Media track ID visibility at BWE level

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 825 matching lines @@
 }
 
 class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     : public AudioSource::Sink {
  public:
   WebRtcAudioSendStream(
       int ch,
       webrtc::AudioTransport* voe_audio_transport,
       uint32_t ssrc,
       const std::string& c_name,
+      const std::string track_id,
       const rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>&
           send_codec_spec,
       const std::vector<webrtc::RtpExtension>& extensions,
       int max_send_bitrate_bps,
       const rtc::Optional<std::string>& audio_network_adaptor_config,
       webrtc::Call* call,
       webrtc::Transport* send_transport,
       const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory)
       : voe_audio_transport_(voe_audio_transport),
         call_(call),
         config_(send_transport),
         send_side_bwe_with_overhead_(
             webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
         max_send_bitrate_bps_(max_send_bitrate_bps),
         rtp_parameters_(CreateRtpParametersWithOneEncoding()) {
     RTC_DCHECK_GE(ch, 0);
     // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
     // RTC_DCHECK(voe_audio_transport);
     RTC_DCHECK(call);
     RTC_DCHECK(encoder_factory);
     config_.rtp.ssrc = ssrc;
     config_.rtp.c_name = c_name;
     config_.voe_channel_id = ch;
     config_.rtp.extensions = extensions;
     config_.audio_network_adaptor_config = audio_network_adaptor_config;
     config_.encoder_factory = encoder_factory;
+    config_.track_id = track_id;
     rtp_parameters_.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc);
 
     if (send_codec_spec) {
       UpdateSendCodecSpec(*send_codec_spec);
     }
 
     stream_ = call_->CreateAudioSendStream(config_);
   }
 
   ~WebRtcAudioSendStream() override {
@@ skipping 981 matching lines @@
   }
 
   // Save the channel to send_streams_, so that RemoveSendStream() can still
   // delete the channel in case failure happens below.
   webrtc::AudioTransport* audio_transport =
       engine()->voe()->base()->audio_transport();
 
   rtc::Optional<std::string> audio_network_adaptor_config =
       GetAudioNetworkAdaptorConfig(options_);
   WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
-      channel, audio_transport, ssrc, sp.cname, send_codec_spec_,
+      channel, audio_transport, ssrc, sp.cname, sp.id, send_codec_spec_,
       send_rtp_extensions_, max_send_bitrate_bps_, audio_network_adaptor_config,
       call_, this, engine()->encoder_factory_);
   send_streams_.insert(std::make_pair(ssrc, stream));
 
   // At this point the stream's local SSRC has been updated. If it is the first
   // send stream, make sure that all the receive streams are updated with the
   // same SSRC in order to send receiver reports.
   if (send_streams_.size() == 1) {
     receiver_reports_ssrc_ = ssrc;
     for (const auto& kv : recv_streams_) {
@@ skipping 481 matching lines @@
                       ssrc);
   if (it != unsignaled_recv_ssrcs_.end()) {
     unsignaled_recv_ssrcs_.erase(it);
     return true;
   }
   return false;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE