OLD | NEW |
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 |
(...skipping 190 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
201 RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0); | 201 RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0); |
202 if (num_video_streams > 0) { | 202 if (num_video_streams > 0) { |
203 video_send_config_ = VideoSendStream::Config(send_transport); | 203 video_send_config_ = VideoSendStream::Config(send_transport); |
204 video_send_config_.encoder_settings.encoder = &fake_encoder_; | 204 video_send_config_.encoder_settings.encoder = &fake_encoder_; |
205 video_send_config_.encoder_settings.payload_name = "FAKE"; | 205 video_send_config_.encoder_settings.payload_name = "FAKE"; |
206 video_send_config_.encoder_settings.payload_type = | 206 video_send_config_.encoder_settings.payload_type = |
207 kFakeVideoSendPayloadType; | 207 kFakeVideoSendPayloadType; |
208 video_send_config_.rtp.extensions.push_back( | 208 video_send_config_.rtp.extensions.push_back( |
209 RtpExtension(RtpExtension::kTransportSequenceNumberUri, | 209 RtpExtension(RtpExtension::kTransportSequenceNumberUri, |
210 kTransportSequenceNumberExtensionId)); | 210 kTransportSequenceNumberExtensionId)); |
211 video_send_config_.rtp.extensions.push_back(RtpExtension( | |
212 RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId)); | |
213 FillEncoderConfiguration(num_video_streams, &video_encoder_config_); | 211 FillEncoderConfiguration(num_video_streams, &video_encoder_config_); |
214 | 212 |
215 for (size_t i = 0; i < num_video_streams; ++i) | 213 for (size_t i = 0; i < num_video_streams; ++i) |
216 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); | 214 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); |
217 video_send_config_.rtp.extensions.push_back(RtpExtension( | 215 video_send_config_.rtp.extensions.push_back(RtpExtension( |
218 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); | 216 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); |
219 } | 217 } |
220 | 218 |
221 if (num_audio_streams > 0) { | 219 if (num_audio_streams > 0) { |
222 audio_send_config_ = AudioSendStream::Config(send_transport); | 220 audio_send_config_ = AudioSendStream::Config(send_transport); |
(...skipping 316 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
539 | 537 |
540 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { | 538 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { |
541 } | 539 } |
542 | 540 |
543 bool EndToEndTest::ShouldCreateReceivers() const { | 541 bool EndToEndTest::ShouldCreateReceivers() const { |
544 return true; | 542 return true; |
545 } | 543 } |
546 | 544 |
547 } // namespace test | 545 } // namespace test |
548 } // namespace webrtc | 546 } // namespace webrtc |
OLD | NEW |