commit e46bc77e941f12ca24b04cf42fca98e15bc139ee
author xians@webrtc.org <xians@webrtc.org> Fri Oct 10 08:36:56 2014 +0000
committer xians@webrtc.org <xians@webrtc.org> Fri Oct 10 08:36:56 2014 +0000
tree 6ac29b4d4726ded91f1e85e12a0caba7278f9bcc
parent c5593ef1aa332e222a87e634e87fac9487981a4f

Reland 28629004: 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

R=andresp@webrtc.org, andrew@webrtc.org, bjornv@webrtc.org, henrike@webrtc.org, henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7418 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_processing/BUILD.gn

diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index d631adc..2ef2bdf 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -157,6 +157,7 @@
   }
 
   deps += [
+    "../../base:rtc_base_approved",
     "../../common_audio",
     "../../system_wrappers",
   ]

webrtc/modules/audio_processing/audio_processing.gypi

diff --git a/webrtc/modules/audio_processing/audio_processing.gypi b/webrtc/modules/audio_processing/audio_processing.gypi
index 9bbcfae..ce65f64 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_approved',
       '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
       '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
     ],

webrtc/modules/audio_processing/audio_processing_impl.cc

diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index d91cbd2..d6e2906 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/platform_file.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 +717,12 @@
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
+int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
+    rtc::PlatformFile handle) {
+  FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
+  return StartDebugRecording(stream);
+}
+
 int AudioProcessingImpl::StopDebugRecording() {
   CriticalSectionScoped crit_scoped(crit_);
 

webrtc/modules/audio_processing/audio_processing_impl.h

diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index 9753423..0817f47 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -125,6 +125,8 @@
   virtual int StartDebugRecording(
       const char filename[kMaxFilenameSize]) OVERRIDE;
   virtual int StartDebugRecording(FILE* handle) OVERRIDE;
+  virtual int StartDebugRecordingForPlatformFile(
+      rtc::PlatformFile handle) OVERRIDE;
   virtual int StopDebugRecording() OVERRIDE;
   virtual EchoCancellation* echo_cancellation() const OVERRIDE;
   virtual EchoControlMobile* echo_control_mobile() const OVERRIDE;

webrtc/modules/audio_processing/include/audio_processing.h

diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index 30f0d9c..10d2b8b 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/platform_file.h"
 #include "webrtc/common.h"
 #include "webrtc/typedefs.h"
 
@@ -325,6 +326,13 @@
   // 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 StartDebugRecordingForPlatformFile(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;