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Side by Side Diff: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Updated RTP/RTCP module to use setter methods instead of passing the event log pointer in the const… Created 4 years, 9 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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68 TransportFeedbackObserver* transport_feedback_callback; 68 TransportFeedbackObserver* transport_feedback_callback;
69 RtcpRttStats* rtt_stats; 69 RtcpRttStats* rtt_stats;
70 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer; 70 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;
71 RtpAudioFeedback* audio_messages; 71 RtpAudioFeedback* audio_messages;
72 RemoteBitrateEstimator* remote_bitrate_estimator; 72 RemoteBitrateEstimator* remote_bitrate_estimator;
73 RtpPacketSender* paced_sender; 73 RtpPacketSender* paced_sender;
74 TransportSequenceNumberAllocator* transport_sequence_number_allocator; 74 TransportSequenceNumberAllocator* transport_sequence_number_allocator;
75 BitrateStatisticsObserver* send_bitrate_observer; 75 BitrateStatisticsObserver* send_bitrate_observer;
76 FrameCountObserver* send_frame_count_observer; 76 FrameCountObserver* send_frame_count_observer;
77 SendSideDelayObserver* send_side_delay_observer; 77 SendSideDelayObserver* send_side_delay_observer;
78 RtcEventLog* event_log;
stefan-webrtc 2016/03/03 10:08:25 Why do we have to make this change? If it really i
the sun 2016/03/03 10:17:15 That's a good idea! Put the proxy among the others
ivoc 2016/03/10 13:15:36 The reason for this change is that we no longer ha
79 78
80 RTC_DISALLOW_COPY_AND_ASSIGN(Configuration); 79 RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
81 }; 80 };
82 81
83 /* 82 /*
84 * Create a RTP/RTCP module object using the system clock. 83 * Create a RTP/RTCP module object using the system clock.
85 * 84 *
86 * configuration - Configuration of the RTP/RTCP module. 85 * configuration - Configuration of the RTP/RTCP module.
87 */ 86 */
88 static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration); 87 static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration);
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642 * return -1 on failure else 0 641 * return -1 on failure else 0
643 */ 642 */
644 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; 643 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0;
645 644
646 /* 645 /*
647 * send a request for a keyframe 646 * send a request for a keyframe
648 * 647 *
649 * return -1 on failure else 0 648 * return -1 on failure else 0
650 */ 649 */
651 virtual int32_t RequestKeyFrame() = 0; 650 virtual int32_t RequestKeyFrame() = 0;
651
652 /*
653 * Set the RtcEventLog object.
654 */
655 virtual void SetRtcEventLog(webrtc::RtcEventLog* event_log) = 0;
652 }; 656 };
653 } // namespace webrtc 657 } // namespace webrtc
654 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ 658 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
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