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Unified Diff: talk/media/webrtc/webrtcvoiceengine.cc

Issue 1374253002: Added functions on libjingle API to start and stop the recording of an RtcEventLog. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added functions to start and stop logging. Created 5 years, 3 months ago
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Index: talk/media/webrtc/webrtcvoiceengine.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 5130232bca4ce880ba5fea277fb0b58d7e39fefa..d39654589d2c043d3bf5671590cdb1a0b19060e1 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -50,6 +50,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
+#include "webrtc/call/rtc_event_log.h"
#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
@@ -1336,6 +1337,26 @@ void WebRtcVoiceEngine::StopAecDump() {
}
}
+bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) {
+ FILE* event_log_file = rtc::FdopenPlatformFileForWriting(file);
+ if (!event_log_file) {
+ LOG(LS_ERROR) << "Could not open AEC dump file stream.";
Henrik Grunell WebRTC 2015/10/02 10:23:42 Update text.
ivoc 2015/10/07 14:27:27 Done.
+ if (!rtc::ClosePlatformFile(file))
pthatcher1 2015/10/06 20:49:09 {}s please
ivoc 2015/10/07 14:27:27 Done.
+ LOG(LS_WARNING) << "Could not close file.";
+ return false;
+ }
+ if (voe_wrapper_->codec()->GetEventLog()->StartLogging(event_log_file) != 0) {
+ LOG_RTCERR0(StartLogging);
pthatcher1 2015/10/06 20:49:09 Why does this go through the voice engine? Why do
ivoc 2015/10/07 14:27:27 See other reply. For now VoE owns the logging obje
pthatcher1 2015/10/07 15:59:23 Then can you put a comment in PeerConnectionFactor
ivoc 2015/10/08 16:20:17 Good idea, I added some comments there like you su
+ fclose(event_log_file);
+ return false;
+ }
+ return true;
+}
+
+void WebRtcVoiceEngine::StopRtcEventLog() {
+ voe_wrapper_->codec()->GetEventLog()->StopLogging();
+}
+
int WebRtcVoiceEngine::CreateVoiceChannel(VoEWrapper* voice_engine_wrapper) {
return voice_engine_wrapper->base()->CreateChannel(voe_config_);
}

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