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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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196 void CallTest::DestroyCalls() { | 196 void CallTest::DestroyCalls() { |
197 sender_call_.reset(); | 197 sender_call_.reset(); |
198 receiver_call_.reset(); | 198 receiver_call_.reset(); |
199 } | 199 } |
200 | 200 |
201 void CallTest::CreateSendConfig(size_t num_video_streams, | 201 void CallTest::CreateSendConfig(size_t num_video_streams, |
202 size_t num_audio_streams, | 202 size_t num_audio_streams, |
203 size_t num_flexfec_streams, | 203 size_t num_flexfec_streams, |
204 Transport* send_transport) { | 204 Transport* send_transport) { |
205 RTC_DCHECK(num_video_streams <= kNumSsrcs); | 205 RTC_DCHECK(num_video_streams <= kNumSsrcs); |
206 RTC_DCHECK_LE(num_audio_streams, 1u); | 206 RTC_DCHECK_LE(num_audio_streams, 1); |
207 RTC_DCHECK_LE(num_flexfec_streams, 1u); | 207 RTC_DCHECK_LE(num_flexfec_streams, 1); |
208 RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0); | 208 RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0); |
209 if (num_video_streams > 0) { | 209 if (num_video_streams > 0) { |
210 video_send_config_ = VideoSendStream::Config(send_transport); | 210 video_send_config_ = VideoSendStream::Config(send_transport); |
211 video_send_config_.encoder_settings.encoder = &fake_encoder_; | 211 video_send_config_.encoder_settings.encoder = &fake_encoder_; |
212 video_send_config_.encoder_settings.payload_name = "FAKE"; | 212 video_send_config_.encoder_settings.payload_name = "FAKE"; |
213 video_send_config_.encoder_settings.payload_type = | 213 video_send_config_.encoder_settings.payload_type = |
214 kFakeVideoSendPayloadType; | 214 kFakeVideoSendPayloadType; |
215 video_send_config_.rtp.extensions.push_back( | 215 video_send_config_.rtp.extensions.push_back( |
216 RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId)); | 216 RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId)); |
217 FillEncoderConfiguration(num_video_streams, &video_encoder_config_); | 217 FillEncoderConfiguration(num_video_streams, &video_encoder_config_); |
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254 test::CreateMatchingDecoder(video_send_config_.encoder_settings); | 254 test::CreateMatchingDecoder(video_send_config_.encoder_settings); |
255 allocated_decoders_.push_back( | 255 allocated_decoders_.push_back( |
256 std::unique_ptr<VideoDecoder>(decoder.decoder)); | 256 std::unique_ptr<VideoDecoder>(decoder.decoder)); |
257 video_config.decoders.clear(); | 257 video_config.decoders.clear(); |
258 video_config.decoders.push_back(decoder); | 258 video_config.decoders.push_back(decoder); |
259 video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i]; | 259 video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i]; |
260 video_receive_configs_.push_back(video_config.Copy()); | 260 video_receive_configs_.push_back(video_config.Copy()); |
261 } | 261 } |
262 } | 262 } |
263 | 263 |
264 RTC_DCHECK_GE(1u, num_audio_streams_); | 264 RTC_DCHECK_GE(1, num_audio_streams_); |
265 if (num_audio_streams_ == 1) { | 265 if (num_audio_streams_ == 1) { |
266 RTC_DCHECK_LE(0, voe_send_.channel_id); | 266 RTC_DCHECK_LE(0, voe_send_.channel_id); |
267 AudioReceiveStream::Config audio_config; | 267 AudioReceiveStream::Config audio_config; |
268 audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; | 268 audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; |
269 audio_config.rtcp_send_transport = rtcp_send_transport; | 269 audio_config.rtcp_send_transport = rtcp_send_transport; |
270 audio_config.voe_channel_id = voe_recv_.channel_id; | 270 audio_config.voe_channel_id = voe_recv_.channel_id; |
271 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc; | 271 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc; |
272 audio_config.decoder_factory = decoder_factory_; | 272 audio_config.decoder_factory = decoder_factory_; |
273 audio_receive_configs_.push_back(audio_config); | 273 audio_receive_configs_.push_back(audio_config); |
274 } | 274 } |
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507 | 507 |
508 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { | 508 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { |
509 } | 509 } |
510 | 510 |
511 bool EndToEndTest::ShouldCreateReceivers() const { | 511 bool EndToEndTest::ShouldCreateReceivers() const { |
512 return true; | 512 return true; |
513 } | 513 } |
514 | 514 |
515 } // namespace test | 515 } // namespace test |
516 } // namespace webrtc | 516 } // namespace webrtc |
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