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
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7337 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/audio_processing.gypi b/webrtc/modules/audio_processing/audio_processing.gypi
index 9bbcfae..969fdb1 100644
--- a/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/webrtc/modules/audio_processing/audio_processing.gypi
@@ -9,6 +9,7 @@
{
'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 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;
