Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(263)

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

Issue 2745473003: Resolve cyclic dependency between audio network adaptor and event log api (Closed)
Patch Set: Revert "Activated checks for rtc_event_log_api" Created 3 years, 8 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/logging/BUILD.gn ('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/modules/audio_coding/audio_network_adaptor/include/audio_networ k_adaptor.h"
17 #include "webrtc/test/gmock.h" 18 #include "webrtc/test/gmock.h"
18 19
19 namespace webrtc { 20 namespace webrtc {
20 21
21 class MockRtcEventLog : public RtcEventLog { 22 class MockRtcEventLog : public RtcEventLog {
22 public: 23 public:
23 MOCK_METHOD2(StartLogging, 24 MOCK_METHOD2(StartLogging,
24 bool(const std::string& file_name, int64_t max_size_bytes)); 25 bool(const std::string& file_name, int64_t max_size_bytes));
25 26
26 MOCK_METHOD2(StartLogging, 27 MOCK_METHOD2(StartLogging,
(...skipping 36 matching lines...) Expand 10 before | Expand all | Expand 10 after
63 64
64 MOCK_METHOD3(LogLossBasedBweUpdate, 65 MOCK_METHOD3(LogLossBasedBweUpdate,
65 void(int32_t bitrate_bps, 66 void(int32_t bitrate_bps,
66 uint8_t fraction_loss, 67 uint8_t fraction_loss,
67 int32_t total_packets)); 68 int32_t total_packets));
68 69
69 MOCK_METHOD2(LogDelayBasedBweUpdate, 70 MOCK_METHOD2(LogDelayBasedBweUpdate,
70 void(int32_t bitrate_bps, BandwidthUsage detector_state)); 71 void(int32_t bitrate_bps, BandwidthUsage detector_state));
71 72
72 MOCK_METHOD1(LogAudioNetworkAdaptation, 73 MOCK_METHOD1(LogAudioNetworkAdaptation,
73 void(const AudioNetworkAdaptor::EncoderRuntimeConfig& config)); 74 void(const AudioEncoderRuntimeConfig& config));
74 75
75 MOCK_METHOD4(LogProbeClusterCreated, 76 MOCK_METHOD4(LogProbeClusterCreated,
76 void(int id, int bitrate_bps, int min_probes, int min_bytes)); 77 void(int id, int bitrate_bps, int min_probes, int min_bytes));
77 78
78 MOCK_METHOD2(LogProbeResultSuccess, void(int id, int bitrate_bps)); 79 MOCK_METHOD2(LogProbeResultSuccess, void(int id, int bitrate_bps));
79 MOCK_METHOD2(LogProbeResultFailure, 80 MOCK_METHOD2(LogProbeResultFailure,
80 void(int id, ProbeFailureReason failure_reason)); 81 void(int id, ProbeFailureReason failure_reason));
81 }; 82 };
82 83
83 } // namespace webrtc 84 } // namespace webrtc
84 85
85 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_MOCK_MOCK_RTC_EVENT_LOG_H_ 86 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_MOCK_MOCK_RTC_EVENT_LOG_H_
OLDNEW
« no previous file with comments | « webrtc/logging/BUILD.gn ('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