Chromium Code Reviews| Index: talk/media/webrtc/webrtcvoiceengine.cc |
| diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc |
| index eed1195975704aea781d564eefca5a9953b164d4..67f88c9f1ad0ecc03aad66673f49202e63f6d41f 100644 |
| --- a/talk/media/webrtc/webrtcvoiceengine.cc |
| +++ b/talk/media/webrtc/webrtcvoiceengine.cc |
| @@ -1278,6 +1278,27 @@ bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) { |
| return true; |
| } |
| +bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file, |
| + int max_size_bytes) { |
| + FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file); |
| + if (!aec_dump_file_stream) { |
| + LOG(LS_ERROR) << "Could not open AEC dump file stream."; |
| + if (!rtc::ClosePlatformFile(file)) |
| + LOG(LS_WARNING) << "Could not close file."; |
| + return false; |
| + } |
| + StopAecDump(); |
| + if (voe_wrapper_->processing()->StartDebugRecording(aec_dump_file_stream, |
| + max_size_bytes) != |
| + webrtc::AudioProcessing::kNoError) { |
| + LOG_RTCERR0(StartDebugRecording); |
| + fclose(aec_dump_file_stream); |
| + return false; |
| + } |
| + is_dumping_aec_ = true; |
| + return true; |
| +} |
|
kwiberg-webrtc
2015/10/25 02:29:12
This duplicates a lot of the preceding function. C
the sun
2015/10/26 10:37:20
See previous comment in PeerConnectionFactory: we
ivoc
2015/11/05 13:14:45
Done.
|
| + |
| void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { |
| if (!is_dumping_aec_) { |
| // Start dumping AEC when we are not dumping. |