Index: talk/media/webrtc/webrtcvoiceengine.cc |
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc |
index 9eee2af20219d710bfad79104c047ad905f33c76..38b6c54b73c050fcc6382e1544c81004f9c19b00 100644 |
--- a/talk/media/webrtc/webrtcvoiceengine.cc |
+++ b/talk/media/webrtc/webrtcvoiceengine.cc |
@@ -1011,7 +1011,8 @@ 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) { |
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file); |
if (!aec_dump_file_stream) { |
@@ -1021,7 +1022,8 @@ bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) { |
return false; |
} |
StopAecDump(); |
- if (voe_wrapper_->processing()->StartDebugRecording(aec_dump_file_stream) != |
+ if (voe_wrapper_->base()->audio_processing()->StartDebugRecording( |
+ aec_dump_file_stream, max_size_bytes) != |
webrtc::AudioProcessing::kNoError) { |
LOG_RTCERR0(StartDebugRecording); |
fclose(aec_dump_file_stream); |
@@ -1035,8 +1037,8 @@ void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { |
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
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(), -1) != webrtc::AudioProcessing::kNoError) { |
LOG_RTCERR1(StartDebugRecording, filename.c_str()); |
} else { |
is_dumping_aec_ = true; |
@@ -1048,7 +1050,7 @@ void WebRtcVoiceEngine::StopAecDump() { |
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
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); |
} |