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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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