From 79a714810840c56bd33afd9a240f6f4c873d6678 Mon Sep 17 00:00:00 2001 From: "xians@webrtc.org" Date: Tue, 30 Sep 2014 15:29:13 +0000 Subject: [PATCH] Revert 7337 "Reland 28629004: adding new AEC dump start interfac..." > Reland 28629004: adding new AEC dump start interface for chrome > > 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/27639004 TBR=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22849004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7342 4adac7df-926f-26a2-2b94-8c16560cd09d --- webrtc/modules/audio_processing/audio_processing.gypi | 1 - webrtc/modules/audio_processing/audio_processing_impl.cc | 6 ------ webrtc/modules/audio_processing/audio_processing_impl.h | 1 - webrtc/modules/audio_processing/include/audio_processing.h | 6 ------ 4 files changed, 14 deletions(-) diff --git a/webrtc/modules/audio_processing/audio_processing.gypi b/webrtc/modules/audio_processing/audio_processing.gypi index 969fdb12c3..9bbcfae7f8 100644 --- a/webrtc/modules/audio_processing/audio_processing.gypi +++ b/webrtc/modules/audio_processing/audio_processing.gypi @@ -9,7 +9,6 @@ { 'variables': { 'audio_processing_dependencies': [ - '<(webrtc_root)/base/base.gyp:rtc_base', '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', ], diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc index 6806523221..d91cbd2fd3 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl.cc +++ b/webrtc/modules/audio_processing/audio_processing_impl.cc @@ -12,7 +12,6 @@ #include -#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_); diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h index d012e7f0ea..9753423d6d 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl.h +++ b/webrtc/modules/audio_processing/audio_processing_impl.h @@ -125,7 +125,6 @@ class AudioProcessingImpl : public AudioProcessing { virtual int StartDebugRecording( const char filename[kMaxFilenameSize]) OVERRIDE; virtual int StartDebugRecording(FILE* handle) OVERRIDE; - virtual int StartDebugRecording(rtc::PlatformFile handle) OVERRIDE; virtual int StopDebugRecording() OVERRIDE; virtual EchoCancellation* echo_cancellation() const OVERRIDE; virtual EchoControlMobile* echo_control_mobile() const OVERRIDE; diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h index 53157ae66f..30f0d9c5d9 100644 --- a/webrtc/modules/audio_processing/include/audio_processing.h +++ b/webrtc/modules/audio_processing/include/audio_processing.h @@ -14,7 +14,6 @@ #include // size_t #include // FILE -#include "webrtc/base/fileutils.h" #include "webrtc/common.h" #include "webrtc/typedefs.h" @@ -326,11 +325,6 @@ class AudioProcessing { // of |handle| and closes it at StopDebugRecording(). virtual int StartDebugRecording(FILE* handle) = 0; - // Same as above but uses an existing PlatformFile handle. Takes ownership - // of |handle| and closes it at StopDebugRecording(). - // TODO(xians): Make this interface pure virtual. - virtual int StartDebugRecording(rtc::PlatformFile handle) { return -1; } - // Stops recording debugging information, and closes the file. Recording // cannot be resumed in the same file (without overwriting it). virtual int StopDebugRecording() = 0;