Support N unsignaled audio streams

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 26 matching lines @@
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace cricket {
 namespace {
 
+constexpr size_t kMaxUnsignaledRecvStreams = 50;
+
 const int kDefaultTraceFilter = webrtc::kTraceNone | webrtc::kTraceTerseInfo |
                                 webrtc::kTraceWarning | webrtc::kTraceError |
                                 webrtc::kTraceCritical;
 const int kElevatedTraceFilter = kDefaultTraceFilter | webrtc::kTraceStateInfo |
                                  webrtc::kTraceInfo;
 
 // On Windows Vista and newer, Microsoft introduced the concept of "Default
 // Communications Device". This means that there are two types of default
 // devices (old Wave Audio style default and Default Communications Device).
 //
@@ skipping 2212 matching lines @@
   if (!ValidateStreamParams(sp)) {
     return false;
   }
 
   const uint32_t ssrc = sp.first_ssrc();
   if (ssrc == 0) {
     LOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported.";
     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);
   }
 
   if (GetReceiveChannelId(ssrc) != -1) {
     LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
     return false;
   }
 
   // Create a new channel for receiving audio data.
   const int channel = CreateVoEChannel();
@@ skipping 43 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
 
   const auto it = recv_streams_.find(ssrc);
   if (it == recv_streams_.end()) {
     LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
                     << " which doesn't exist.";
     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();
 
   // Clean up and delete the receive stream+channel.
   LOG(LS_INFO) << "Removing audio receive stream " << ssrc
                << " with VoiceEngine channel #" << channel << ".";
   it->second->SetRawAudioSink(nullptr);
   delete it->second;
   recv_streams_.erase(it);
   return DeleteVoEChannel(channel);
@@ skipping 39 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   int highest = 0;
   for (const auto& ch : recv_streams_) {
     highest = std::max(GetOutputLevel(ch.second->channel()), highest);
   }
   return highest;
 }
 
 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;
+    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;
     }
-    ssrc = static_cast<uint32_t>(default_recv_ssrc_);
+    it->second->SetOutputVolume(volume);
+    LOG(LS_INFO) << "SetOutputVolume() to " << volume
+                 << " for recv stream with ssrc " << ssrc;
   }
-  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;
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::CanInsertDtmf() {
   return dtmf_payload_type_ ? true : false;
 }
 
 bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc, int event,
                                          int duration) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@@ skipping 30 matching lines @@
   const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                               packet_time.not_before);
   webrtc::PacketReceiver::DeliveryStatus delivery_result =
       call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
                                        packet->cdata(), packet->size(),
                                        webrtc_packet_time);
   if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
     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(default_recv_ssrc_, default_recv_volume_);
-  if (default_sink_) {
+  SetOutputVolume(ssrc, default_recv_volume_);
+
+  // 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(),
                                                      webrtc_packet_time);
   RTC_DCHECK_NE(webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC, delivery_result);
 }
 
 void WebRtcVoiceMediaChannel::OnRtcpReceived(
     rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@@ skipping 144 matching lines @@
   return true;
 }
 
 void WebRtcVoiceMediaChannel::SetRawAudioSink(
     uint32_t ssrc,
     std::unique_ptr<webrtc::AudioSinkInterface> sink) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   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));
 }
 
 int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
   unsigned int ulevel = 0;
   int ret = engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
   return (ret == 0) ? static_cast<int>(ulevel) : -1;
 }
 
 int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = recv_streams_.find(ssrc);
   if (it != recv_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 
 int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   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