Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(146)

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

Issue 2380683005: Moved RtcEventLog files from call/ to logging/ (new top level dir) (Closed)
Patch Set: Rebase to master Created 4 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
11 #include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h" 11 #include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
12 12
13 #include <assert.h> 13 #include <assert.h>
14 #include <string.h> 14 #include <string.h>
15 #include <iostream> 15 #include <iostream>
16 #include <limits> 16 #include <limits>
17 17
18 #include "webrtc/base/checks.h" 18 #include "webrtc/base/checks.h"
19 #include "webrtc/call.h" 19 #include "webrtc/call.h"
20 #include "webrtc/call/rtc_event_log.h"
21 #include "webrtc/modules/audio_coding/neteq/tools/packet.h" 20 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
22 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" 21 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
23 22
24 23
25 namespace webrtc { 24 namespace webrtc {
26 namespace test { 25 namespace test {
27 26
28 RtcEventLogSource* RtcEventLogSource::Create(const std::string& file_name) { 27 RtcEventLogSource* RtcEventLogSource::Create(const std::string& file_name) {
29 RtcEventLogSource* source = new RtcEventLogSource(); 28 RtcEventLogSource* source = new RtcEventLogSource();
30 RTC_CHECK(source->OpenFile(file_name)); 29 RTC_CHECK(source->OpenFile(file_name));
(...skipping 63 matching lines...) Expand 10 before | Expand all | Expand 10 after
94 93
95 RtcEventLogSource::RtcEventLogSource() 94 RtcEventLogSource::RtcEventLogSource()
96 : PacketSource(), parser_(RtpHeaderParser::Create()) {} 95 : PacketSource(), parser_(RtpHeaderParser::Create()) {}
97 96
98 bool RtcEventLogSource::OpenFile(const std::string& file_name) { 97 bool RtcEventLogSource::OpenFile(const std::string& file_name) {
99 return parsed_stream_.ParseFile(file_name); 98 return parsed_stream_.ParseFile(file_name);
100 } 99 }
101 100
102 } // namespace test 101 } // namespace test
103 } // namespace webrtc 102 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h ('k') | webrtc/modules/bitrate_controller/DEPS » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698