Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(27)

Side by Side Diff: webrtc/call/rtc_event_log.h

Issue 1411673003: Added protobuf message for loss-based BWE events, and wired it up to the send side bandwidth estima… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Logging for loss based BWE Created 5 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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 55 matching lines...) Expand 10 before | Expand all | Expand 10 after
66 66
67 // Logs an incoming or outgoing RTCP packet. 67 // Logs an incoming or outgoing RTCP packet.
68 virtual void LogRtcpPacket(bool incoming, 68 virtual void LogRtcpPacket(bool incoming,
69 MediaType media_type, 69 MediaType media_type,
70 const uint8_t* packet, 70 const uint8_t* packet,
71 size_t length) = 0; 71 size_t length) = 0;
72 72
73 // Logs an audio playout event 73 // Logs an audio playout event
74 virtual void LogAudioPlayout(uint32_t ssrc) = 0; 74 virtual void LogAudioPlayout(uint32_t ssrc) = 0;
75 75
76 // Logs a bitrate update from the bandwidth estimator based on packet loss.
77 virtual void LogBwePacketLossEvent(int32_t bitrate,
stefan-webrtc 2015/10/19 08:43:21 I think we can make this a regular int. Same for t
terelius 2015/10/26 17:40:18 The type here should correspond to the protobuf I
stefan-webrtc 2015/10/28 16:03:18 That is true, although I think it's mostly legacy
terelius 2015/10/30 10:51:40 It is mostly that I don't think it makes sense to
stefan-webrtc 2015/10/30 10:58:04 Agree it doesn't make sense to cast just to cast b
78 uint8_t fraction_loss,
79 int32_t total_packets) = 0;
80
76 // Reads an RtcEventLog file and returns true when reading was successful. 81 // Reads an RtcEventLog file and returns true when reading was successful.
77 // The result is stored in the given EventStream object. 82 // The result is stored in the given EventStream object.
78 static bool ParseRtcEventLog(const std::string& file_name, 83 static bool ParseRtcEventLog(const std::string& file_name,
79 rtclog::EventStream* result); 84 rtclog::EventStream* result);
80 }; 85 };
81 86
82 } // namespace webrtc 87 } // namespace webrtc
83 88
84 #endif // WEBRTC_CALL_RTC_EVENT_LOG_H_ 89 #endif // WEBRTC_CALL_RTC_EVENT_LOG_H_
OLDNEW
« no previous file with comments | « webrtc/call/call.cc ('k') | webrtc/call/rtc_event_log.cc » ('j') | webrtc/call/rtc_event_log.cc » ('J')

Powered by Google App Engine
This is Rietveld 408576698