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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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102 const Config& config, | 102 const Config& config, |
103 Transport* transport, | 103 Transport* transport, |
104 ProcessThread* module_process_thread, | 104 ProcessThread* module_process_thread, |
105 RtcpIntraFrameObserver* intra_frame_observer, | 105 RtcpIntraFrameObserver* intra_frame_observer, |
106 RtcpBandwidthObserver* bandwidth_observer, | 106 RtcpBandwidthObserver* bandwidth_observer, |
107 RemoteBitrateEstimator* remote_bitrate_estimator, | 107 RemoteBitrateEstimator* remote_bitrate_estimator, |
108 RtcpRttStats* rtt_stats, | 108 RtcpRttStats* rtt_stats, |
109 PacedSender* paced_sender, | 109 PacedSender* paced_sender, |
110 PacketRouter* packet_router, | 110 PacketRouter* packet_router, |
111 size_t max_rtp_streams, | 111 size_t max_rtp_streams, |
112 bool sender, | 112 bool sender); |
113 bool disable_default_encoder); | |
114 ~ViEChannel(); | 113 ~ViEChannel(); |
115 | 114 |
116 int32_t Init(); | 115 int32_t Init(); |
117 | 116 |
118 // Sets the encoder to use for the channel. |new_stream| indicates the encoder | 117 // Sets the encoder to use for the channel. |new_stream| indicates the encoder |
119 // type has changed and we should start a new RTP stream. | 118 // type has changed and we should start a new RTP stream. |
120 int32_t SetSendCodec(const VideoCodec& video_codec, bool new_stream = true); | 119 int32_t SetSendCodec(const VideoCodec& video_codec, bool new_stream = true); |
121 int32_t SetReceiveCodec(const VideoCodec& video_codec); | 120 int32_t SetReceiveCodec(const VideoCodec& video_codec); |
122 int32_t RegisterCodecObserver(ViEDecoderObserver* observer); | 121 int32_t RegisterCodecObserver(ViEDecoderObserver* observer); |
123 // Registers an external decoder. |buffered_rendering| means that the decoder | 122 // Registers an external decoder. |buffered_rendering| means that the decoder |
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471 PacedSender* const paced_sender_; | 470 PacedSender* const paced_sender_; |
472 PacketRouter* const packet_router_; | 471 PacketRouter* const packet_router_; |
473 | 472 |
474 const rtc::scoped_ptr<RtcpBandwidthObserver> bandwidth_observer_; | 473 const rtc::scoped_ptr<RtcpBandwidthObserver> bandwidth_observer_; |
475 | 474 |
476 bool decoder_reset_ GUARDED_BY(crit_); | 475 bool decoder_reset_ GUARDED_BY(crit_); |
477 // Current receive codec used for codec change callback. | 476 // Current receive codec used for codec change callback. |
478 VideoCodec receive_codec_ GUARDED_BY(crit_); | 477 VideoCodec receive_codec_ GUARDED_BY(crit_); |
479 rtc::scoped_ptr<ThreadWrapper> decode_thread_; | 478 rtc::scoped_ptr<ThreadWrapper> decode_thread_; |
480 | 479 |
481 // Used to skip default encoder in the new API. | |
482 const bool disable_default_encoder_; | |
483 | |
484 int nack_history_size_sender_; | 480 int nack_history_size_sender_; |
485 int max_nack_reordering_threshold_; | 481 int max_nack_reordering_threshold_; |
486 I420FrameCallback* pre_render_callback_ GUARDED_BY(crit_); | 482 I420FrameCallback* pre_render_callback_ GUARDED_BY(crit_); |
487 | 483 |
488 const rtc::scoped_ptr<ReportBlockStats> report_block_stats_sender_; | 484 const rtc::scoped_ptr<ReportBlockStats> report_block_stats_sender_; |
489 | 485 |
490 // RtpRtcp modules, declared last as they use other members on construction. | 486 // RtpRtcp modules, declared last as they use other members on construction. |
491 const std::vector<RtpRtcp*> rtp_rtcp_modules_; | 487 const std::vector<RtpRtcp*> rtp_rtcp_modules_; |
492 size_t num_active_rtp_rtcp_modules_ GUARDED_BY(crit_); | 488 size_t num_active_rtp_rtcp_modules_ GUARDED_BY(crit_); |
493 }; | 489 }; |
494 | 490 |
495 } // namespace webrtc | 491 } // namespace webrtc |
496 | 492 |
497 #endif // WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_ | 493 #endif // WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_ |
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