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| 1 /* | 1 /* |
| 2 * libjingle | 2 * libjingle |
| 3 * Copyright 2012 Google Inc. | 3 * Copyright 2012 Google Inc. |
| 4 * | 4 * |
| 5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
| 6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
| 7 * | 7 * |
| 8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
| 9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
| 10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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| 32 | 32 |
| 33 #include "talk/app/webrtc/audiotrack.h" | 33 #include "talk/app/webrtc/audiotrack.h" |
| 34 #include "talk/app/webrtc/dtmfsender.h" | 34 #include "talk/app/webrtc/dtmfsender.h" |
| 35 #include "talk/app/webrtc/jsepicecandidate.h" | 35 #include "talk/app/webrtc/jsepicecandidate.h" |
| 36 #include "talk/app/webrtc/jsepsessiondescription.h" | 36 #include "talk/app/webrtc/jsepsessiondescription.h" |
| 37 #include "talk/app/webrtc/mediaconstraintsinterface.h" | 37 #include "talk/app/webrtc/mediaconstraintsinterface.h" |
| 38 #include "talk/app/webrtc/mediastream.h" | 38 #include "talk/app/webrtc/mediastream.h" |
| 39 #include "talk/app/webrtc/mediastreamproxy.h" | 39 #include "talk/app/webrtc/mediastreamproxy.h" |
| 40 #include "talk/app/webrtc/mediastreamtrackproxy.h" | 40 #include "talk/app/webrtc/mediastreamtrackproxy.h" |
| 41 #include "talk/app/webrtc/remoteaudiosource.h" | 41 #include "talk/app/webrtc/remoteaudiosource.h" |
| 42 #include "talk/app/webrtc/remoteaudiotrack.h" | |
| 43 #include "talk/app/webrtc/remotevideocapturer.h" | 42 #include "talk/app/webrtc/remotevideocapturer.h" |
| 44 #include "talk/app/webrtc/rtpreceiver.h" | 43 #include "talk/app/webrtc/rtpreceiver.h" |
| 45 #include "talk/app/webrtc/rtpsender.h" | 44 #include "talk/app/webrtc/rtpsender.h" |
| 46 #include "talk/app/webrtc/streamcollection.h" | 45 #include "talk/app/webrtc/streamcollection.h" |
| 47 #include "talk/app/webrtc/videosource.h" | 46 #include "talk/app/webrtc/videosource.h" |
| 48 #include "talk/app/webrtc/videotrack.h" | 47 #include "talk/app/webrtc/videotrack.h" |
| 49 #include "talk/media/sctp/sctpdataengine.h" | 48 #include "talk/media/sctp/sctpdataengine.h" |
| 50 #include "talk/session/media/channelmanager.h" | 49 #include "talk/session/media/channelmanager.h" |
| 51 #include "webrtc/base/arraysize.h" | 50 #include "webrtc/base/arraysize.h" |
| 52 #include "webrtc/base/logging.h" | 51 #include "webrtc/base/logging.h" |
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| 446 rtc::scoped_refptr<MediaStreamInterface> CreateMediaStream( | 445 rtc::scoped_refptr<MediaStreamInterface> CreateMediaStream( |
| 447 const std::string& stream_label) { | 446 const std::string& stream_label) { |
| 448 return MediaStreamProxy::Create(signaling_thread_, | 447 return MediaStreamProxy::Create(signaling_thread_, |
| 449 MediaStream::Create(stream_label)); | 448 MediaStream::Create(stream_label)); |
| 450 } | 449 } |
| 451 | 450 |
| 452 AudioTrackInterface* AddAudioTrack(uint32_t ssrc, | 451 AudioTrackInterface* AddAudioTrack(uint32_t ssrc, |
| 453 AudioProviderInterface* provider, | 452 AudioProviderInterface* provider, |
| 454 webrtc::MediaStreamInterface* stream, | 453 webrtc::MediaStreamInterface* stream, |
| 455 const std::string& track_id) { | 454 const std::string& track_id) { |
| 456 return AddTrack<AudioTrackInterface, RemoteAudioTrack, AudioTrackProxy>( | 455 return AddTrack<AudioTrackInterface, AudioTrack, AudioTrackProxy>( |
| 457 stream, track_id, RemoteAudioSource::Create(ssrc, provider)); | 456 stream, track_id, RemoteAudioSource::Create(ssrc, provider)); |
| 458 } | 457 } |
| 459 | 458 |
| 460 VideoTrackInterface* AddVideoTrack(webrtc::MediaStreamInterface* stream, | 459 VideoTrackInterface* AddVideoTrack(webrtc::MediaStreamInterface* stream, |
| 461 const std::string& track_id) { | 460 const std::string& track_id) { |
| 462 return AddTrack<VideoTrackInterface, VideoTrack, VideoTrackProxy>( | 461 return AddTrack<VideoTrackInterface, VideoTrack, VideoTrackProxy>( |
| 463 stream, track_id, | 462 stream, track_id, |
| 464 VideoSource::Create(channel_manager_, new RemoteVideoCapturer(), | 463 VideoSource::Create(channel_manager_, new RemoteVideoCapturer(), |
| 465 nullptr) | 464 nullptr, true) |
| 466 .get()); | 465 .get()); |
| 467 } | 466 } |
| 468 | 467 |
| 469 private: | 468 private: |
| 470 template <typename TI, typename T, typename TP, typename S> | 469 template <typename TI, typename T, typename TP, typename S> |
| 471 TI* AddTrack(MediaStreamInterface* stream, | 470 TI* AddTrack(MediaStreamInterface* stream, |
| 472 const std::string& track_id, | 471 const std::string& track_id, |
| 473 const S& source) { | 472 const S& source) { |
| 474 rtc::scoped_refptr<TI> track( | 473 rtc::scoped_refptr<TI> track( |
| 475 TP::Create(signaling_thread_, T::Create(track_id, source))); | 474 TP::Create(signaling_thread_, T::Create(track_id, source))); |
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| 2020 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const { | 2019 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const { |
| 2021 for (const auto& channel : sctp_data_channels_) { | 2020 for (const auto& channel : sctp_data_channels_) { |
| 2022 if (channel->id() == sid) { | 2021 if (channel->id() == sid) { |
| 2023 return channel; | 2022 return channel; |
| 2024 } | 2023 } |
| 2025 } | 2024 } |
| 2026 return nullptr; | 2025 return nullptr; |
| 2027 } | 2026 } |
| 2028 | 2027 |
| 2029 } // namespace webrtc | 2028 } // namespace webrtc |
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