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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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210 test::VideoRenderer::Create("Playback Video", 640, 480)); | 210 test::VideoRenderer::Create("Playback Video", 640, 480)); |
211 FileRenderPassthrough file_passthrough(flags::OutBase(), | 211 FileRenderPassthrough file_passthrough(flags::OutBase(), |
212 playback_video.get()); | 212 playback_video.get()); |
213 | 213 |
214 std::unique_ptr<Call> call(Call::Create(Call::Config())); | 214 std::unique_ptr<Call> call(Call::Create(Call::Config())); |
215 | 215 |
216 test::NullTransport transport; | 216 test::NullTransport transport; |
217 VideoReceiveStream::Config receive_config(&transport); | 217 VideoReceiveStream::Config receive_config(&transport); |
218 receive_config.rtp.remote_ssrc = flags::Ssrc(); | 218 receive_config.rtp.remote_ssrc = flags::Ssrc(); |
219 receive_config.rtp.local_ssrc = kReceiverLocalSsrc; | 219 receive_config.rtp.local_ssrc = kReceiverLocalSsrc; |
220 receive_config.rtp.fec.ulpfec_payload_type = flags::FecPayloadType(); | 220 receive_config.rtp.ulpfec.ulpfec_payload_type = flags::FecPayloadType(); |
221 receive_config.rtp.fec.red_payload_type = flags::RedPayloadType(); | 221 receive_config.rtp.ulpfec.red_payload_type = flags::RedPayloadType(); |
222 receive_config.rtp.nack.rtp_history_ms = 1000; | 222 receive_config.rtp.nack.rtp_history_ms = 1000; |
223 if (flags::TransmissionOffsetId() != -1) { | 223 if (flags::TransmissionOffsetId() != -1) { |
224 receive_config.rtp.extensions.push_back(RtpExtension( | 224 receive_config.rtp.extensions.push_back(RtpExtension( |
225 RtpExtension::kTimestampOffsetUri, flags::TransmissionOffsetId())); | 225 RtpExtension::kTimestampOffsetUri, flags::TransmissionOffsetId())); |
226 } | 226 } |
227 if (flags::AbsSendTimeId() != -1) { | 227 if (flags::AbsSendTimeId() != -1) { |
228 receive_config.rtp.extensions.push_back( | 228 receive_config.rtp.extensions.push_back( |
229 RtpExtension(RtpExtension::kAbsSendTimeUri, flags::AbsSendTimeId())); | 229 RtpExtension(RtpExtension::kAbsSendTimeUri, flags::AbsSendTimeId())); |
230 } | 230 } |
231 receive_config.renderer = &file_passthrough; | 231 receive_config.renderer = &file_passthrough; |
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318 } | 318 } |
319 } // namespace webrtc | 319 } // namespace webrtc |
320 | 320 |
321 int main(int argc, char* argv[]) { | 321 int main(int argc, char* argv[]) { |
322 ::testing::InitGoogleTest(&argc, argv); | 322 ::testing::InitGoogleTest(&argc, argv); |
323 google::ParseCommandLineFlags(&argc, &argv, true); | 323 google::ParseCommandLineFlags(&argc, &argv, true); |
324 | 324 |
325 webrtc::test::RunTest(webrtc::RtpReplay); | 325 webrtc::test::RunTest(webrtc::RtpReplay); |
326 return 0; | 326 return 0; |
327 } | 327 } |
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