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 <[email protected]> Reviewed-by: Fredrik Solenberg <[email protected]> Reviewed-by: Stefan Holmer <[email protected]> Reviewed-by: Taylor Brandstetter <[email protected]> Cr-Commit-Position: refs/heads/master@{#22610}
darkGoldC · Mar 26, 2018 · bb50ce5 · bb50ce5
1 parent 1707168
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Showing 18 changed files with 98 additions and 8 deletions.
diff --git a/api/rtpparameters.cc b/api/rtpparameters.cc
@@ -114,6 +114,9 @@ const char RtpExtension::kVideoTimingUri[] =
     "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
 const int RtpExtension::kVideoTimingDefaultId = 8;
 
+const char RtpExtension::kMidUri[] = "urn:ietf:params:rtp-hdrext:sdes:mid";
+const int RtpExtension::kMidDefaultId = 9;
+
 const char RtpExtension::kEncryptHeaderExtensionsUri[] =
     "urn:ietf:params:rtp-hdrext:encrypt";
 
@@ -122,7 +125,8 @@ const int RtpExtension::kMaxId = 14;
 
 bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
   return uri == webrtc::RtpExtension::kAudioLevelUri ||
-         uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
+         uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
+         uri == webrtc::RtpExtension::kMidUri;
 }
 
 bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
@@ -132,7 +136,8 @@ bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
          uri == webrtc::RtpExtension::kPlayoutDelayUri ||
          uri == webrtc::RtpExtension::kVideoContentTypeUri ||
-         uri == webrtc::RtpExtension::kVideoTimingUri;
+         uri == webrtc::RtpExtension::kVideoTimingUri ||
+         uri == webrtc::RtpExtension::kMidUri;
 }
 
 bool RtpExtension::IsEncryptionSupported(const std::string& uri) {
@@ -149,7 +154,8 @@ bool RtpExtension::IsEncryptionSupported(const std::string& uri) {
          uri == webrtc::RtpExtension::kVideoRotationUri ||
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
          uri == webrtc::RtpExtension::kPlayoutDelayUri ||
-         uri == webrtc::RtpExtension::kVideoContentTypeUri;
+         uri == webrtc::RtpExtension::kVideoContentTypeUri ||
+         uri == webrtc::RtpExtension::kMidUri;
 }
 
 const RtpExtension* RtpExtension::FindHeaderExtensionByUri(

diff --git a/api/rtpparameters.h b/api/rtpparameters.h
@@ -276,6 +276,11 @@ struct RtpExtension {
   static const char kPlayoutDelayUri[];
   static const int kPlayoutDelayDefaultId;
 
+  // Header extension for identifying media section within a transport.
+  // https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-49#section-15
+  static const char kMidUri[];
+  static const int kMidDefaultId;
+
   // Encryption of Header Extensions, see RFC 6904 for details:
   // https://tools.ietf.org/html/rfc6904
   static const char kEncryptHeaderExtensionsUri[];

diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
@@ -184,6 +184,8 @@ AudioSendStream::ExtensionIds AudioSendStream::FindExtensionIds(
       ids.audio_level = extension.id;
     } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       ids.transport_sequence_number = extension.id;
+    } else if (extension.uri == RtpExtension::kMidUri) {
+      ids.mid = extension.id;
     }
   }
   return ids;
@@ -260,6 +262,12 @@ void AudioSendStream::ConfigureStream(
                                                           bandwidth_observer);
   }
 
+  // MID RTP header extension.
+  if ((first_time || new_ids.mid != old_ids.mid) && new_ids.mid != 0 &&
+      !new_config.rtp.mid.empty()) {
+    channel_proxy->SetMid(new_config.rtp.mid, new_ids.mid);
+  }
+
   if (!ReconfigureSendCodec(stream, new_config)) {
     RTC_LOG(LS_ERROR) << "Failed to set up send codec state.";
   }

diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
@@ -158,6 +158,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   struct ExtensionIds {
     int audio_level = 0;
     int transport_sequence_number = 0;
+    int mid = 0;
   };
   static ExtensionIds FindExtensionIds(
       const std::vector<RtpExtension>& extensions);

diff --git a/audio/channel.cc b/audio/channel.cc
@@ -1042,6 +1042,12 @@ int Channel::SetLocalSSRC(unsigned int ssrc) {
   return 0;
 }
 
+void Channel::SetMid(const std::string& mid, int extension_id) {
+  int ret = SetSendRtpHeaderExtension(true, kRtpExtensionMid, extension_id);
+  RTC_DCHECK_EQ(0, ret);
+  _rtpRtcpModule->SetMid(mid);
+}
+
 int Channel::GetRemoteSSRC(unsigned int& ssrc) {
   ssrc = rtp_receiver_->SSRC();
   return 0;

diff --git a/audio/channel.h b/audio/channel.h
@@ -219,6 +219,7 @@ class Channel
 
   // RTP+RTCP
   int SetLocalSSRC(unsigned int ssrc);
+  void SetMid(const std::string& mid, int extension_id);
   int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
   void EnableSendTransportSequenceNumber(int id);
 

diff --git a/audio/channel_proxy.cc b/audio/channel_proxy.cc
@@ -53,6 +53,11 @@ void ChannelProxy::SetLocalSSRC(uint32_t ssrc) {
   RTC_DCHECK_EQ(0, error);
 }
 
