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Side by Side Diff: webrtc/api/webrtcsession_unittest.cc

Issue 1766653002: Replace SetCapturer and SetCaptureDevice by SetSource. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Make SetSource tolerate unknown ssrc and source == NULL. Created 4 years, 9 months ago
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1 /* 1 /*
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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236 cricket::TransportChannel* data_rtp_transport_channel() { 236 cricket::TransportChannel* data_rtp_transport_channel() {
237 return rtp_transport_channel(data_channel()); 237 return rtp_transport_channel(data_channel());
238 } 238 }
239 239
240 cricket::TransportChannel* data_rtcp_transport_channel() { 240 cricket::TransportChannel* data_rtcp_transport_channel() {
241 return rtcp_transport_channel(data_channel()); 241 return rtcp_transport_channel(data_channel());
242 } 242 }
243 243
244 using webrtc::WebRtcSession::SetAudioPlayout; 244 using webrtc::WebRtcSession::SetAudioPlayout;
245 using webrtc::WebRtcSession::SetAudioSend; 245 using webrtc::WebRtcSession::SetAudioSend;
246 using webrtc::WebRtcSession::SetCaptureDevice; 246 using webrtc::WebRtcSession::SetSource;
247 using webrtc::WebRtcSession::SetVideoPlayout; 247 using webrtc::WebRtcSession::SetVideoPlayout;
248 using webrtc::WebRtcSession::SetVideoSend; 248 using webrtc::WebRtcSession::SetVideoSend;
249 249
250 private: 250 private:
251 cricket::TransportChannel* rtp_transport_channel(cricket::BaseChannel* ch) { 251 cricket::TransportChannel* rtp_transport_channel(cricket::BaseChannel* ch) {
252 if (!ch) { 252 if (!ch) {
253 return nullptr; 253 return nullptr;
254 } 254 }
255 return ch->transport_channel(); 255 return ch->transport_channel();
256 } 256 }
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4313 } 4313 }
4314 4314
4315 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled. That test 4315 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled. That test
4316 // currently fails because upon disconnection and reconnection OnIceComplete is 4316 // currently fails because upon disconnection and reconnection OnIceComplete is
4317 // called more than once without returning to IceGatheringGathering. 4317 // called more than once without returning to IceGatheringGathering.
4318 4318
4319 INSTANTIATE_TEST_CASE_P(WebRtcSessionTests, 4319 INSTANTIATE_TEST_CASE_P(WebRtcSessionTests,
4320 WebRtcSessionTest, 4320 WebRtcSessionTest,
4321 testing::Values(ALREADY_GENERATED, 4321 testing::Values(ALREADY_GENERATED,
4322 DTLS_IDENTITY_STORE)); 4322 DTLS_IDENTITY_STORE));
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