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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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1064 if (is_dumping_aec_) { | 1063 if (is_dumping_aec_) { |
1065 // Stop dumping AEC when we are dumping. | 1064 // Stop dumping AEC when we are dumping. |
1066 if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() != | 1065 if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() != |
1067 webrtc::AudioProcessing::kNoError) { | 1066 webrtc::AudioProcessing::kNoError) { |
1068 LOG_RTCERR0(StopDebugRecording); | 1067 LOG_RTCERR0(StopDebugRecording); |
1069 } | 1068 } |
1070 is_dumping_aec_ = false; | 1069 is_dumping_aec_ = false; |
1071 } | 1070 } |
1072 } | 1071 } |
1073 | 1072 |
1074 bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file, | |
1075 int64_t max_size_bytes) { | |
1076 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | |
1077 webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog(); | |
1078 if (event_log) { | |
1079 return event_log->StartLogging(file, max_size_bytes); | |
1080 } | |
1081 LOG_RTCERR0(StartRtcEventLog); | |
1082 return false; | |
1083 } | |
1084 | |
1085 void WebRtcVoiceEngine::StopRtcEventLog() { | |
1086 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | |
1087 webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog(); | |
1088 if (event_log) { | |
1089 event_log->StopLogging(); | |
1090 return; | |
1091 } | |
1092 LOG_RTCERR0(StopRtcEventLog); | |
1093 } | |
1094 | |
1095 int WebRtcVoiceEngine::CreateVoEChannel() { | 1073 int WebRtcVoiceEngine::CreateVoEChannel() { |
1096 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 1074 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
1097 return voe_wrapper_->base()->CreateChannel(voe_config_); | 1075 return voe_wrapper_->base()->CreateChannel(voe_config_); |
1098 } | 1076 } |
1099 | 1077 |
1100 webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() { | 1078 webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() { |
1101 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 1079 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
1102 RTC_DCHECK(adm_); | 1080 RTC_DCHECK(adm_); |
1103 return adm_; | 1081 return adm_; |
1104 } | 1082 } |
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2638 } | 2616 } |
2639 } else { | 2617 } else { |
2640 LOG(LS_INFO) << "Stopping playout for channel #" << channel; | 2618 LOG(LS_INFO) << "Stopping playout for channel #" << channel; |
2641 engine()->voe()->base()->StopPlayout(channel); | 2619 engine()->voe()->base()->StopPlayout(channel); |
2642 } | 2620 } |
2643 return true; | 2621 return true; |
2644 } | 2622 } |
2645 } // namespace cricket | 2623 } // namespace cricket |
2646 | 2624 |
2647 #endif // HAVE_WEBRTC_VOICE | 2625 #endif // HAVE_WEBRTC_VOICE |
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