpSendBaseSeqs_n;
+
+  // Thread safe
+  Atomic<bool> mTransportActive = Atomic<bool>(false);
+  MediaEventProducer<void> mRtcpByeEvent;
+  MediaEventProducer<void> mRtcpTimeoutEvent;
+  MediaEventProducer<void> mRtpPacketEvent;
+  MediaEventProducerExc<MediaPacket> mSenderRtpSendEvent;
+  MediaEventProducerExc<MediaPacket> mSenderRtcpSendEvent;
+  MediaEventProducerExc<MediaPacket> mReceiverRtcpSendEvent;
+
+  // Assigned and revoked on mStsThread. Listeners for receiving packets.
+  MediaEventListener mSenderRtcpEventListener;    // Rtp-transmitting pipeline
+  MediaEventListener mReceiverRtcpEventListener;  // Rtp-receiving pipeline
+  MediaEventListener mReceiverRtpEventListener;   // Rtp-receiving pipeline
+};
+
+}  // namespace mozilla
+
+#endif