Move DTMF sender to RtpSender (as opposed to WebRtcSession).

webrtc/pc/webrtcsession.cc

 /*
  *  Copyright 2012 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 213 matching lines @@
     }
     if (tinfo->description.ice_ufrag.empty() ||
         tinfo->description.ice_pwd.empty()) {
       LOG(LS_ERROR) << "Session description must have ice ufrag and pwd.";
       return false;
     }
   }
   return true;
 }
 
-static bool GetAudioSsrcByTrackId(const SessionDescription* session_description,
-                                  const std::string& track_id,
-                                  uint32_t* ssrc) {
-  const cricket::ContentInfo* audio_info =
-      cricket::GetFirstAudioContent(session_description);
-  if (!audio_info) {
-    LOG(LS_ERROR) << "Audio not used in this call";
-    return false;
-  }
-
-  const cricket::MediaContentDescription* audio_content =
-      static_cast<const cricket::MediaContentDescription*>(
-          audio_info->description);
-  const cricket::StreamParams* stream =
-      cricket::GetStreamByIds(audio_content->streams(), "", track_id);
-  if (!stream) {
-    return false;
-  }
-
-  *ssrc = stream->first_ssrc();
-  return true;
-}
-
 static bool GetTrackIdBySsrc(const SessionDescription* session_description,
                              uint32_t ssrc,
                              std::string* track_id) {
   RTC_DCHECK(track_id != NULL);
 
   const cricket::ContentInfo* audio_info =
       cricket::GetFirstAudioContent(session_description);
   if (audio_info) {
     const cricket::MediaContentDescription* audio_content =
         static_cast<const cricket::MediaContentDescription*>(
@@ skipping 249 matching lines @@
   if (sctp_transport_) {
     SignalDataChannelDestroyed();
     network_thread_->Invoke<void>(
         RTC_FROM_HERE, rtc::Bind(&WebRtcSession::DestroySctpTransport_n, this));
   }
 #ifdef HAVE_QUIC
   if (quic_data_transport_) {
     quic_data_transport_.reset();
   }
 #endif
-  SignalDestroyed();
 
   LOG(LS_INFO) << "Session: " << id() << " is destroyed.";
 }
 
 bool WebRtcSession::Initialize(
     const PeerConnectionFactoryInterface::Options& options,
     std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
     const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
   bundle_policy_ = rtc_configuration.bundle_policy;
   rtcp_mux_policy_ = rtc_configuration.rtcp_mux_policy;
@@ skipping 704 matching lines @@
   return webrtc::GetTrackIdBySsrc(remote_description()->description(), ssrc,
                                   track_id);
 }
 
 std::string WebRtcSession::BadStateErrMsg(State state) {
   std::ostringstream desc;
   desc << "Called in wrong state: " << GetStateString(state);
   return desc.str();
 }
 
-bool WebRtcSession::CanInsertDtmf(const std::string& track_id) {
-  RTC_DCHECK(signaling_thread()->IsCurrent());
-  if (!voice_channel_) {
-    LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
-    return false;
-  }
-  uint32_t send_ssrc = 0;
-  // The Dtmf is negotiated per channel not ssrc, so we only check if the ssrc
-  // exists.
-  if (!local_description() ||
-      !GetAudioSsrcByTrackId(local_description()->description(), track_id,
-                             &send_ssrc)) {
-    LOG(LS_ERROR) << "CanInsertDtmf: Track does not exist: " << track_id;
-    return false;
-  }
-  return voice_channel_->CanInsertDtmf();
-}
-
-bool WebRtcSession::InsertDtmf(const std::string& track_id,
-                               int code, int duration) {
-  RTC_DCHECK(signaling_thread()->IsCurrent());
-  if (!voice_channel_) {
-    LOG(LS_ERROR) << "InsertDtmf: No audio channel exists.";
-    return false;
-  }
-  uint32_t send_ssrc = 0;
-  if (!(local_description() &&
-        GetAudioSsrcByTrackId(local_description()->description(),
-                              track_id, &send_ssrc))) {
-    LOG(LS_ERROR) << "InsertDtmf: Track does not exist: " << track_id;
-    return false;
-  }
-  if (!voice_channel_->InsertDtmf(send_ssrc, code, duration)) {
-    LOG(LS_ERROR) << "Failed to insert DTMF to channel.";
-    return false;
-  }
-  return true;
-}
-
-sigslot::signal0<>* WebRtcSession::GetOnDestroyedSignal() {
-  return &SignalDestroyed;
-}
-
 bool WebRtcSession::SendData(const cricket::SendDataParams& params,
                              const rtc::CopyOnWriteBuffer& payload,
                              cricket::SendDataResult* result) {
   if (!rtp_data_channel_ && !sctp_transport_) {
     LOG(LS_ERROR) << "SendData called when rtp_data_channel_ "
                   << "and sctp_transport_ are NULL.";
     return false;
   }
   return rtp_data_channel_
              ? rtp_data_channel_->SendData(params, payload, result)
@@ skipping 1121 matching lines @@
       (rtp_data_channel_->rtcp_dtls_transport() != nullptr);
   channel_manager_->DestroyRtpDataChannel(rtp_data_channel_.release());
   transport_controller_->DestroyDtlsTransport(
       transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
   if (need_to_delete_rtcp) {
     transport_controller_->DestroyDtlsTransport(
         transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
   }
 }
 }  // namespace webrtc