+void ChannelProxy::SetMid(const std::string& mid, int extension_id) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  channel_->SetMid(mid, extension_id);
+}
+
 void ChannelProxy::SetRTCP_CNAME(const std::string& c_name) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   // Note: VoERTP_RTCP::SetRTCP_CNAME() accepts a char[256] array.

diff --git a/audio/channel_proxy.h b/audio/channel_proxy.h
@@ -62,6 +62,7 @@ class ChannelProxy : public RtpPacketSinkInterface {
 
   virtual void SetRTCPStatus(bool enable);
   virtual void SetLocalSSRC(uint32_t ssrc);
+  virtual void SetMid(const std::string& mid, int extension_id);
   virtual void SetRTCP_CNAME(const std::string& c_name);
   virtual void SetNACKStatus(bool enable, int max_packets);
   virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);

diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
@@ -74,6 +74,10 @@ class AudioSendStream {
       // Sender SSRC.
       uint32_t ssrc = 0;
 
+      // The value to send in the MID RTP header extension if the extension is
+      // included in the list of extensions.
+      std::string mid;
+
       // RTP header extensions used for the sent stream.
       std::vector<RtpExtension> extensions;
 

diff --git a/call/video_send_stream.h b/call/video_send_stream.h
@@ -142,6 +142,10 @@ class VideoSendStream {
 
       std::vector<uint32_t> ssrcs;
 
+      // The value to send in the MID RTP header extension if the extension is
+      // included in the list of extensions.
+      std::string mid;
+
       // See RtcpMode for description.
       RtcpMode rtcp_mode = RtcpMode::kCompound;
 

diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
@@ -615,11 +615,15 @@ struct RtpSendParameters : RtpParameters<Codec> {
     ost << "codecs: " << VectorToString(this->codecs) << ", ";
     ost << "extensions: " << VectorToString(this->extensions) << ", ";
     ost << "max_bandwidth_bps: " << max_bandwidth_bps << ", ";
+    ost << "mid: " << (mid.empty() ? "<not set>" : mid) << ", ";
     ost << "}";
     return ost.str();
   }
 
   int max_bandwidth_bps = -1;
+  // This is the value to be sent in the MID RTP header extension (if the header
+  // extension in included in the list of extensions).
+  std::string mid;
 };
 
 struct AudioSendParameters : RtpSendParameters<AudioCodec> {

diff --git 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
@@ -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
@@ -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
@@ -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_;

diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -50,6 +50,8 @@ RTPExtensionType StringToRtpExtensionType(const std::string& extension) {
     return kRtpExtensionVideoContentType;
   if (extension == RtpExtension::kVideoTimingUri)
     return kRtpExtensionVideoTiming;
+  if (extension == RtpExtension::kMidUri)
+    return kRtpExtensionMid;
   RTC_NOTREACHED() << "Looking up unsupported RTP extension.";
   return kRtpExtensionNone;
 }

diff --git a/pc/channel.cc b/pc/channel.cc
@@ -1265,6 +1265,7 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
   AudioSendParameters send_params = last_send_params_;
   RtpSendParametersFromMediaDescription(audio, rtp_header_extensions,
       &send_params);
+  send_params.mid = content_name();
 
   bool parameters_applied = media_channel()->SetSendParameters(send_params);
   if (!parameters_applied) {
@@ -1410,6 +1411,7 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
   if (video->conference_mode()) {
     send_params.conference_mode = true;
   }
+  send_params.mid = content_name();
 
   bool parameters_applied = media_channel()->SetSendParameters(send_params);
 

diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc
@@ -147,12 +147,10 @@ std::unique_ptr<FlexfecSender> MaybeCreateFlexfecSender(
   }
 
   RTC_DCHECK_EQ(1U, config.rtp.flexfec.protected_media_ssrcs.size());
-  // TODO(bugs.webrtc.org/4050): Pass down MID value once it is exposed in the
-  // API.
-  return std::unique_ptr<FlexfecSender>(new FlexfecSender(
+  return rtc::MakeUnique<FlexfecSender>(
       config.rtp.flexfec.payload_type, config.rtp.flexfec.ssrc,
       config.rtp.flexfec.protected_media_ssrcs[0], config.rtp.extensions,
-      RTPSender::FecExtensionSizes(), rtp_state, Clock::GetRealTimeClock()));
+      RTPSender::FecExtensionSizes(), rtp_state, Clock::GetRealTimeClock());
 }
 
 bool TransportSeqNumExtensionConfigured(const VideoSendStream::Config& config) {
@@ -822,6 +820,12 @@ VideoSendStreamImpl::VideoSendStreamImpl(
   ConfigureProtection();
   ConfigureSsrcs();
 
+  if (!config_->rtp.mid.empty()) {
+    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
+      rtp_rtcp->SetMid(config_->rtp.mid);
+    }
+  }
+
   // TODO(pbos): Should we set CNAME on all RTP modules?
   rtp_rtcp_modules_.front()->SetCNAME(config_->rtp.c_name.c_str());