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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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| 149 rtc::CriticalSection lock_; | 149 rtc::CriticalSection lock_; |
| 150 State state_; | 150 State state_; |
| 151 }; | 151 }; |
| 152 | 152 |
| 153 class Channel | 153 class Channel |
| 154 : public RtpData, | 154 : public RtpData, |
| 155 public RtpFeedback, | 155 public RtpFeedback, |
| 156 public FileCallback, // receiving notification from file player & | 156 public FileCallback, // receiving notification from file player & |
| 157 // recorder | 157 // recorder |
| 158 public Transport, | 158 public Transport, |
| 159 public RtpAudioFeedback, | |
| 160 public AudioPacketizationCallback, // receive encoded packets from the | 159 public AudioPacketizationCallback, // receive encoded packets from the |
| 161 // ACM | 160 // ACM |
| 162 public ACMVADCallback, // receive voice activity from the ACM | 161 public ACMVADCallback, // receive voice activity from the ACM |
| 163 public MixerParticipant // supplies output mixer with audio frames | 162 public MixerParticipant // supplies output mixer with audio frames |
| 164 { | 163 { |
| 165 public: | 164 public: |
| 166 friend class VoERtcpObserver; | 165 friend class VoERtcpObserver; |
| 167 | 166 |
| 168 enum { KNumSocketThreads = 1 }; | 167 enum { KNumSocketThreads = 1 }; |
| 169 enum { KNumberOfSocketBuffers = 8 }; | 168 enum { KNumberOfSocketBuffers = 8 }; |
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| 378 | 377 |
| 379 // From RtpFeedback in the RTP/RTCP module | 378 // From RtpFeedback in the RTP/RTCP module |
| 380 int32_t OnInitializeDecoder(int8_t payloadType, | 379 int32_t OnInitializeDecoder(int8_t payloadType, |
| 381 const char payloadName[RTP_PAYLOAD_NAME_SIZE], | 380 const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
| 382 int frequency, | 381 int frequency, |
| 383 size_t channels, | 382 size_t channels, |
| 384 uint32_t rate) override; | 383 uint32_t rate) override; |
| 385 void OnIncomingSSRCChanged(uint32_t ssrc) override; | 384 void OnIncomingSSRCChanged(uint32_t ssrc) override; |
| 386 void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override; | 385 void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override; |
| 387 | 386 |
| 388 // From RtpAudioFeedback in the RTP/RTCP module | |
| 389 void OnPlayTelephoneEvent(uint8_t event, | |
| 390 uint16_t lengthMs, | |
| 391 uint8_t volume) override; | |
| 392 | |
| 393 // From Transport (called by the RTP/RTCP module) | 387 // From Transport (called by the RTP/RTCP module) |
| 394 bool SendRtp(const uint8_t* data, | 388 bool SendRtp(const uint8_t* data, |
| 395 size_t len, | 389 size_t len, |
| 396 const PacketOptions& packet_options) override; | 390 const PacketOptions& packet_options) override; |
| 397 bool SendRtcp(const uint8_t* data, size_t len) override; | 391 bool SendRtcp(const uint8_t* data, size_t len) override; |
| 398 | 392 |
| 399 // From MixerParticipant | 393 // From MixerParticipant |
| 400 int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override; | 394 int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override; |
| 401 int32_t NeededFrequency(int32_t id) const override; | 395 int32_t NeededFrequency(int32_t id) const override; |
| 402 | 396 |
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| 573 PacketRouter* packet_router_ = nullptr; | 567 PacketRouter* packet_router_ = nullptr; |
| 574 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; | 568 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; |
| 575 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; | 569 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; |
| 576 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; | 570 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; |
| 577 }; | 571 }; |
| 578 | 572 |
| 579 } // namespace voe | 573 } // namespace voe |
| 580 } // namespace webrtc | 574 } // namespace webrtc |
| 581 | 575 |
| 582 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 576 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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