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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "content/renderer/media/rtc_peer_connection_handler.h" | 5 #include "content/renderer/media/rtc_peer_connection_handler.h" |
6 | 6 |
7 #include <string.h> | 7 #include <string.h> |
8 | 8 |
9 #include <string> | 9 #include <string> |
10 #include <utility> | 10 #include <utility> |
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1357 return new RtcDataChannelHandler(base::ThreadTaskRunnerHandle::Get(), | 1357 return new RtcDataChannelHandler(base::ThreadTaskRunnerHandle::Get(), |
1358 webrtc_channel); | 1358 webrtc_channel); |
1359 } | 1359 } |
1360 | 1360 |
1361 blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender( | 1361 blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender( |
1362 const blink::WebMediaStreamTrack& track) { | 1362 const blink::WebMediaStreamTrack& track) { |
1363 DCHECK(thread_checker_.CalledOnValidThread()); | 1363 DCHECK(thread_checker_.CalledOnValidThread()); |
1364 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createDTMFSender"); | 1364 TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createDTMFSender"); |
1365 DVLOG(1) << "createDTMFSender."; | 1365 DVLOG(1) << "createDTMFSender."; |
1366 | 1366 |
1367 MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::GetTrack(track); | 1367 MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::From(track); |
1368 if (!native_track || !native_track->is_local_track() || | 1368 if (!native_track || !native_track->is_local_track() || |
1369 track.source().type() != blink::WebMediaStreamSource::TypeAudio) { | 1369 track.source().type() != blink::WebMediaStreamSource::TypeAudio) { |
1370 DLOG(ERROR) << "The DTMF sender requires a local audio track."; | 1370 DLOG(ERROR) << "The DTMF sender requires a local audio track."; |
1371 return nullptr; | 1371 return nullptr; |
1372 } | 1372 } |
1373 | 1373 |
1374 scoped_refptr<webrtc::AudioTrackInterface> audio_track = | 1374 scoped_refptr<webrtc::AudioTrackInterface> audio_track = |
1375 native_track->GetAudioAdapter(); | 1375 native_track->GetAudioAdapter(); |
1376 rtc::scoped_refptr<webrtc::DtmfSenderInterface> sender( | 1376 rtc::scoped_refptr<webrtc::DtmfSenderInterface> sender( |
1377 native_peer_connection_->CreateDtmfSender(audio_track.get())); | 1377 native_peer_connection_->CreateDtmfSender(audio_track.get())); |
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1671 } | 1671 } |
1672 | 1672 |
1673 void RTCPeerConnectionHandler::ResetUMAStats() { | 1673 void RTCPeerConnectionHandler::ResetUMAStats() { |
1674 DCHECK(thread_checker_.CalledOnValidThread()); | 1674 DCHECK(thread_checker_.CalledOnValidThread()); |
1675 num_local_candidates_ipv6_ = 0; | 1675 num_local_candidates_ipv6_ = 0; |
1676 num_local_candidates_ipv4_ = 0; | 1676 num_local_candidates_ipv4_ = 0; |
1677 ice_connection_checking_start_ = base::TimeTicks(); | 1677 ice_connection_checking_start_ = base::TimeTicks(); |
1678 memset(ice_state_seen_, 0, sizeof(ice_state_seen_)); | 1678 memset(ice_state_seen_, 0, sizeof(ice_state_seen_)); |
1679 } | 1679 } |
1680 } // namespace content | 1680 } // namespace content |
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