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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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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192 send_config->rtp.ulpfec.red_payload_type; | 192 send_config->rtp.ulpfec.red_payload_type; |
193 recv_config.rtp.ulpfec.ulpfec_payload_type = | 193 recv_config.rtp.ulpfec.ulpfec_payload_type = |
194 send_config->rtp.ulpfec.ulpfec_payload_type; | 194 send_config->rtp.ulpfec.ulpfec_payload_type; |
195 if (rtx_) { | 195 if (rtx_) { |
196 recv_config.rtp.ulpfec.red_rtx_payload_type = | 196 recv_config.rtp.ulpfec.red_rtx_payload_type = |
197 send_config->rtp.ulpfec.red_rtx_payload_type; | 197 send_config->rtp.ulpfec.red_rtx_payload_type; |
198 } | 198 } |
199 } | 199 } |
200 | 200 |
201 if (rtx_) { | 201 if (rtx_) { |
202 recv_config.rtp.rtx[send_config->encoder_settings.payload_type].ssrc = | 202 recv_config.rtp.rtx_ssrc = video_rtx_ssrcs_[i]; |
203 video_rtx_ssrcs_[i]; | 203 recv_config.rtp |
204 recv_config.rtp.rtx[send_config->encoder_settings.payload_type] | 204 .rtx_payload_types[send_config->encoder_settings.payload_type] = |
205 .payload_type = send_config->rtp.rtx.payload_type; | 205 send_config->rtp.rtx.payload_type; |
206 } | 206 } |
207 ++i; | 207 ++i; |
208 } | 208 } |
209 } | 209 } |
210 | 210 |
211 void RampUpTester::ModifyAudioConfigs( | 211 void RampUpTester::ModifyAudioConfigs( |
212 AudioSendStream::Config* send_config, | 212 AudioSendStream::Config* send_config, |
213 std::vector<AudioReceiveStream::Config>* receive_configs) { | 213 std::vector<AudioReceiveStream::Config>* receive_configs) { |
214 if (num_audio_streams_ == 0) | 214 if (num_audio_streams_ == 0) |
215 return; | 215 return; |
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550 false); | 550 false); |
551 RunBaseTest(&test); | 551 RunBaseTest(&test); |
552 } | 552 } |
553 | 553 |
554 TEST_F(RampUpTest, TransportSequenceNumberSimulcastRedRtx) { | 554 TEST_F(RampUpTest, TransportSequenceNumberSimulcastRedRtx) { |
555 RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumberUri, true, | 555 RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumberUri, true, |
556 true); | 556 true); |
557 RunBaseTest(&test); | 557 RunBaseTest(&test); |
558 } | 558 } |
559 } // namespace webrtc | 559 } // namespace webrtc |
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