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

webrtc/audio/audio_send_stream.h

Index: webrtc/audio/audio_send_stream.h
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index dcc28c9dbcb5c807912fe0ec3892bbfbdff57207..964416947bd62138a82ef6e55dd569040b5ff700 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -19,7 +19,7 @@
 #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 {
@@ -44,7 +44,8 @@ class AudioSendStream final : public webrtc::AudioSendStream,
                   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.
@@ -74,6 +75,8 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   const webrtc::AudioSendStream::Config& config() const;
   void SetTransportOverhead(int transport_overhead_per_packet);
 
+  RtpState GetRtpState() const;
+
  private:
   VoiceEngine* voice_engine() const;
 
@@ -111,6 +114,9 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   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