OLD | NEW |
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 |
(...skipping 363 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
374 int32_t OnReceivedPayloadData(const uint8_t* payloadData, | 374 int32_t OnReceivedPayloadData(const uint8_t* payloadData, |
375 size_t payloadSize, | 375 size_t payloadSize, |
376 const WebRtcRTPHeader* rtpHeader) override; | 376 const WebRtcRTPHeader* rtpHeader) override; |
377 bool OnRecoveredPacket(const uint8_t* packet, | 377 bool OnRecoveredPacket(const uint8_t* packet, |
378 size_t packet_length) override; | 378 size_t packet_length) override; |
379 | 379 |
380 // From RtpFeedback in the RTP/RTCP module | 380 // From RtpFeedback in the RTP/RTCP module |
381 int32_t OnInitializeDecoder(int8_t payloadType, | 381 int32_t OnInitializeDecoder(int8_t payloadType, |
382 const char payloadName[RTP_PAYLOAD_NAME_SIZE], | 382 const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
383 int frequency, | 383 int frequency, |
384 uint8_t channels, | 384 size_t channels, |
385 uint32_t rate) override; | 385 uint32_t rate) override; |
386 void OnIncomingSSRCChanged(uint32_t ssrc) override; | 386 void OnIncomingSSRCChanged(uint32_t ssrc) override; |
387 void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override; | 387 void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override; |
388 | 388 |
389 // From RtpAudioFeedback in the RTP/RTCP module | 389 // From RtpAudioFeedback in the RTP/RTCP module |
390 void OnPlayTelephoneEvent(uint8_t event, | 390 void OnPlayTelephoneEvent(uint8_t event, |
391 uint16_t lengthMs, | 391 uint16_t lengthMs, |
392 uint8_t volume) override; | 392 uint8_t volume) override; |
393 | 393 |
394 // From Transport (called by the RTP/RTCP module) | 394 // From Transport (called by the RTP/RTCP module) |
(...skipping 49 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
444 { | 444 { |
445 return _outputAudioLevel.Level(); | 445 return _outputAudioLevel.Level(); |
446 } | 446 } |
447 uint32_t Demultiplex(const AudioFrame& audioFrame); | 447 uint32_t Demultiplex(const AudioFrame& audioFrame); |
448 // Demultiplex the data to the channel's |_audioFrame|. The difference | 448 // Demultiplex the data to the channel's |_audioFrame|. The difference |
449 // between this method and the overloaded method above is that |audio_data| | 449 // between this method and the overloaded method above is that |audio_data| |
450 // does not go through transmit_mixer and APM. | 450 // does not go through transmit_mixer and APM. |
451 void Demultiplex(const int16_t* audio_data, | 451 void Demultiplex(const int16_t* audio_data, |
452 int sample_rate, | 452 int sample_rate, |
453 size_t number_of_frames, | 453 size_t number_of_frames, |
454 int number_of_channels); | 454 size_t number_of_channels); |
455 uint32_t PrepareEncodeAndSend(int mixingFrequency); | 455 uint32_t PrepareEncodeAndSend(int mixingFrequency); |
456 uint32_t EncodeAndSend(); | 456 uint32_t EncodeAndSend(); |
457 | 457 |
458 // Associate to a send channel. | 458 // Associate to a send channel. |
459 // Used for obtaining RTT for a receive-only channel. | 459 // Used for obtaining RTT for a receive-only channel. |
460 void set_associate_send_channel(const ChannelOwner& channel) { | 460 void set_associate_send_channel(const ChannelOwner& channel) { |
461 assert(_channelId != channel.channel()->ChannelId()); | 461 assert(_channelId != channel.channel()->ChannelId()); |
462 CriticalSectionScoped lock(assoc_send_channel_lock_.get()); | 462 CriticalSectionScoped lock(assoc_send_channel_lock_.get()); |
463 associate_send_channel_ = channel; | 463 associate_send_channel_ = channel; |
464 } | 464 } |
(...skipping 139 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
604 PacketRouter* packet_router_ = nullptr; | 604 PacketRouter* packet_router_ = nullptr; |
605 rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_; | 605 rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_; |
606 rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; | 606 rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; |
607 rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; | 607 rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; |
608 }; | 608 }; |
609 | 609 |
610 } // namespace voe | 610 } // namespace voe |
611 } // namespace webrtc | 611 } // namespace webrtc |
612 | 612 |
613 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 613 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
OLD | NEW |