Wire up MID send value to the PeerConnection API

Bug: webrtc:4050
Change-Id: I522cf8621e2cb639f54be2402174befd23e4af59
Reviewed-on: https://webrtc-review.googlesource.com/60962
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22610}

4 files changed, 38 insertions, 1 deletions

diff --git a/media/engine/webrtcvideoengine.cc b/media/engine/webrtcvideoengine.cc
index 6442c44337..df0b33d322 100644
--- a/media/engine/webrtcvideoengine.cc
+++ b/media/engine/webrtcvideoengine.cc
@@ -596,6 +596,10 @@ RtpCapabilities WebRtcVideoEngine::GetCapabilities() const {
   capabilities.header_extensions.push_back(
         webrtc::RtpExtension(webrtc::RtpExtension::kVideoTimingUri,
                              webrtc::RtpExtension::kVideoTimingDefaultId));
+  // TODO(bugs.webrtc.org/4050): Add MID header extension as capability once MID
+  // demuxing is completed.
+  // capabilities.header_extensions.push_back(webrtc::RtpExtension(
+  //     webrtc::RtpExtension::kMidUri, webrtc::RtpExtension::kMidDefaultId));
   return capabilities;
 }
 
@@ -715,6 +719,10 @@ bool WebRtcVideoChannel::GetChangedSendParameters(
         rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
   }
 
+  if (params.mid != send_params_.mid) {
+    changed_params->mid = params.mid;
+  }
+
   // Handle max bitrate.
   if (params.max_bandwidth_bps != send_params_.max_bandwidth_bps &&
       params.max_bandwidth_bps >= -1) {
@@ -1632,6 +1640,8 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
   parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
                                          ? webrtc::RtcpMode::kReducedSize
                                          : webrtc::RtcpMode::kCompound;
+  parameters_.config.rtp.mid = send_params.mid;
+
   if (codec_settings) {
     SetCodec(*codec_settings);
   }
@@ -1776,6 +1786,10 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::SetSendParameters(
     parameters_.config.rtp.extensions = *params.rtp_header_extensions;
     recreate_stream = true;
   }
+  if (params.mid) {
+    parameters_.config.rtp.mid = *params.mid;
+    recreate_stream = true;
+  }
   if (params.max_bandwidth_bps) {
     parameters_.max_bitrate_bps = *params.max_bandwidth_bps;
     ReconfigureEncoder();
diff --git a/media/engine/webrtcvideoengine.h b/media/engine/webrtcvideoengine.h
index 04096e66c7..07e6c70e93 100644
--- a/media/engine/webrtcvideoengine.h
+++ b/media/engine/webrtcvideoengine.h
@@ -213,6 +213,7 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
     // These optionals are unset if not changed.
     rtc::Optional<VideoCodecSettings> codec;
     rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
+    rtc::Optional<std::string> mid;
     rtc::Optional<int> max_bandwidth_bps;
     rtc::Optional<bool> conference_mode;
     rtc::Optional<webrtc::RtcpMode> rtcp_mode;
diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
index 04f299779b..d6ec44b58b 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -600,6 +600,10 @@ RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
         webrtc::RtpExtension::kTransportSequenceNumberUri,
         webrtc::RtpExtension::kTransportSequenceNumberDefaultId));
   }
+  // TODO(bugs.webrtc.org/4050): Add MID header extension as capability once MID
+  // demuxing is completed.
+  // capabilities.header_extensions.push_back(webrtc::RtpExtension(
+  //     webrtc::RtpExtension::kMidUri, webrtc::RtpExtension::kMidDefaultId));
   return capabilities;
 }
 
@@ -742,6 +746,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
  public:
   WebRtcAudioSendStream(
       uint32_t ssrc,
+      const std::string& mid,
       const std::string& c_name,
       const std::string track_id,
       const rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>&
@@ -762,6 +767,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     RTC_DCHECK(call);
     RTC_DCHECK(encoder_factory);
     config_.rtp.ssrc = ssrc;
+    config_.rtp.mid = mid;
     config_.rtp.c_name = c_name;
     config_.rtp.extensions = extensions;
     config_.audio_network_adaptor_config = audio_network_adaptor_config;
@@ -795,6 +801,15 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     ReconfigureAudioSendStream();
   }
 
+  void SetMid(const std::string& mid) {
+    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+    if (config_.rtp.mid == mid) {
+      return;
+    }
+    config_.rtp.mid = mid;
+    ReconfigureAudioSendStream();
+  }
+
   void SetAudioNetworkAdaptorConfig(
       const rtc::Optional<std::string>& audio_network_adaptor_config) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@@ -1302,6 +1317,12 @@ bool WebRtcVoiceMediaChannel::SetSendParameters(
       it.second->SetRtpExtensions(send_rtp_extensions_);
     }
   }
+  if (!params.mid.empty()) {
+    mid_ = params.mid;
+    for (auto& it : send_streams_) {
+      it.second->SetMid(params.mid);
+    }
+  }
 
   if (!SetMaxSendBitrate(params.max_bandwidth_bps)) {
     return false;
@@ -1768,7 +1789,7 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
   rtc::Optional<std::string> audio_network_adaptor_config =
       GetAudioNetworkAdaptorConfig(options_);
   WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
-      ssrc, sp.cname, sp.id, send_codec_spec_, send_rtp_extensions_,
+      ssrc, mid_, sp.cname, sp.id, send_codec_spec_, send_rtp_extensions_,
       max_send_bitrate_bps_, audio_network_adaptor_config, call_, this,
       engine()->encoder_factory_, codec_pair_id_);
   send_streams_.insert(std::make_pair(ssrc, stream));
diff --git a/media/engine/webrtcvoiceengine.h b/media/engine/webrtcvoiceengine.h
index 950b72b5f5..c66aa95a38 100644
--- a/media/engine/webrtcvoiceengine.h
+++ b/media/engine/webrtcvoiceengine.h
@@ -264,6 +264,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   class WebRtcAudioSendStream;
   std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
   std::vector<webrtc::RtpExtension> send_rtp_extensions_;
+  std::string mid_;
 
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;