Support N unsignaled audio streams

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index f96c8e741e04ad94075ad69add97fab351230e01..d3c168e290d24e05e489013c21d84b320398728a 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -44,6 +44,8 @@
 namespace cricket {
 namespace {
 
+constexpr size_t kMaxUnsignaledRecvStreams = 50;
+
 const int kDefaultTraceFilter = webrtc::kTraceNone | webrtc::kTraceTerseInfo |
                                 webrtc::kTraceWarning | webrtc::kTraceError |
                                 webrtc::kTraceCritical;
@@ -2276,9 +2278,8 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
     return false;
   }
 
-  // Remove the default receive stream if one had been created with this ssrc;
-  // we'll recreate it then.
-  if (IsDefaultRecvStream(ssrc)) {
+  // If this stream was previously received unsignaled, remove it first.
+  if (TryDeregisterUnsignaledRecvStream(ssrc)) {
     RemoveRecvStream(ssrc);
   }
 
@@ -2342,10 +2343,7 @@ bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
     return false;
   }
 
-  // Deregister default channel, if that's the one being destroyed.
-  if (IsDefaultRecvStream(ssrc)) {
-    default_recv_ssrc_ = -1;
-  }
+  TryDeregisterUnsignaledRecvStream(ssrc);
 
   const int channel = it->second->channel();
 
@@ -2405,21 +2403,21 @@ int WebRtcVoiceMediaChannel::GetOutputLevel() {
 
 bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  std::vector<uint32_t> ssrcs(1, ssrc);
   if (ssrc == 0) {
     default_recv_volume_ = volume;
-    if (default_recv_ssrc_ == -1) {
-      return true;
-    }
-    ssrc = static_cast<uint32_t>(default_recv_ssrc_);
+    ssrcs = unsignaled_recv_ssrcs_;
   }
-  const auto it = recv_streams_.find(ssrc);
-  if (it == recv_streams_.end()) {
-    LOG(LS_WARNING) << "SetOutputVolume: no recv stream" << ssrc;
-    return false;
+  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;
+    }
+    it->second->SetOutputVolume(volume);
+    LOG(LS_INFO) << "SetOutputVolume() to " << volume
+                 << " for recv stream with ssrc " << ssrc;
   }
-  it->second->SetOutputVolume(volume);
-  LOG(LS_INFO) << "SetOutputVolume() to " << volume
-               << " for recv stream with ssrc " << ssrc;
   return true;
 }
 
@@ -2470,35 +2468,44 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
     return;
   }
 
-  // Create a default receive stream for this unsignalled and previously not
-  // received ssrc. If there already is a default receive stream, delete it.
+  // Create an unsignaled receive stream for this previously not received ssrc.
+  // If there already is N unsignaled receive streams, delete the oldest.
   // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
   uint32_t ssrc = 0;
   if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
     return;
   }
+  RTC_DCHECK(std::find(unsignaled_recv_ssrcs_.begin(),
+      unsignaled_recv_ssrcs_.end(), ssrc) == unsignaled_recv_ssrcs_.end());
 
+  // Add new stream.
   StreamParams sp;
   sp.ssrcs.push_back(ssrc);
-  LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
+  LOG(LS_INFO) << "Creating unsignaled receive stream for SSRC=" << ssrc;
   if (!AddRecvStream(sp)) {
-    LOG(LS_WARNING) << "Could not create default receive stream.";
+    LOG(LS_WARNING) << "Could not create unsignaled receive stream.";
     return;
   }
-  if (default_recv_ssrc_ != -1) {
-    LOG(LS_INFO) << "Removing default receive stream with ssrc "
-                 << default_recv_ssrc_;
-    RTC_DCHECK_NE(ssrc, default_recv_ssrc_);
-    RemoveRecvStream(default_recv_ssrc_);
+  unsignaled_recv_ssrcs_.push_back(ssrc);
+
+  // Remove oldest unsignaled stream, if we have too many.
+  if (unsignaled_recv_ssrcs_.size() > kMaxUnsignaledRecvStreams) {
+    uint32_t remove_ssrc = unsignaled_recv_ssrcs_.front();
+    LOG(LS_INFO) << "Removing unsignaled receive stream with SSRC="
+                 << remove_ssrc;
+    RemoveRecvStream(remove_ssrc);
   }
-  default_recv_ssrc_ = ssrc;
+  RTC_DCHECK_GE(kMaxUnsignaledRecvStreams, unsignaled_recv_ssrcs_.size());
+
+  SetOutputVolume(ssrc, default_recv_volume_);
 
-  SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
-  if (default_sink_) {
+  // Only set audio sink for first unsignaled stream.

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

Is this ok? And should there be a test for this?

[the sun, 2017/02/17 10:10:56]

A) No. I changed it to adding the default sink to the latest unsignaled stream.
That way we will support endpoints switching SSRC for a (logically) single
unsignaled stream, e.g. webrtc:5208.
B) Now there is.

+  if (default_sink_ && unsignaled_recv_ssrcs_.size() == 1) {
     std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
         new ProxySink(default_sink_.get()));
-    SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
+    SetRawAudioSink(ssrc, std::move(proxy_sink));
   }
+
   delivery_result = call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
                                                      packet->cdata(),
                                                      packet->size(),
@@ -2663,17 +2670,17 @@ void WebRtcVoiceMediaChannel::SetRawAudioSink(
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:" << ssrc
                   << " " << (sink ? "(ptr)" : "NULL");
   if (ssrc == 0) {
-    if (default_recv_ssrc_ != -1) {
+    if (unsignaled_recv_ssrcs_.size() == 1) {
       std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
           sink ? new ProxySink(sink.get()) : nullptr);
-      SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
+      SetRawAudioSink(unsignaled_recv_ssrcs_.front(), std::move(proxy_sink));
     }
     default_sink_ = std::move(sink);
     return;
   }
   const auto it = recv_streams_.find(ssrc);
   if (it == recv_streams_.end()) {
-    LOG(LS_WARNING) << "SetRawAudioSink: no recv stream" << ssrc;
+    LOG(LS_WARNING) << "SetRawAudioSink: no recv stream " << ssrc;
     return;
   }
   it->second->SetRawAudioSink(std::move(sink));
@@ -2702,6 +2709,18 @@ int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const {
   }
   return -1;
 }
+
+bool WebRtcVoiceMediaChannel::TryDeregisterUnsignaledRecvStream(uint32_t ssrc) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  auto it = std::find(unsignaled_recv_ssrcs_.begin(),
+                      unsignaled_recv_ssrcs_.end(),
+                      ssrc);
+  if (it != unsignaled_recv_ssrcs_.end()) {
+    unsignaled_recv_ssrcs_.erase(it);
+    return true;
+  }
+  return false;
+}
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE