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Reland 28629004: adding new AEC dump start interface for chrome.

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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
This commit is contained in:
xians@webrtc.org
2014-10-10 08:36:56 +00:00
parent c5593ef1aa
commit e46bc77e94
11 changed files with 118 additions and 52 deletions
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1
webrtc/modules/audio_processing/BUILD.gn
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@ -157,6 +157,7 @@ source_set("audio_processing") {
}
deps += [
"../../base:rtc_base_approved",
"../../common_audio",
"../../system_wrappers",
]

1
webrtc/modules/audio_processing/audio_processing.gypi
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@ -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',
],

7
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -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 @@ int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
#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_);

2
webrtc/modules/audio_processing/audio_processing_impl.h
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@ -125,6 +125,8 @@ class AudioProcessingImpl : public AudioProcessing {
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;

8
webrtc/modules/audio_processing/include/audio_processing.h
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@ -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 @@ 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 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;