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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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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22 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" | 22 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" |
23 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 23 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
24 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" | 24 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" |
25 #include "webrtc/modules/audio_processing/rms_level.h" | 25 #include "webrtc/modules/audio_processing/rms_level.h" |
26 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" | 26 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
27 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 27 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
28 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 28 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
29 #include "webrtc/modules/utility/include/file_player.h" | 29 #include "webrtc/modules/utility/include/file_player.h" |
30 #include "webrtc/modules/utility/include/file_recorder.h" | 30 #include "webrtc/modules/utility/include/file_recorder.h" |
31 #include "webrtc/voice_engine/include/voe_audio_processing.h" | 31 #include "webrtc/voice_engine/include/voe_audio_processing.h" |
| 32 #include "webrtc/voice_engine/include/voe_base.h" |
32 #include "webrtc/voice_engine/include/voe_network.h" | 33 #include "webrtc/voice_engine/include/voe_network.h" |
33 #include "webrtc/voice_engine/level_indicator.h" | 34 #include "webrtc/voice_engine/level_indicator.h" |
34 #include "webrtc/voice_engine/network_predictor.h" | 35 #include "webrtc/voice_engine/network_predictor.h" |
35 #include "webrtc/voice_engine/shared_data.h" | 36 #include "webrtc/voice_engine/shared_data.h" |
36 #include "webrtc/voice_engine/voice_engine_defines.h" | 37 #include "webrtc/voice_engine/voice_engine_defines.h" |
37 | 38 |
38 namespace rtc { | 39 namespace rtc { |
39 class TimestampWrapAroundHandler; | 40 class TimestampWrapAroundHandler; |
40 } | 41 } |
41 | 42 |
42 namespace webrtc { | 43 namespace webrtc { |
43 | 44 |
44 class AudioDeviceModule; | 45 class AudioDeviceModule; |
45 class Config; | |
46 class FileWrapper; | 46 class FileWrapper; |
47 class PacketRouter; | 47 class PacketRouter; |
48 class ProcessThread; | 48 class ProcessThread; |
49 class RateLimiter; | 49 class RateLimiter; |
50 class ReceiveStatistics; | 50 class ReceiveStatistics; |
51 class RemoteNtpTimeEstimator; | 51 class RemoteNtpTimeEstimator; |
52 class RtcEventLog; | 52 class RtcEventLog; |
53 class RTPPayloadRegistry; | 53 class RTPPayloadRegistry; |
54 class RtpReceiver; | 54 class RtpReceiver; |
55 class RTPReceiverAudio; | 55 class RTPReceiverAudio; |
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168 public: | 168 public: |
169 friend class VoERtcpObserver; | 169 friend class VoERtcpObserver; |
170 | 170 |
171 enum { KNumSocketThreads = 1 }; | 171 enum { KNumSocketThreads = 1 }; |
172 enum { KNumberOfSocketBuffers = 8 }; | 172 enum { KNumberOfSocketBuffers = 8 }; |
173 virtual ~Channel(); | 173 virtual ~Channel(); |
174 static int32_t CreateChannel( | 174 static int32_t CreateChannel( |
175 Channel*& channel, | 175 Channel*& channel, |
176 int32_t channelId, | 176 int32_t channelId, |
177 uint32_t instanceId, | 177 uint32_t instanceId, |
178 const Config& config, | 178 const VoEBase::ChannelConfig& config); |
179 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory); | |
180 Channel(int32_t channelId, | 179 Channel(int32_t channelId, |
181 uint32_t instanceId, | 180 uint32_t instanceId, |
182 const Config& config, | 181 const VoEBase::ChannelConfig& config); |
183 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory); | |
184 int32_t Init(); | 182 int32_t Init(); |
185 int32_t SetEngineInformation(Statistics& engineStatistics, | 183 int32_t SetEngineInformation(Statistics& engineStatistics, |
186 OutputMixer& outputMixer, | 184 OutputMixer& outputMixer, |
187 TransmitMixer& transmitMixer, | 185 TransmitMixer& transmitMixer, |
188 ProcessThread& moduleProcessThread, | 186 ProcessThread& moduleProcessThread, |
189 AudioDeviceModule& audioDeviceModule, | 187 AudioDeviceModule& audioDeviceModule, |
190 VoiceEngineObserver* voiceEngineObserver, | 188 VoiceEngineObserver* voiceEngineObserver, |
191 rtc::CriticalSection* callbackCritSect); | 189 rtc::CriticalSection* callbackCritSect); |
192 int32_t UpdateLocalTimeStamp(); | 190 int32_t UpdateLocalTimeStamp(); |
193 | 191 |
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584 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; | 582 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
585 | 583 |
586 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 584 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
587 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 585 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
588 }; | 586 }; |
589 | 587 |
590 } // namespace voe | 588 } // namespace voe |
591 } // namespace webrtc | 589 } // namespace webrtc |
592 | 590 |
593 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 591 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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