Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(104)

Side by Side Diff: webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc

Issue 2912323003: Fix a bug in RtcEventLogSource (Closed)
Patch Set: Making a local solution in the test code Created 3 years, 6 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 57 matching lines...) Expand 10 before | Expand all | Expand 10 after
68 } 68 }
69 69
70 if (parsed_stream_.GetMediaType(packet->header().ssrc, direction) != 70 if (parsed_stream_.GetMediaType(packet->header().ssrc, direction) !=
71 webrtc::ParsedRtcEventLog::MediaType::AUDIO) { 71 webrtc::ParsedRtcEventLog::MediaType::AUDIO) {
72 continue; 72 continue;
73 } 73 }
74 74
75 // Check if the packet should not be filtered out. 75 // Check if the packet should not be filtered out.
76 if (!filter_.test(packet->header().payloadType) && 76 if (!filter_.test(packet->header().payloadType) &&
77 !(use_ssrc_filter_ && packet->header().ssrc != ssrc_)) { 77 !(use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
78 ++rtp_packet_index_;
78 return packet; 79 return packet;
79 } 80 }
80 } 81 }
81 } 82 }
82 return nullptr; 83 return nullptr;
83 } 84 }
84 85
85 int64_t RtcEventLogSource::NextAudioOutputEventMs() { 86 int64_t RtcEventLogSource::NextAudioOutputEventMs() {
86 while (audio_output_index_ < parsed_stream_.GetNumberOfEvents()) { 87 while (audio_output_index_ < parsed_stream_.GetNumberOfEvents()) {
87 if (parsed_stream_.GetEventType(audio_output_index_) == 88 if (parsed_stream_.GetEventType(audio_output_index_) ==
(...skipping 12 matching lines...) Expand all
100 101
101 RtcEventLogSource::RtcEventLogSource() 102 RtcEventLogSource::RtcEventLogSource()
102 : PacketSource(), parser_(RtpHeaderParser::Create()) {} 103 : PacketSource(), parser_(RtpHeaderParser::Create()) {}
103 104
104 bool RtcEventLogSource::OpenFile(const std::string& file_name) { 105 bool RtcEventLogSource::OpenFile(const std::string& file_name) {
105 return parsed_stream_.ParseFile(file_name); 106 return parsed_stream_.ParseFile(file_name);
106 } 107 }
107 108
108 } // namespace test 109 } // namespace test
109 } // namespace webrtc 110 } // namespace webrtc
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698