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
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7334 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index d91cbd2..6806523 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -12,6 +12,7 @@
#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"
@@ -716,6 +717,11 @@
#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 9753423..d012e7f 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -125,6 +125,7 @@
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 30f0d9c..53157ae 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -14,6 +14,7 @@
#include <stddef.h> // size_t
#include <stdio.h> // FILE
+#include "webrtc/base/fileutils.h"
#include "webrtc/common.h"
#include "webrtc/typedefs.h"
@@ -325,6 +326,11 @@
// 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;
