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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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195 RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0); | 195 RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0); |
196 if (num_video_streams > 0) { | 196 if (num_video_streams > 0) { |
197 video_send_config_ = VideoSendStream::Config(send_transport); | 197 video_send_config_ = VideoSendStream::Config(send_transport); |
198 video_send_config_.encoder_settings.encoder = &fake_encoder_; | 198 video_send_config_.encoder_settings.encoder = &fake_encoder_; |
199 video_send_config_.encoder_settings.payload_name = "FAKE"; | 199 video_send_config_.encoder_settings.payload_name = "FAKE"; |
200 video_send_config_.encoder_settings.payload_type = | 200 video_send_config_.encoder_settings.payload_type = |
201 kFakeVideoSendPayloadType; | 201 kFakeVideoSendPayloadType; |
202 video_send_config_.rtp.extensions.push_back( | 202 video_send_config_.rtp.extensions.push_back( |
203 RtpExtension(RtpExtension::kTransportSequenceNumberUri, | 203 RtpExtension(RtpExtension::kTransportSequenceNumberUri, |
204 kTransportSequenceNumberExtensionId)); | 204 kTransportSequenceNumberExtensionId)); |
| 205 video_send_config_.rtp.extensions.push_back( |
| 206 RtpExtension(RtpExtension::kVideoContentTypeUri, |
| 207 kVideoContentTypeExtensionId)); |
205 FillEncoderConfiguration(num_video_streams, &video_encoder_config_); | 208 FillEncoderConfiguration(num_video_streams, &video_encoder_config_); |
206 | 209 |
207 for (size_t i = 0; i < num_video_streams; ++i) | 210 for (size_t i = 0; i < num_video_streams; ++i) |
208 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); | 211 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); |
209 video_send_config_.rtp.extensions.push_back(RtpExtension( | 212 video_send_config_.rtp.extensions.push_back(RtpExtension( |
210 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); | 213 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); |
211 } | 214 } |
212 | 215 |
213 if (num_audio_streams > 0) { | 216 if (num_audio_streams > 0) { |
214 audio_send_config_ = AudioSendStream::Config(send_transport); | 217 audio_send_config_ = AudioSendStream::Config(send_transport); |
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498 | 501 |
499 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { | 502 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { |
500 } | 503 } |
501 | 504 |
502 bool EndToEndTest::ShouldCreateReceivers() const { | 505 bool EndToEndTest::ShouldCreateReceivers() const { |
503 return true; | 506 return true; |
504 } | 507 } |
505 | 508 |
506 } // namespace test | 509 } // namespace test |
507 } // namespace webrtc | 510 } // namespace webrtc |
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