Index: talk/media/webrtc/webrtcvoiceengine.cc |
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc |
index eed1195975704aea781d564eefca5a9953b164d4..27254a4f252cb12c6ee9209631cd1d9d1dd3e8c2 100644 |
--- a/talk/media/webrtc/webrtcvoiceengine.cc |
+++ b/talk/media/webrtc/webrtcvoiceengine.cc |
@@ -1259,7 +1259,8 @@ bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm) { |
return true; |
} |
-bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) { |
+bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file, |
+ int64_t max_size_bytes) { |
the sun
2015/11/11 15:54:42
I don't see where max_size_bytes is used in this f
ivoc
2015/11/11 16:44:31
Wow, good find, what was I thinking? :-)
|
FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file); |
if (!aec_dump_file_stream) { |
LOG(LS_ERROR) << "Could not open AEC dump file stream."; |
@@ -1268,8 +1269,8 @@ bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) { |
return false; |
} |
StopAecDump(); |
- if (voe_wrapper_->processing()->StartDebugRecording(aec_dump_file_stream) != |
- webrtc::AudioProcessing::kNoError) { |
+ if (voe_wrapper_->base()->audio_processing()->StartDebugRecording( |
+ aec_dump_file_stream) != webrtc::AudioProcessing::kNoError) { |
LOG_RTCERR0(StartDebugRecording); |
fclose(aec_dump_file_stream); |
return false; |
@@ -1281,8 +1282,8 @@ bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) { |
void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { |
if (!is_dumping_aec_) { |
// Start dumping AEC when we are not dumping. |
- if (voe_wrapper_->processing()->StartDebugRecording( |
- filename.c_str()) != webrtc::AudioProcessing::kNoError) { |
+ if (voe_wrapper_->base()->audio_processing()->StartDebugRecording( |
+ filename.c_str()) != webrtc::AudioProcessing::kNoError) { |
LOG_RTCERR1(StartDebugRecording, filename.c_str()); |
} else { |
is_dumping_aec_ = true; |
@@ -1293,7 +1294,7 @@ void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { |
void WebRtcVoiceEngine::StopAecDump() { |
if (is_dumping_aec_) { |
// Stop dumping AEC when we are dumping. |
- if (voe_wrapper_->processing()->StopDebugRecording() != |
+ if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() != |
webrtc::AudioProcessing::kNoError) { |
LOG_RTCERR0(StopDebugRecording); |
} |