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Side by Side Diff: webrtc/video_engine/vie_channel.h

Issue 1251163002: Remove base channel for video receivers. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: fix data race Created 5 years, 5 months ago
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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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