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Issue 1537213002: Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2004 Google Inc. 3 * Copyright 2004 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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1004 adm_->Release(); 1004 adm_->Release();
1005 adm_ = NULL; 1005 adm_ = NULL;
1006 } 1006 }
1007 if (adm) { 1007 if (adm) {
1008 adm_ = adm; 1008 adm_ = adm;
1009 adm_->AddRef(); 1009 adm_->AddRef();
1010 } 1010 }
1011 return true; 1011 return true;
1012 } 1012 }
1013 1013
1014 bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file, 1014 bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) {
1015 int64_t max_size_bytes) {
1016 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1015 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1017 FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file); 1016 FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
1018 if (!aec_dump_file_stream) { 1017 if (!aec_dump_file_stream) {
1019 LOG(LS_ERROR) << "Could not open AEC dump file stream."; 1018 LOG(LS_ERROR) << "Could not open AEC dump file stream.";
1020 if (!rtc::ClosePlatformFile(file)) 1019 if (!rtc::ClosePlatformFile(file))
1021 LOG(LS_WARNING) << "Could not close file."; 1020 LOG(LS_WARNING) << "Could not close file.";
1022 return false; 1021 return false;
1023 } 1022 }
1024 StopAecDump(); 1023 StopAecDump();
1025 if (voe_wrapper_->base()->audio_processing()->StartDebugRecording( 1024 if (voe_wrapper_->processing()->StartDebugRecording(aec_dump_file_stream) !=
1026 aec_dump_file_stream, max_size_bytes) !=
1027 webrtc::AudioProcessing::kNoError) { 1025 webrtc::AudioProcessing::kNoError) {
1028 LOG_RTCERR0(StartDebugRecording); 1026 LOG_RTCERR0(StartDebugRecording);
1029 fclose(aec_dump_file_stream); 1027 fclose(aec_dump_file_stream);
1030 return false; 1028 return false;
1031 } 1029 }
1032 is_dumping_aec_ = true; 1030 is_dumping_aec_ = true;
1033 return true; 1031 return true;
1034 } 1032 }
1035 1033
1036 void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { 1034 void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
1037 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1035 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1038 if (!is_dumping_aec_) { 1036 if (!is_dumping_aec_) {
1039 // Start dumping AEC when we are not dumping. 1037 // Start dumping AEC when we are not dumping.
1040 if (voe_wrapper_->base()->audio_processing()->StartDebugRecording( 1038 if (voe_wrapper_->processing()->StartDebugRecording(
1041 filename.c_str(), -1) != webrtc::AudioProcessing::kNoError) { 1039 filename.c_str()) != webrtc::AudioProcessing::kNoError) {
1042 LOG_RTCERR1(StartDebugRecording, filename.c_str()); 1040 LOG_RTCERR1(StartDebugRecording, filename.c_str());
1043 } else { 1041 } else {
1044 is_dumping_aec_ = true; 1042 is_dumping_aec_ = true;
1045 } 1043 }
1046 } 1044 }
1047 } 1045 }
1048 1046
1049 void WebRtcVoiceEngine::StopAecDump() { 1047 void WebRtcVoiceEngine::StopAecDump() {
1050 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1048 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1051 if (is_dumping_aec_) { 1049 if (is_dumping_aec_) {
1052 // Stop dumping AEC when we are dumping. 1050 // Stop dumping AEC when we are dumping.
1053 if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() != 1051 if (voe_wrapper_->processing()->StopDebugRecording() !=
1054 webrtc::AudioProcessing::kNoError) { 1052 webrtc::AudioProcessing::kNoError) {
1055 LOG_RTCERR0(StopDebugRecording); 1053 LOG_RTCERR0(StopDebugRecording);
1056 } 1054 }
1057 is_dumping_aec_ = false; 1055 is_dumping_aec_ = false;
1058 } 1056 }
1059 } 1057 }
1060 1058
1061 bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) { 1059 bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) {
1062 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1060 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1063 return voe_wrapper_->codec()->GetEventLog()->StartLogging(file); 1061 return voe_wrapper_->codec()->GetEventLog()->StartLogging(file);
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2502 } 2500 }
2503 } else { 2501 } else {
2504 LOG(LS_INFO) << "Stopping playout for channel #" << channel; 2502 LOG(LS_INFO) << "Stopping playout for channel #" << channel;
2505 engine()->voe()->base()->StopPlayout(channel); 2503 engine()->voe()->base()->StopPlayout(channel);
2506 } 2504 }
2507 return true; 2505 return true;
2508 } 2506 }
2509 } // namespace cricket 2507 } // namespace cricket
2510 2508
2511 #endif // HAVE_WEBRTC_VOICE 2509 #endif // HAVE_WEBRTC_VOICE
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