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

Issue 1541633002: 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: Fix to prevent Chrome from breaking. Created 5 years 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 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);
}
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