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Revert 7334 "adding new AEC dump start interface for chrome."

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> adding new AEC dump start interface for chrome.
> 
> This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
> http://msdn.microsoft.com/en-us/library/ms235460.aspx
> 
> Chromium bug:crbug/415935
> TEST=bots
> R=bjornv@webrtc.org, kwiberg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28629004

TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7335 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
xians@webrtc.org
2014-09-30 13:30:05 +00:00
parent 2e417d6428
commit 875206196c
3 changed files with 0 additions and 13 deletions
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6
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -12,7 +12,6 @@
#include <assert.h>
#include "webrtc/base/fileutils.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
@ -717,11 +716,6 @@ int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
int AudioProcessingImpl::StartDebugRecording(rtc::PlatformFile handle) {
FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
return StartDebugRecording(stream);
}
int AudioProcessingImpl::StopDebugRecording() {
CriticalSectionScoped crit_scoped(crit_);