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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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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170 | 170 |
171 task_queue_.SendTask([&]() { | 171 task_queue_.SendTask([&]() { |
172 metrics::Reset(); | 172 metrics::Reset(); |
173 audio_processing = AudioProcessing::Create(); | 173 audio_processing = AudioProcessing::Create(); |
174 voice_engine = VoiceEngine::Create(); | 174 voice_engine = VoiceEngine::Create(); |
175 voe_base = VoEBase::GetInterface(voice_engine); | 175 voe_base = VoEBase::GetInterface(voice_engine); |
176 fake_audio_device = rtc::MakeUnique<FakeAudioDevice>( | 176 fake_audio_device = rtc::MakeUnique<FakeAudioDevice>( |
177 FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000), | 177 FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000), |
178 FakeAudioDevice::CreateDiscardRenderer(48000), audio_rtp_speed); | 178 FakeAudioDevice::CreateDiscardRenderer(48000), audio_rtp_speed); |
179 EXPECT_EQ(0, voe_base->Init(fake_audio_device.get(), audio_processing.get(), | 179 EXPECT_EQ(0, voe_base->Init(fake_audio_device.get(), audio_processing.get(), |
180 decoder_factory_)); | 180 audio_decoder_factory_)); |
181 VoEBase::ChannelConfig config; | 181 VoEBase::ChannelConfig config; |
182 config.enable_voice_pacing = true; | 182 config.enable_voice_pacing = true; |
183 send_channel_id = voe_base->CreateChannel(config); | 183 send_channel_id = voe_base->CreateChannel(config); |
184 recv_channel_id = voe_base->CreateChannel(); | 184 recv_channel_id = voe_base->CreateChannel(); |
185 | 185 |
186 AudioState::Config send_audio_state_config; | 186 AudioState::Config send_audio_state_config; |
187 send_audio_state_config.voice_engine = voice_engine; | 187 send_audio_state_config.voice_engine = voice_engine; |
188 send_audio_state_config.audio_mixer = AudioMixerImpl::Create(); | 188 send_audio_state_config.audio_mixer = AudioMixerImpl::Create(); |
189 send_audio_state_config.audio_processing = audio_processing; | 189 send_audio_state_config.audio_processing = audio_processing; |
190 Call::Config sender_config(event_log_.get()); | 190 Call::Config sender_config(event_log_.get()); |
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244 } | 244 } |
245 video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000; | 245 video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000; |
246 video_receive_configs_[0].renderer = &observer; | 246 video_receive_configs_[0].renderer = &observer; |
247 video_receive_configs_[0].sync_group = kSyncGroup; | 247 video_receive_configs_[0].sync_group = kSyncGroup; |
248 | 248 |
249 AudioReceiveStream::Config audio_recv_config; | 249 AudioReceiveStream::Config audio_recv_config; |
250 audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc; | 250 audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc; |
251 audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc; | 251 audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc; |
252 audio_recv_config.voe_channel_id = recv_channel_id; | 252 audio_recv_config.voe_channel_id = recv_channel_id; |
253 audio_recv_config.sync_group = kSyncGroup; | 253 audio_recv_config.sync_group = kSyncGroup; |
254 audio_recv_config.decoder_factory = decoder_factory_; | 254 audio_recv_config.decoder_factory = audio_decoder_factory_; |
255 audio_recv_config.decoder_map = { | 255 audio_recv_config.decoder_map = { |
256 {kAudioSendPayloadType, {"ISAC", 16000, 1}}}; | 256 {kAudioSendPayloadType, {"ISAC", 16000, 1}}}; |
257 | 257 |
258 if (create_first == CreateOrder::kAudioFirst) { | 258 if (create_first == CreateOrder::kAudioFirst) { |
259 audio_receive_stream = | 259 audio_receive_stream = |
260 receiver_call_->CreateAudioReceiveStream(audio_recv_config); | 260 receiver_call_->CreateAudioReceiveStream(audio_recv_config); |
261 CreateVideoStreams(); | 261 CreateVideoStreams(); |
262 } else { | 262 } else { |
263 CreateVideoStreams(); | 263 CreateVideoStreams(); |
264 audio_receive_stream = | 264 audio_receive_stream = |
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779 uint32_t last_set_bitrate_kbps_; | 779 uint32_t last_set_bitrate_kbps_; |
780 VideoSendStream* send_stream_; | 780 VideoSendStream* send_stream_; |
781 test::FrameGeneratorCapturer* frame_generator_; | 781 test::FrameGeneratorCapturer* frame_generator_; |
782 VideoEncoderConfig encoder_config_; | 782 VideoEncoderConfig encoder_config_; |
783 } test; | 783 } test; |
784 | 784 |
785 RunBaseTest(&test); | 785 RunBaseTest(&test); |
786 } | 786 } |
787 | 787 |
788 } // namespace webrtc | 788 } // namespace webrtc |
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