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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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159 FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename, | 159 FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename, |
160 audio_rtp_speed); | 160 audio_rtp_speed); |
161 EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_)); | 161 EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_)); |
162 VoEBase::ChannelConfig config; | 162 VoEBase::ChannelConfig config; |
163 config.enable_voice_pacing = true; | 163 config.enable_voice_pacing = true; |
164 int send_channel_id = voe_base->CreateChannel(config); | 164 int send_channel_id = voe_base->CreateChannel(config); |
165 int recv_channel_id = voe_base->CreateChannel(); | 165 int recv_channel_id = voe_base->CreateChannel(); |
166 | 166 |
167 AudioState::Config send_audio_state_config; | 167 AudioState::Config send_audio_state_config; |
168 send_audio_state_config.voice_engine = voice_engine; | 168 send_audio_state_config.voice_engine = voice_engine; |
| 169 send_audio_state_config.audio_device_module = &fake_audio_device; |
169 Call::Config sender_config; | 170 Call::Config sender_config; |
170 sender_config.audio_state = AudioState::Create(send_audio_state_config); | 171 sender_config.audio_state = AudioState::Create(send_audio_state_config); |
171 Call::Config receiver_config; | 172 Call::Config receiver_config; |
172 receiver_config.audio_state = sender_config.audio_state; | 173 receiver_config.audio_state = sender_config.audio_state; |
173 CreateCalls(sender_config, receiver_config); | 174 CreateCalls(sender_config, receiver_config); |
174 | 175 |
175 | 176 |
176 VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock()); | 177 VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock()); |
177 | 178 |
178 // Helper class to ensure we deliver correct media_type to the receiving call. | 179 // Helper class to ensure we deliver correct media_type to the receiving call. |
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697 int encoder_inits_; | 698 int encoder_inits_; |
698 uint32_t last_set_bitrate_; | 699 uint32_t last_set_bitrate_; |
699 VideoSendStream* send_stream_; | 700 VideoSendStream* send_stream_; |
700 VideoEncoderConfig encoder_config_; | 701 VideoEncoderConfig encoder_config_; |
701 } test; | 702 } test; |
702 | 703 |
703 RunBaseTest(&test); | 704 RunBaseTest(&test); |
704 } | 705 } |
705 | 706 |
706 } // namespace webrtc | 707 } // namespace webrtc |
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