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 171a38868b2963049303ccb57406f61a734b1c77..4f649c740e0a2c13c24cb83b2d4d6ce75e90ec19 100644 |
--- a/content/renderer/media/rtc_peer_connection_handler.cc |
+++ b/content/renderer/media/rtc_peer_connection_handler.cc |
@@ -30,8 +30,9 @@ |
#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_track.h" |
+#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.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" |
@@ -1500,10 +1501,16 @@ blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender( |
return nullptr; |
} |
- scoped_refptr<webrtc::AudioTrackInterface> audio_track = |
- native_track->GetAudioAdapter(); |
+ ProcessedLocalAudioTrack* const rtc_audio_track = |
+ ProcessedLocalAudioTrack::From(native_track); |
+ if (!rtc_audio_track) { |
+ DLOG(ERROR) << "WebRTC features are not available on this audio track."; |
+ return nullptr; |
+ } |
+ |
rtc::scoped_refptr<webrtc::DtmfSenderInterface> sender( |
- native_peer_connection_->CreateDtmfSender(audio_track.get())); |
+ native_peer_connection_->CreateDtmfSender( |
+ rtc_audio_track->adapter().get())); |
if (!sender) { |
DLOG(ERROR) << "Could not create native DTMF sender."; |
return nullptr; |