Fix RtpReceiver.GetParameters when SSRCs aren't signaled.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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.
  */
 
@@ skipping 1737 matching lines @@
 
   // Codecs are handled at the WebRtcVoiceMediaChannel level.
   webrtc::RtpParameters reduced_params = parameters;
   reduced_params.codecs.clear();
   return it->second->SetRtpParameters(reduced_params);
 }
 
 webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters(
     uint32_t ssrc) const {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  auto it = recv_streams_.find(ssrc);
-  if (it == recv_streams_.end()) {
-    LOG(LS_WARNING) << "Attempting to get RTP receive parameters for stream "
-                    << "with ssrc " << ssrc << " which doesn't exist.";
-    return webrtc::RtpParameters();
+  webrtc::RtpParameters rtp_params;
+  // SSRC of 0 represents the default receive stream.
+  if (ssrc == 0) {
+    if (!default_sink_) {

[the sun, 2017/04/18 18:27:20]

Checking !unsignaled_recv_ssrcs_.empty() should be enough, if I'm not mistaken.

[Taylor Brandstetter, 2017/04/20 07:16:22]

The reason I check default_sink_ instead is that I want
"GetRtpReceiveParameters(0)" to return at least the codecs as long as an
"unsignaled stream AudioRtpReceiver" is configured. Which equates to
default_sink_ being non-null at this level.

In other words, "!default_sink_" is the equivalent to "it ==
recv_streams_.end()" for unsignaled streams; it means "not configured to play
out yet". Well, at least in Chrome.

+      LOG(LS_WARNING) << "Attempting to get RTP parameters for the default, "
+                         "unsignaled audio receive stream, but not yet "
+                         "configured to receive such a stream.";
+      return rtp_params;
+    }
+    rtp_params.encodings.emplace_back();
+    if (!unsignaled_recv_ssrcs_.empty()) {
+      rtp_params.encodings[0].ssrc.emplace(unsignaled_recv_ssrcs_.back());
+    }
+  } else {
+    auto it = recv_streams_.find(ssrc);
+    if (it == recv_streams_.end()) {
+      LOG(LS_WARNING) << "Attempting to get RTP receive parameters for stream "
+                      << "with ssrc " << ssrc << " which doesn't exist.";
+      return webrtc::RtpParameters();
+    }
+    rtp_params.encodings.emplace_back();
+    // TODO(deadbeef): Return stream-specific parameters.
+    rtp_params.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc);
   }
 
-  // TODO(deadbeef): Return stream-specific parameters.
-  webrtc::RtpParameters rtp_params = CreateRtpParametersWithOneEncoding();
   for (const AudioCodec& codec : recv_codecs_) {
     rtp_params.codecs.push_back(codec.ToCodecParameters());
   }
-  rtp_params.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc);
   return rtp_params;
 }
 
 bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  auto it = recv_streams_.find(ssrc);
-  if (it == recv_streams_.end()) {
-    LOG(LS_WARNING) << "Attempting to set RTP receive parameters for stream "
-                    << "with ssrc " << ssrc << " which doesn't exist.";
-    return false;
+  // SSRC of 0 represents the default receive stream.
+  if (ssrc == 0) {
+    if (!default_sink_) {
+      LOG(LS_WARNING) << "Attempting to set RTP parameters for the default, "
+                         "unsignaled audio receive stream, but not yet "
+                         "configured to receive such a stream.";
+      return false;
+    }
+  } else {
+    auto it = recv_streams_.find(ssrc);
+    if (it == recv_streams_.end()) {
+      LOG(LS_WARNING) << "Attempting to set RTP receive parameters for stream "
+                      << "with ssrc " << ssrc << " which doesn't exist.";
+      return false;
+    }
   }
 
   webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
   if (current_parameters != parameters) {
     LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
                   << "unsupported.";
     return false;
   }
   return true;
 }
@@ skipping 520 matching lines @@
   int highest = 0;
   for (const auto& ch : recv_streams_) {
     highest = std::max(ch.second->GetOutputLevel(), highest);
   }
   return highest;
 }
 
 bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   std::vector<uint32_t> ssrcs(1, ssrc);
+  // SSRC of 0 represents the default receive stream.
   if (ssrc == 0) {
     default_recv_volume_ = volume;
     ssrcs = unsignaled_recv_ssrcs_;
   }
   for (uint32_t ssrc : ssrcs) {
     const auto it = recv_streams_.find(ssrc);
     if (it == recv_streams_.end()) {
       LOG(LS_WARNING) << "SetOutputVolume: no recv stream " << ssrc;
       return false;
     }
@@ skipping 308 matching lines @@
                       ssrc);
   if (it != unsignaled_recv_ssrcs_.end()) {
     unsignaled_recv_ssrcs_.erase(it);
     return true;
   }
   return false;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE