Arun Raghavan 753eada3aa Update audio_processing module
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Pull in third party license file

 * Replace .gypi files with BUILD.gn to keep track of what changes
   upstream

 * Bunch of new filse pulled in as dependencies

 * Won't build yet due to changes needed on top of these
2015-10-15 16:18:45 +05:30

50 lines
1.2 KiB
C++

/*
* Copyright 2014 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/base/platform_file.h"
#if defined(WEBRTC_WIN)
#include <io.h>
#else
#include <unistd.h>
#endif
namespace rtc {
#if defined(WEBRTC_WIN)
const PlatformFile kInvalidPlatformFileValue = INVALID_HANDLE_VALUE;
FILE* FdopenPlatformFileForWriting(PlatformFile file) {
if (file == kInvalidPlatformFileValue)
return NULL;
int fd = _open_osfhandle(reinterpret_cast<intptr_t>(file), 0);
if (fd < 0)
return NULL;
return _fdopen(fd, "w");
}
bool ClosePlatformFile(PlatformFile file) {
return CloseHandle(file) != 0;
}
#else
const PlatformFile kInvalidPlatformFileValue = -1;
FILE* FdopenPlatformFileForWriting(PlatformFile file) {
return fdopen(file, "w");
}
bool ClosePlatformFile(PlatformFile file) {
return close(file);
}
#endif
} // namespace rtc