Support N unsignaled audio streams

webrtc/media/engine/webrtcvoiceengine.h

Index: webrtc/media/engine/webrtcvoiceengine.h
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index ea48544924ae0ce5c096c05a404f7ff5798319d1..e60395ecb6022e00c7f068bf394784f790a0c29b 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -233,12 +233,10 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   void ChangePlayout(bool playout);
   int CreateVoEChannel();
   bool DeleteVoEChannel(int channel);
-  bool IsDefaultRecvStream(uint32_t ssrc) {
-    return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
-  }
   bool SetMaxSendBitrate(int bps);
   bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
   void SetupRecording();
+  bool TryDeregisterUnsignaledRecvStream(uint32_t ssrc);

[Taylor Brandstetter, 2017/02/17 08:36:14]

nit: "Try" sounds like something that might fail. I'd use "Maybe" instead. Also,
a brief comment above its declaration would be good.

[the sun, 2017/02/17 10:10:57]

"Maybe" is fine with me.

 
   rtc::ThreadChecker worker_thread_checker_;
 
@@ -257,11 +255,12 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   webrtc::Call* const call_ = nullptr;
   webrtc::Call::Config::BitrateConfig bitrate_config_;
 
-  // SSRC of unsignalled receive stream, or -1 if there isn't one.
-  int64_t default_recv_ssrc_ = -1;
-  // Volume for unsignalled stream, which may be set before the stream exists.
+  // Queue of unsignaled SSRCs; oldest at the beginning.
+  std::vector<uint32_t> unsignaled_recv_ssrcs_;
+
+  // Volume for unsignaled stream, which may be set before the stream exists.

[Taylor Brandstetter, 2017/02/17 08:36:14]

nit: "stream(s)" now?

[the sun, 2017/02/17 10:10:57]

Done.

   double default_recv_volume_ = 1.0;
-  // Sink for unsignalled stream, which may be set before the stream exists.
+  // Sink for unsignaled stream, which may be set before the stream exists.
   std::unique_ptr<webrtc::AudioSinkInterface> default_sink_;
   // Default SSRC to use for RTCP receiver reports in case of no signaled
   // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740