Store/restore RTP state for audio streams with same SSRC within a call

webrtc/audio/audio_send_stream.h

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/bitrate_allocator.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h"
 
 namespace webrtc {
 class VoiceEngine;
 class RtcEventLog;
 class RtcpBandwidthObserver;
 class RtcpRttStats;
 class RtpTransportControllerSendInterface;
 
 namespace voe {
 class ChannelProxy;
 }  // namespace voe
 
 namespace internal {
 class AudioSendStream final : public webrtc::AudioSendStream,
                               public webrtc::BitrateAllocatorObserver,
                               public webrtc::PacketFeedbackObserver {
  public:
   AudioSendStream(const webrtc::AudioSendStream::Config& config,
                   const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
                   rtc::TaskQueue* worker_queue,
                   RtpTransportControllerSendInterface* transport,
                   BitrateAllocator* bitrate_allocator,
                   RtcEventLog* event_log,
-                  RtcpRttStats* rtcp_rtt_stats);
+                  RtcpRttStats* rtcp_rtt_stats,
+                  const rtc::Optional<RtpState>& suspended_rtp_state);

[kwiberg-webrtc, 2017/05/19 01:05:09]

const RtpState* ? That way, you don't force the caller to have an Optional.

Or are all callers expected to have an Optional<RtpState> already?

[ossu, 2017/05/22 17:13:59]

I think the Optional better communicates the usage, so I'll keep it for now,

   ~AudioSendStream() override;
 
   // webrtc::AudioSendStream implementation.
   void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
 
   void Start() override;
   void Stop() override;
   bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
                           int duration_ms) override;
   void SetMuted(bool muted) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   void SignalNetworkState(NetworkState state);
   bool DeliverRtcp(const uint8_t* packet, size_t length);
 
   // Implements BitrateAllocatorObserver.
   uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
                             uint8_t fraction_loss,
                             int64_t rtt,
                             int64_t probing_interval_ms) override;
 
   // From PacketFeedbackObserver.
   void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override;
   void OnPacketFeedbackVector(
       const std::vector<PacketFeedback>& packet_feedback_vector) override;
 
   const webrtc::AudioSendStream::Config& config() const;
   void SetTransportOverhead(int transport_overhead_per_packet);
 
+  RtpState GetRtpState() const;
+
  private:
   VoiceEngine* voice_engine() const;
 
   // These are all static to make it less likely that (the old) config_ is
   // accessed unintentionally.
   static void ConfigureStream(AudioSendStream* stream,
                               const Config& new_config,
                               bool first_time);
   static bool SetupSendCodec(AudioSendStream* stream, const Config& new_config);
   static bool ReconfigureSendCodec(AudioSendStream* stream,
@@ skipping 17 matching lines @@
   RtcEventLog* const event_log_;
 
   BitrateAllocator* const bitrate_allocator_;
   RtpTransportControllerSendInterface* const transport_;
   std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
 
   rtc::CriticalSection packet_loss_tracker_cs_;
   TransportFeedbackPacketLossTracker packet_loss_tracker_
       GUARDED_BY(&packet_loss_tracker_cs_);
 
+  RtpRtcp* rtp_rtcp_module_;
+  rtc::Optional<RtpState> const suspended_rtp_state_;
+
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_