commit 749f6604a1d93b326b79cfaf6b90d245f5906860
author Saurav Das <dinosaurav@chromium.org> Wed Dec 04 09:31:36 2019 -0800
committer Commit Bot <commit-bot@chromium.org> Wed Dec 04 23:49:04 2019 +0000
tree b53032588d726bf1147792224cbb3a596d0bacca
parent 503d7237ce78abc163e0d039382162e8b3fc24ac

Enable SSRC 0 in MediaChannel methods

Refactor voice engine and video engine to use default methods instead of
treating 0 as a special value.

Bug: webrtc:8694
Change-Id: I47c211c6e870cdec737d6b0d05df29a9b534a011
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158600
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30010}

media/base/fake_media_engine.h

diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index aa713d0..3df8f85 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -168,6 +168,9 @@
     }
     return webrtc::RtpParameters();
   }
+  virtual webrtc::RtpParameters GetDefaultRtpReceiveParameters() const {
+    return webrtc::RtpParameters();
+  }
 
   bool IsStreamMuted(uint32_t ssrc) const {
     bool ret = muted_streams_.find(ssrc) != muted_streams_.end();
@@ -338,6 +341,8 @@
   bool InsertDtmf(uint32_t ssrc, int event_code, int duration) override;
 
   bool SetOutputVolume(uint32_t ssrc, double volume) override;
+  bool SetDefaultOutputVolume(double volume) override;
+
   bool GetOutputVolume(uint32_t ssrc, double* volume);
 
   bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
@@ -349,6 +354,8 @@
   void SetRawAudioSink(
       uint32_t ssrc,
       std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
+  void SetDefaultRawAudioSink(
+      std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
 
   std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
 
@@ -415,6 +422,8 @@
   bool GetSendCodec(VideoCodec* send_codec) override;
   bool SetSink(uint32_t ssrc,
                rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+  void SetDefaultSink(
+      rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
   bool HasSink(uint32_t ssrc) const;
 
   bool SetSend(bool send) override;