 Chromium Code Reviews
 Chromium Code Reviews Issue 2887733002:
  Store/restore RTP state for audio streams with same SSRC within a call  (Closed)
    
  
    Issue 2887733002:
  Store/restore RTP state for audio streams with same SSRC within a call  (Closed) 
  | Index: webrtc/call/BUILD.gn | 
| diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn | 
| index 1f29e618221a4cdae82fa46280c69bfc9b56ea2d..6cf2b3f49366652794492851f8e6f904bbd4ed75 100644 | 
| --- a/webrtc/call/BUILD.gn | 
| +++ b/webrtc/call/BUILD.gn | 
| @@ -88,6 +88,7 @@ if (rtc_include_tests) { | 
| ] | 
| deps = [ | 
| ":call", | 
| + "../api:mock_audio_mixer", | 
| "../base:rtc_base_approved", | 
| "../logging:rtc_event_log_api", | 
| "../modules/audio_device:mock_audio_device", |