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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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| 249 for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) { | 249 for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) { |
| 250 VideoReceiveStream::Decoder decoder = | 250 VideoReceiveStream::Decoder decoder = |
| 251 test::CreateMatchingDecoder(video_send_config_.encoder_settings); | 251 test::CreateMatchingDecoder(video_send_config_.encoder_settings); |
| 252 allocated_decoders_.push_back( | 252 allocated_decoders_.push_back( |
| 253 std::unique_ptr<VideoDecoder>(decoder.decoder)); | 253 std::unique_ptr<VideoDecoder>(decoder.decoder)); |
| 254 video_config.decoders.clear(); | 254 video_config.decoders.clear(); |
| 255 video_config.decoders.push_back(decoder); | 255 video_config.decoders.push_back(decoder); |
| 256 video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i]; | 256 video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i]; |
| 257 video_receive_configs_.push_back(video_config.Copy()); | 257 video_receive_configs_.push_back(video_config.Copy()); |
| 258 } | 258 } |
| 259 video_receive_configs_[0].rtp.protected_by_flexfec = |
| 260 (num_flexfec_streams_ == 1); |
| 259 } | 261 } |
| 260 | 262 |
| 261 RTC_DCHECK_GE(1, num_audio_streams_); | 263 RTC_DCHECK_GE(1, num_audio_streams_); |
| 262 if (num_audio_streams_ == 1) { | 264 if (num_audio_streams_ == 1) { |
| 263 RTC_DCHECK_LE(0, voe_send_.channel_id); | 265 RTC_DCHECK_LE(0, voe_send_.channel_id); |
| 264 AudioReceiveStream::Config audio_config; | 266 AudioReceiveStream::Config audio_config; |
| 265 audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; | 267 audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; |
| 266 audio_config.rtcp_send_transport = rtcp_send_transport; | 268 audio_config.rtcp_send_transport = rtcp_send_transport; |
| 267 audio_config.voe_channel_id = voe_recv_.channel_id; | 269 audio_config.voe_channel_id = voe_recv_.channel_id; |
| 268 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc; | 270 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc; |
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| 603 | 605 |
| 604 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { | 606 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { |
| 605 } | 607 } |
| 606 | 608 |
| 607 bool EndToEndTest::ShouldCreateReceivers() const { | 609 bool EndToEndTest::ShouldCreateReceivers() const { |
| 608 return true; | 610 return true; |
| 609 } | 611 } |
| 610 | 612 |
| 611 } // namespace test | 613 } // namespace test |
| 612 } // namespace webrtc | 614 } // namespace webrtc |
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