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

webrtc/pc/rtpsender.cc

 /*
  *  Copyright 2015 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 39 matching lines @@
                                StatsCollector* stats)
     : id_(track->id()),
       stream_id_(stream_id),
       channel_(channel),
       stats_(stats),
       track_(track),
       cached_track_enabled_(track->enabled()),
       sink_adapter_(new LocalAudioSinkAdapter()) {
   track_->RegisterObserver(this);
   track_->AddSink(sink_adapter_.get());
+  CreateDtmfSender();
 }
 
 AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
                                cricket::VoiceChannel* channel,
                                StatsCollector* stats)
     : id_(track->id()),
       stream_id_(rtc::CreateRandomUuid()),
       channel_(channel),
       stats_(stats),
       track_(track),
       cached_track_enabled_(track->enabled()),
       sink_adapter_(new LocalAudioSinkAdapter()) {
   track_->RegisterObserver(this);
   track_->AddSink(sink_adapter_.get());
+  CreateDtmfSender();
 }
 
 AudioRtpSender::AudioRtpSender(cricket::VoiceChannel* channel,
                                StatsCollector* stats)
     : id_(rtc::CreateRandomUuid()),
       stream_id_(rtc::CreateRandomUuid()),
       channel_(channel),
       stats_(stats),
-      sink_adapter_(new LocalAudioSinkAdapter()) {}
+      sink_adapter_(new LocalAudioSinkAdapter()) {
+  CreateDtmfSender();
+}
 
 AudioRtpSender::~AudioRtpSender() {
+  // For DtmfSender.
+  SignalDestroyed();
   Stop();
 }
 
+bool AudioRtpSender::CanInsertDtmf() {
+  if (!channel_) {
+    LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
+    return false;
+  }
+  // Check that this RTP sender is active (description has been applied that
+  // matches an SSRC to its ID).
+  if (!ssrc_) {
+    LOG(LS_ERROR) << "CanInsertDtmf: Sender does not have SSRC.";
+    return false;
+  }
+  return channel_->CanInsertDtmf();
+}
+
+bool AudioRtpSender::InsertDtmf(int code, int duration) {
+  if (!channel_) {
+    LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
+    return false;
+  }
+  if (!ssrc_) {
+    LOG(LS_ERROR) << "CanInsertDtmf: Sender does not have SSRC.";
+    return false;
+  }
+  if (!channel_->InsertDtmf(ssrc_, code, duration)) {
+    LOG(LS_ERROR) << "Failed to insert DTMF to channel.";
+    return false;
+  }
+  return true;
+}
+
+sigslot::signal0<>* AudioRtpSender::GetOnDestroyedSignal() {
+  return &SignalDestroyed;
+}
+
 void AudioRtpSender::OnChanged() {
   TRACE_EVENT0("webrtc", "AudioRtpSender::OnChanged");
   RTC_DCHECK(!stopped_);
   if (cached_track_enabled_ != track_->enabled()) {
     cached_track_enabled_ = track_->enabled();
     if (can_send_track()) {
       SetAudioSend();
     }
   }
 }
@@ skipping 53 matching lines @@
 }
 
 bool AudioRtpSender::SetParameters(const RtpParameters& parameters) {
   TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters");
   if (!channel_ || stopped_) {
     return false;
   }
   return channel_->SetRtpSendParameters(ssrc_, parameters);
 }
 
+rtc::scoped_refptr<DtmfSenderInterface> AudioRtpSender::GetDtmfSender() const {
+  return dtmf_sender_proxy_;
+}
+
 void AudioRtpSender::SetSsrc(uint32_t ssrc) {
   TRACE_EVENT0("webrtc", "AudioRtpSender::SetSsrc");
   if (stopped_ || ssrc == ssrc_) {
     return;
   }
   // If we are already sending with a particular SSRC, stop sending.
   if (can_send_track()) {
     ClearAudioSend();
     if (stats_) {
       stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
@@ skipping 59 matching lines @@
   if (!channel_) {
     LOG(LS_WARNING) << "ClearAudioSend: No audio channel exists.";
     return;
   }
   cricket::AudioOptions options;
   if (!channel_->SetAudioSend(ssrc_, false, &options, nullptr)) {
     LOG(LS_WARNING) << "ClearAudioSend: ssrc is incorrect: " << ssrc_;
   }
 }
 
+void AudioRtpSender::CreateDtmfSender() {
+  // Should be on signaling thread.
+  // TODO(deadbeef): Add thread checking to RtpSender/RtpReceiver
+  // implementations.
+  rtc::scoped_refptr<DtmfSenderInterface> sender(
+      DtmfSender::Create(track_, rtc::Thread::Current(), this));
+  if (!sender.get()) {
+    LOG(LS_ERROR) << "CreateDtmfSender failed on DtmfSender::Create.";
+    RTC_NOTREACHED();
+  }
+  dtmf_sender_proxy_ =
+      DtmfSenderProxy::Create(rtc::Thread::Current(), sender.get());
+}
+
 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
                                const std::string& stream_id,
                                cricket::VideoChannel* channel)
     : id_(track->id()),
       stream_id_(stream_id),
       channel_(channel),
       track_(track),
       cached_track_enabled_(track->enabled()),
       cached_track_content_hint_(track->content_hint()) {
   track_->RegisterObserver(this);
@@ skipping 79 matching lines @@
 }
 
 bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
   TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters");
   if (!channel_ || stopped_) {
     return false;
   }
   return channel_->SetRtpSendParameters(ssrc_, parameters);
 }
 
+rtc::scoped_refptr<DtmfSenderInterface> VideoRtpSender::GetDtmfSender() const {
+  LOG(LS_ERROR) << "Tried to get DTMF sender from video sender.";
+  return nullptr;
+}
+
 void VideoRtpSender::SetSsrc(uint32_t ssrc) {
   TRACE_EVENT0("webrtc", "VideoRtpSender::SetSsrc");
   if (stopped_ || ssrc == ssrc_) {
     return;
   }
   // If we are already sending with a particular SSRC, stop sending.
   if (can_send_track()) {
     ClearVideoSend();
   }
   ssrc_ = ssrc;
@@ skipping 51 matching lines @@
     LOG(LS_WARNING) << "SetVideoSend: No video channel exists.";
     return;
   }
   // Allow SetVideoSend to fail since |enable| is false and |source| is null.
   // This the normal case when the underlying media channel has already been
   // deleted.
   channel_->SetVideoSend(ssrc_, false, nullptr, nullptr);
 }
 
 }  // namespace webrtc