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Side by Side Diff: webrtc/media/engine/webrtcvoiceengine.cc

Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Another rebase and accompanying changes. Created 4 years, 5 months ago
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1 /* 1 /*
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2004 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
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21 #include "webrtc/base/arraysize.h" 21 #include "webrtc/base/arraysize.h"
22 #include "webrtc/base/base64.h" 22 #include "webrtc/base/base64.h"
23 #include "webrtc/base/byteorder.h" 23 #include "webrtc/base/byteorder.h"
24 #include "webrtc/base/common.h" 24 #include "webrtc/base/common.h"
25 #include "webrtc/base/constructormagic.h" 25 #include "webrtc/base/constructormagic.h"
26 #include "webrtc/base/helpers.h" 26 #include "webrtc/base/helpers.h"
27 #include "webrtc/base/logging.h" 27 #include "webrtc/base/logging.h"
28 #include "webrtc/base/stringencode.h" 28 #include "webrtc/base/stringencode.h"
29 #include "webrtc/base/stringutils.h" 29 #include "webrtc/base/stringutils.h"
30 #include "webrtc/base/trace_event.h" 30 #include "webrtc/base/trace_event.h"
31 #include "webrtc/call/rtc_event_log.h"
32 #include "webrtc/common.h" 31 #include "webrtc/common.h"
33 #include "webrtc/media/base/audiosource.h" 32 #include "webrtc/media/base/audiosource.h"
34 #include "webrtc/media/base/mediaconstants.h" 33 #include "webrtc/media/base/mediaconstants.h"
35 #include "webrtc/media/base/streamparams.h" 34 #include "webrtc/media/base/streamparams.h"
36 #include "webrtc/media/engine/webrtcmediaengine.h" 35 #include "webrtc/media/engine/webrtcmediaengine.h"
37 #include "webrtc/media/engine/webrtcvoe.h" 36 #include "webrtc/media/engine/webrtcvoe.h"
38 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" 37 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
39 #include "webrtc/modules/audio_processing/include/audio_processing.h" 38 #include "webrtc/modules/audio_processing/include/audio_processing.h"
40 #include "webrtc/system_wrappers/include/field_trial.h" 39 #include "webrtc/system_wrappers/include/field_trial.h"
41 #include "webrtc/system_wrappers/include/trace.h" 40 #include "webrtc/system_wrappers/include/trace.h"
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1033 if (is_dumping_aec_) { 1032 if (is_dumping_aec_) {
1034 // Stop dumping AEC when we are dumping. 1033 // Stop dumping AEC when we are dumping.
1035 if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() != 1034 if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() !=
1036 webrtc::AudioProcessing::kNoError) { 1035 webrtc::AudioProcessing::kNoError) {
1037 LOG_RTCERR0(StopDebugRecording); 1036 LOG_RTCERR0(StopDebugRecording);
1038 } 1037 }
1039 is_dumping_aec_ = false; 1038 is_dumping_aec_ = false;
1040 } 1039 }
1041 } 1040 }
1042 1041
1043 bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file,
1044 int64_t max_size_bytes) {
1045 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1046 webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
1047 if (event_log) {
1048 return event_log->StartLogging(file, max_size_bytes);
1049 }
1050 LOG_RTCERR0(StartRtcEventLog);
1051 return false;
1052 }
1053
1054 void WebRtcVoiceEngine::StopRtcEventLog() {
1055 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1056 webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
1057 if (event_log) {
1058 event_log->StopLogging();
1059 return;
1060 }
1061 LOG_RTCERR0(StopRtcEventLog);
1062 }
1063
1064 int WebRtcVoiceEngine::CreateVoEChannel() { 1042 int WebRtcVoiceEngine::CreateVoEChannel() {
1065 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1043 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1066 return voe_wrapper_->base()->CreateChannel(voe_config_); 1044 return voe_wrapper_->base()->CreateChannel(voe_config_);
1067 } 1045 }
1068 1046
1069 webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() { 1047 webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
1070 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1048 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1071 RTC_DCHECK(adm_); 1049 RTC_DCHECK(adm_);
1072 return adm_; 1050 return adm_;
1073 } 1051 }
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2607 } 2585 }
2608 } else { 2586 } else {
2609 LOG(LS_INFO) << "Stopping playout for channel #" << channel; 2587 LOG(LS_INFO) << "Stopping playout for channel #" << channel;
2610 engine()->voe()->base()->StopPlayout(channel); 2588 engine()->voe()->base()->StopPlayout(channel);
2611 } 2589 }
2612 return true; 2590 return true;
2613 } 2591 }
2614 } // namespace cricket 2592 } // namespace cricket
2615 2593
2616 #endif // HAVE_WEBRTC_VOICE 2594 #endif // HAVE_WEBRTC_VOICE
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