Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(31)

Side by Side Diff: webrtc/logging/rtc_event_log/rtc_event_log_parser.h

Issue 2631703002: Revert of Log audio network adapter decisions in event log. (Closed)
Patch Set: Created 3 years, 11 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) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 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 #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_ 10 #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
(...skipping 29 matching lines...) Expand all
40 LOG_START = 1, 40 LOG_START = 1,
41 LOG_END = 2, 41 LOG_END = 2,
42 RTP_EVENT = 3, 42 RTP_EVENT = 3,
43 RTCP_EVENT = 4, 43 RTCP_EVENT = 4,
44 AUDIO_PLAYOUT_EVENT = 5, 44 AUDIO_PLAYOUT_EVENT = 5,
45 BWE_PACKET_LOSS_EVENT = 6, 45 BWE_PACKET_LOSS_EVENT = 6,
46 BWE_PACKET_DELAY_EVENT = 7, 46 BWE_PACKET_DELAY_EVENT = 7,
47 VIDEO_RECEIVER_CONFIG_EVENT = 8, 47 VIDEO_RECEIVER_CONFIG_EVENT = 8,
48 VIDEO_SENDER_CONFIG_EVENT = 9, 48 VIDEO_SENDER_CONFIG_EVENT = 9,
49 AUDIO_RECEIVER_CONFIG_EVENT = 10, 49 AUDIO_RECEIVER_CONFIG_EVENT = 10,
50 AUDIO_SENDER_CONFIG_EVENT = 11, 50 AUDIO_SENDER_CONFIG_EVENT = 11
51 AUDIO_NETWORK_ADAPTATION_EVENT = 16
52 }; 51 };
53 52
54 // Reads an RtcEventLog file and returns true if parsing was successful. 53 // Reads an RtcEventLog file and returns true if parsing was successful.
55 bool ParseFile(const std::string& file_name); 54 bool ParseFile(const std::string& file_name);
56 55
57 // Reads an RtcEventLog from a string and returns true if successful. 56 // Reads an RtcEventLog from a string and returns true if successful.
58 bool ParseString(const std::string& s); 57 bool ParseString(const std::string& s);
59 58
60 // Reads an RtcEventLog from an istream and returns true if successful. 59 // Reads an RtcEventLog from an istream and returns true if successful.
61 bool ParseStream(std::istream& stream); 60 bool ParseStream(std::istream& stream);
(...skipping 55 matching lines...) Expand 10 before | Expand all | Expand 10 after
117 // Reads bitrate, fraction loss (as defined in RFC 1889) and total number of 116 // Reads bitrate, fraction loss (as defined in RFC 1889) and total number of
118 // expected packets from the BWE event at |index| and stores the values in 117 // expected packets from the BWE event at |index| and stores the values in
119 // the corresponding output parameters. The output parameters can be set to 118 // the corresponding output parameters. The output parameters can be set to
120 // nullptr if those values aren't needed. 119 // nullptr if those values aren't needed.
121 // NB: The packet must have space for at least IP_PACKET_SIZE bytes. 120 // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
122 void GetBwePacketLossEvent(size_t index, 121 void GetBwePacketLossEvent(size_t index,
123 int32_t* bitrate, 122 int32_t* bitrate,
124 uint8_t* fraction_loss, 123 uint8_t* fraction_loss,
125 int32_t* total_packets) const; 124 int32_t* total_packets) const;
126 125
127 // Reads a audio network adaptation event to a (non-NULL)
128 // AudioNetworkAdaptor::EncoderRuntimeConfig struct. Only the fields that are
129 // stored in the protobuf will be written.
130 void GetAudioNetworkAdaptation(
131 size_t index,
132 AudioNetworkAdaptor::EncoderRuntimeConfig* config) const;
133
134 private: 126 private:
135 std::vector<rtclog::Event> events_; 127 std::vector<rtclog::Event> events_;
136 }; 128 };
137 129
138 } // namespace webrtc 130 } // namespace webrtc
139 131
140 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_ 132 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
OLDNEW
« no previous file with comments | « webrtc/logging/rtc_event_log/rtc_event_log.proto ('k') | webrtc/logging/rtc_event_log/rtc_event_log_parser.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698