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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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235 RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0); | 235 RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0); |
236 if (num_video_streams > 0) { | 236 if (num_video_streams > 0) { |
237 video_send_config_ = VideoSendStream::Config(send_transport); | 237 video_send_config_ = VideoSendStream::Config(send_transport); |
238 video_send_config_.encoder_settings.encoder = &fake_encoder_; | 238 video_send_config_.encoder_settings.encoder = &fake_encoder_; |
239 video_send_config_.encoder_settings.payload_name = "FAKE"; | 239 video_send_config_.encoder_settings.payload_name = "FAKE"; |
240 video_send_config_.encoder_settings.payload_type = | 240 video_send_config_.encoder_settings.payload_type = |
241 kFakeVideoSendPayloadType; | 241 kFakeVideoSendPayloadType; |
242 video_send_config_.rtp.extensions.push_back( | 242 video_send_config_.rtp.extensions.push_back( |
243 RtpExtension(RtpExtension::kTransportSequenceNumberUri, | 243 RtpExtension(RtpExtension::kTransportSequenceNumberUri, |
244 kTransportSequenceNumberExtensionId)); | 244 kTransportSequenceNumberExtensionId)); |
245 video_send_config_.rtp.extensions.push_back(RtpExtension( | |
246 RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId)); | |
247 FillEncoderConfiguration(num_video_streams, &video_encoder_config_); | 245 FillEncoderConfiguration(num_video_streams, &video_encoder_config_); |
248 | 246 |
249 for (size_t i = 0; i < num_video_streams; ++i) | 247 for (size_t i = 0; i < num_video_streams; ++i) |
250 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); | 248 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); |
251 video_send_config_.rtp.extensions.push_back(RtpExtension( | 249 video_send_config_.rtp.extensions.push_back(RtpExtension( |
252 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); | 250 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); |
253 } | 251 } |
254 | 252 |
255 if (num_audio_streams > 0) { | 253 if (num_audio_streams > 0) { |
256 audio_send_config_ = AudioSendStream::Config(send_transport); | 254 audio_send_config_ = AudioSendStream::Config(send_transport); |
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576 | 574 |
577 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { | 575 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { |
578 } | 576 } |
579 | 577 |
580 bool EndToEndTest::ShouldCreateReceivers() const { | 578 bool EndToEndTest::ShouldCreateReceivers() const { |
581 return true; | 579 return true; |
582 } | 580 } |
583 | 581 |
584 } // namespace test | 582 } // namespace test |
585 } // namespace webrtc | 583 } // namespace webrtc |
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