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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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| 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)); |
| 211 FillEncoderConfiguration(num_video_streams, &video_encoder_config_); | 213 FillEncoderConfiguration(num_video_streams, &video_encoder_config_); |
| 212 | 214 |
| 213 for (size_t i = 0; i < num_video_streams; ++i) | 215 for (size_t i = 0; i < num_video_streams; ++i) |
| 214 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); | 216 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); |
| 215 video_send_config_.rtp.extensions.push_back(RtpExtension( | 217 video_send_config_.rtp.extensions.push_back(RtpExtension( |
| 216 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); | 218 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); |
| 217 } | 219 } |
| 218 | 220 |
| 219 if (num_audio_streams > 0) { | 221 if (num_audio_streams > 0) { |
| 220 audio_send_config_ = AudioSendStream::Config(send_transport); | 222 audio_send_config_ = AudioSendStream::Config(send_transport); |
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| 537 | 539 |
| 538 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { | 540 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { |
| 539 } | 541 } |
| 540 | 542 |
| 541 bool EndToEndTest::ShouldCreateReceivers() const { | 543 bool EndToEndTest::ShouldCreateReceivers() const { |
| 542 return true; | 544 return true; |
| 543 } | 545 } |
| 544 | 546 |
| 545 } // namespace test | 547 } // namespace test |
| 546 } // namespace webrtc | 548 } // namespace webrtc |
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