Index: content/renderer/media/rtc_peer_connection_handler.cc |
diff --git a/content/renderer/media/rtc_peer_connection_handler.cc b/content/renderer/media/rtc_peer_connection_handler.cc |
index 6b12cd114bd4f93f728e432140268deb66da2f77..7b147c01b4d94d24214023b79ba848d9a3b41164 100644 |
--- a/content/renderer/media/rtc_peer_connection_handler.cc |
+++ b/content/renderer/media/rtc_peer_connection_handler.cc |
@@ -31,8 +31,9 @@ |
#include "content/renderer/media/rtc_data_channel_handler.h" |
#include "content/renderer/media/rtc_dtmf_sender_handler.h" |
#include "content/renderer/media/webrtc/peer_connection_dependency_factory.h" |
+#include "content/renderer/media/webrtc/processed_local_audio_source.h" |
+#include "content/renderer/media/webrtc/webrtc_audio_sink.h" |
#include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h" |
-#include "content/renderer/media/webrtc_audio_capturer.h" |
#include "content/renderer/media/webrtc_audio_device_impl.h" |
#include "content/renderer/media/webrtc_uma_histograms.h" |
#include "content/renderer/render_thread_impl.h" |
@@ -1486,20 +1487,32 @@ blink::WebRTCDataChannelHandler* RTCPeerConnectionHandler::createDataChannel( |
blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender( |
const blink::WebMediaStreamTrack& track) { |
DCHECK(thread_checker_.CalledOnValidThread()); |
+ DCHECK(!track.isNull()); |
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createDTMFSender"); |
DVLOG(1) << "createDTMFSender."; |
- MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::From(track); |
- if (!native_track || !native_track->is_local_track() || |
- track.source().getType() != blink::WebMediaStreamSource::TypeAudio) { |
- DLOG(ERROR) << "The DTMF sender requires a local audio track."; |
+ ProcessedLocalAudioSource* const rtc_audio_source = |
+ ProcessedLocalAudioSource::From( |
+ MediaStreamAudioSource::From(track.source())); |
+ if (!rtc_audio_source) { |
+ DLOG(ERROR) << "WebRTC features are not available on this audio track."; |
return nullptr; |
} |
- scoped_refptr<webrtc::AudioTrackInterface> audio_track = |
- native_track->GetAudioAdapter(); |
+ // HACK: Create a temporary WebRtcAudioSink that can provide an instance of |
+ // webrtc::AudioTrackInterface to the DtmfSender, as the interface requires. |
+ // |
+ // TODO(miu): The implementation only needs the track.id() string. Thus, the |
+ // interface declaring the CreateDtmfSender method should be changed to only |
+ // only take the track id as an argument here. Then, we can get rid of |
+ // |dummy_sink|. |
+ const std::unique_ptr<WebRtcAudioSink> dummy_sink(new WebRtcAudioSink( |
perkj_chrome
2016/04/20 13:34:54
Can you instead find the correct webrtc audio trac
miu
2016/04/20 22:04:53
Done. Yes! This is what I was looking for. :)
|
+ track.id().utf8(), rtc_audio_source->rtc_source(), |
+ dependency_factory_->GetWebRtcSignalingThread())); |
+ |
rtc::scoped_refptr<webrtc::DtmfSenderInterface> sender( |
- native_peer_connection_->CreateDtmfSender(audio_track.get())); |
+ native_peer_connection_->CreateDtmfSender( |
+ dummy_sink->webrtc_audio_track())); |
if (!sender) { |
DLOG(ERROR) << "Could not create native DTMF sender."; |
return nullptr; |