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

Issue 1413483003: 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: Processed first batch of reviews. Created 5 years, 1 month 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 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);
}

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