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 197 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
208 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); | 208 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); |
209 video_send_config_.rtp.extensions.push_back(RtpExtension( | 209 video_send_config_.rtp.extensions.push_back(RtpExtension( |
210 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); | 210 RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); |
211 } | 211 } |
212 | 212 |
213 if (num_audio_streams > 0) { | 213 if (num_audio_streams > 0) { |
214 audio_send_config_ = AudioSendStream::Config(send_transport); | 214 audio_send_config_ = AudioSendStream::Config(send_transport); |
215 audio_send_config_.voe_channel_id = voe_send_.channel_id; | 215 audio_send_config_.voe_channel_id = voe_send_.channel_id; |
216 audio_send_config_.rtp.ssrc = kAudioSendSsrc; | 216 audio_send_config_.rtp.ssrc = kAudioSendSsrc; |
217 audio_send_config_.send_codec_spec.codec_inst = | 217 audio_send_config_.send_codec_spec.codec_inst = |
218 CodecInst{kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000}; | 218 CodecInst{kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000}; |
219 } | 219 } |
220 | 220 |
221 // TODO(brandtr): Update this when we support multistream protection. | 221 // TODO(brandtr): Update this when we support multistream protection. |
222 if (num_flexfec_streams > 0) { | 222 if (num_flexfec_streams > 0) { |
223 video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType; | 223 video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType; |
224 video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc; | 224 video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc; |
225 video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]}; | 225 video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]}; |
226 } | 226 } |
227 } | 227 } |
228 | 228 |
(...skipping 266 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
495 | 495 |
496 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { | 496 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { |
497 } | 497 } |
498 | 498 |
499 bool EndToEndTest::ShouldCreateReceivers() const { | 499 bool EndToEndTest::ShouldCreateReceivers() const { |
500 return true; | 500 return true; |
501 } | 501 } |
502 | 502 |
503 } // namespace test | 503 } // namespace test |
504 } // namespace webrtc | 504 } // namespace webrtc |
OLD | NEW |