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Issue 1273363005: Add send transports to individual webrtc::Call streams. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase+comment Created 5 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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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207 private: 207 private:
208 FILE* file_; 208 FILE* file_;
209 }; 209 };
210 210
211 void RtpReplay() { 211 void RtpReplay() {
212 rtc::scoped_ptr<test::VideoRenderer> playback_video( 212 rtc::scoped_ptr<test::VideoRenderer> playback_video(
213 test::VideoRenderer::Create("Playback Video", 640, 480)); 213 test::VideoRenderer::Create("Playback Video", 640, 480));
214 FileRenderPassthrough file_passthrough(flags::OutBase(), 214 FileRenderPassthrough file_passthrough(flags::OutBase(),
215 playback_video.get()); 215 playback_video.get());
216 216
217 // TODO(pbos): Might be good to have a transport that prints keyframe requests 217 rtc::scoped_ptr<Call> call(Call::Create(Call::Config()));
218 // etc. 218
219 test::NullTransport transport; 219 test::NullTransport transport;
220 Call::Config call_config(&transport); 220 VideoReceiveStream::Config receive_config(&transport);
221 rtc::scoped_ptr<Call> call(Call::Create(call_config));
222
223 VideoReceiveStream::Config receive_config;
224 receive_config.rtp.remote_ssrc = flags::Ssrc(); 221 receive_config.rtp.remote_ssrc = flags::Ssrc();
225 receive_config.rtp.local_ssrc = kReceiverLocalSsrc; 222 receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
226 receive_config.rtp.fec.ulpfec_payload_type = flags::FecPayloadType(); 223 receive_config.rtp.fec.ulpfec_payload_type = flags::FecPayloadType();
227 receive_config.rtp.fec.red_payload_type = flags::RedPayloadType(); 224 receive_config.rtp.fec.red_payload_type = flags::RedPayloadType();
228 receive_config.rtp.nack.rtp_history_ms = 1000; 225 receive_config.rtp.nack.rtp_history_ms = 1000;
229 if (flags::TransmissionOffsetId() != -1) { 226 if (flags::TransmissionOffsetId() != -1) {
230 receive_config.rtp.extensions.push_back( 227 receive_config.rtp.extensions.push_back(
231 RtpExtension(RtpExtension::kTOffset, flags::TransmissionOffsetId())); 228 RtpExtension(RtpExtension::kTOffset, flags::TransmissionOffsetId()));
232 } 229 }
233 if (flags::AbsSendTimeId() != -1) { 230 if (flags::AbsSendTimeId() != -1) {
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324 } 321 }
325 } // namespace webrtc 322 } // namespace webrtc
326 323
327 int main(int argc, char* argv[]) { 324 int main(int argc, char* argv[]) {
328 ::testing::InitGoogleTest(&argc, argv); 325 ::testing::InitGoogleTest(&argc, argv);
329 google::ParseCommandLineFlags(&argc, &argv, true); 326 google::ParseCommandLineFlags(&argc, &argv, true);
330 327
331 webrtc::test::RunTest(webrtc::RtpReplay); 328 webrtc::test::RunTest(webrtc::RtpReplay);
332 return 0; 329 return 0;
333 } 330 }
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