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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_sender.h

Issue 1329083005: Add TransportFeedback adapter, adapting remote feedback to bwe estiamtor (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Bad merge, test issue Created 5 years, 3 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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86 86
87 class RTPSender : public RTPSenderInterface { 87 class RTPSender : public RTPSenderInterface {
88 public: 88 public:
89 RTPSender(int32_t id, 89 RTPSender(int32_t id,
90 bool audio, 90 bool audio,
91 Clock* clock, 91 Clock* clock,
92 Transport* transport, 92 Transport* transport,
93 RtpAudioFeedback* audio_feedback, 93 RtpAudioFeedback* audio_feedback,
94 PacedSender* paced_sender, 94 PacedSender* paced_sender,
95 PacketRouter* packet_router, 95 PacketRouter* packet_router,
96 SendTimeObserver* send_time_observer, 96 TransportFeedbackObserver* transport_feedback_callback,
97 BitrateStatisticsObserver* bitrate_callback, 97 BitrateStatisticsObserver* bitrate_callback,
98 FrameCountObserver* frame_count_observer, 98 FrameCountObserver* frame_count_observer,
99 SendSideDelayObserver* send_side_delay_observer); 99 SendSideDelayObserver* send_side_delay_observer);
100 virtual ~RTPSender(); 100 virtual ~RTPSender();
101 101
102 void ProcessBitrate(); 102 void ProcessBitrate();
103 103
104 uint16_t ActualSendBitrateKbit() const override; 104 uint16_t ActualSendBitrateKbit() const override;
105 105
106 uint32_t VideoBitrateSent() const; 106 uint32_t VideoBitrateSent() const;
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389 Bitrate total_bitrate_sent_; 389 Bitrate total_bitrate_sent_;
390 390
391 int32_t id_; 391 int32_t id_;
392 392
393 const bool audio_configured_; 393 const bool audio_configured_;
394 rtc::scoped_ptr<RTPSenderAudio> audio_; 394 rtc::scoped_ptr<RTPSenderAudio> audio_;
395 rtc::scoped_ptr<RTPSenderVideo> video_; 395 rtc::scoped_ptr<RTPSenderVideo> video_;
396 396
397 PacedSender* const paced_sender_; 397 PacedSender* const paced_sender_;
398 PacketRouter* const packet_router_; 398 PacketRouter* const packet_router_;
399 SendTimeObserver* const send_time_observer_; 399 TransportFeedbackObserver* const transport_feedback_observer_;
400 int64_t last_capture_time_ms_sent_; 400 int64_t last_capture_time_ms_sent_;
401 rtc::scoped_ptr<CriticalSectionWrapper> send_critsect_; 401 rtc::scoped_ptr<CriticalSectionWrapper> send_critsect_;
402 402
403 Transport *transport_; 403 Transport *transport_;
404 bool sending_media_ GUARDED_BY(send_critsect_); 404 bool sending_media_ GUARDED_BY(send_critsect_);
405 405
406 size_t max_payload_length_; 406 size_t max_payload_length_;
407 uint16_t packet_over_head_; 407 uint16_t packet_over_head_;
408 408
409 int8_t payload_type_ GUARDED_BY(send_critsect_); 409 int8_t payload_type_ GUARDED_BY(send_critsect_);
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461 // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember 461 // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember
462 // that by the time the function returns there is no guarantee 462 // that by the time the function returns there is no guarantee
463 // that the target bitrate is still valid. 463 // that the target bitrate is still valid.
464 rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_; 464 rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
465 uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_); 465 uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
466 }; 466 };
467 467
468 } // namespace webrtc 468 } // namespace webrtc
469 469
470 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ 470 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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