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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 #ifndef WEBRTC_TEST_CALL_TEST_H_ | 10 #ifndef WEBRTC_TEST_CALL_TEST_H_ |
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132 std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_; | 132 std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_; |
133 std::vector<FlexfecReceiveStream*> flexfec_receive_streams_; | 133 std::vector<FlexfecReceiveStream*> flexfec_receive_streams_; |
134 | 134 |
135 std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_; | 135 std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_; |
136 test::FakeEncoder fake_encoder_; | 136 test::FakeEncoder fake_encoder_; |
137 std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders_; | 137 std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders_; |
138 size_t num_video_streams_; | 138 size_t num_video_streams_; |
139 size_t num_audio_streams_; | 139 size_t num_audio_streams_; |
140 size_t num_flexfec_streams_; | 140 size_t num_flexfec_streams_; |
141 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 141 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
| 142 rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_; |
142 test::FakeVideoRenderer fake_renderer_; | 143 test::FakeVideoRenderer fake_renderer_; |
143 | 144 |
144 PayloadDemuxer receive_demuxer_; | 145 PayloadDemuxer receive_demuxer_; |
145 PayloadDemuxer send_demuxer_; | 146 PayloadDemuxer send_demuxer_; |
146 | 147 |
147 private: | 148 private: |
148 // TODO(holmer): Remove once VoiceEngine is fully refactored to the new API. | 149 // TODO(holmer): Remove once VoiceEngine is fully refactored to the new API. |
149 // These methods are used to set up legacy voice engines and channels which is | 150 // These methods are used to set up legacy voice engines and channels which is |
150 // necessary while voice engine is being refactored to the new stream API. | 151 // necessary while voice engine is being refactored to the new stream API. |
151 struct VoiceEngineState { | 152 struct VoiceEngineState { |
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242 EndToEndTest(); | 243 EndToEndTest(); |
243 explicit EndToEndTest(unsigned int timeout_ms); | 244 explicit EndToEndTest(unsigned int timeout_ms); |
244 | 245 |
245 bool ShouldCreateReceivers() const override; | 246 bool ShouldCreateReceivers() const override; |
246 }; | 247 }; |
247 | 248 |
248 } // namespace test | 249 } // namespace test |
249 } // namespace webrtc | 250 } // namespace webrtc |
250 | 251 |
251 #endif // WEBRTC_TEST_CALL_TEST_H_ | 252 #endif // WEBRTC_TEST_CALL_TEST_H_ |
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