Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(18)

Side by Side Diff: webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h

Issue 3012233002: Remove #include of rtc_stream_config.h from rtc_event_log.h (Closed)
Patch Set: Rebased Created 3 years, 3 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
« no previous file with comments | « webrtc/call/call.cc ('k') | webrtc/logging/rtc_event_log/rtc_event_log.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
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 10
11 #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_MOCK_MOCK_RTC_EVENT_LOG_H_ 11 #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_MOCK_MOCK_RTC_EVENT_LOG_H_
12 #define WEBRTC_LOGGING_RTC_EVENT_LOG_MOCK_MOCK_RTC_EVENT_LOG_H_ 12 #define WEBRTC_LOGGING_RTC_EVENT_LOG_MOCK_MOCK_RTC_EVENT_LOG_H_
13 13
14 #include <string> 14 #include <string>
15 15
16 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" 16 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
17 #include "webrtc/logging/rtc_event_log/rtc_stream_config.h"
17 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_networ k_adaptor.h" 18 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_networ k_adaptor.h"
18 #include "webrtc/test/gmock.h" 19 #include "webrtc/test/gmock.h"
19 20
20 namespace webrtc { 21 namespace webrtc {
21 22
22 class MockRtcEventLog : public RtcEventLog { 23 class MockRtcEventLog : public RtcEventLog {
23 public: 24 public:
24 MOCK_METHOD2(StartLogging, 25 MOCK_METHOD2(StartLogging,
25 bool(const std::string& file_name, int64_t max_size_bytes)); 26 bool(const std::string& file_name, int64_t max_size_bytes));
26 27
(...skipping 47 matching lines...) Expand 10 before | Expand all | Expand 10 after
74 void(int id, int bitrate_bps, int min_probes, int min_bytes)); 75 void(int id, int bitrate_bps, int min_probes, int min_bytes));
75 76
76 MOCK_METHOD2(LogProbeResultSuccess, void(int id, int bitrate_bps)); 77 MOCK_METHOD2(LogProbeResultSuccess, void(int id, int bitrate_bps));
77 MOCK_METHOD2(LogProbeResultFailure, 78 MOCK_METHOD2(LogProbeResultFailure,
78 void(int id, ProbeFailureReason failure_reason)); 79 void(int id, ProbeFailureReason failure_reason));
79 }; 80 };
80 81
81 } // namespace webrtc 82 } // namespace webrtc
82 83
83 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_MOCK_MOCK_RTC_EVENT_LOG_H_ 84 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_MOCK_MOCK_RTC_EVENT_LOG_H_
OLDNEW
« no previous file with comments | « webrtc/call/call.cc ('k') | webrtc/logging/rtc_event_log/rtc_event_log.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698