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Change WebRtc_Word8 to char

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Review URL: https://webrtc-codereview.appspot.com/405003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1777 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
leozwang@webrtc.org 2012-02-27 19:18:25 +00:00
parent 3e9e0f0497
commit 2559cbf7b7
9 changed files with 25 additions and 25 deletions
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6
src/modules/utility/interface/file_player.h
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@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 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
@ -45,7 +45,7 @@ public:
// API for playing audio from fileName to channel.
// Note: codecInst is used for pre-encoded files.
virtual WebRtc_Word32 StartPlayingFile(
const WebRtc_Word8* fileName,
const char* fileName,
bool loop,
WebRtc_UWord32 startPosition,
float volumeScaling,
@ -83,7 +83,7 @@ public:
virtual WebRtc_Word32 TimeUntilNextVideoFrame() { return -1;}
virtual WebRtc_Word32 StartPlayingVideoFile(
const WebRtc_Word8* /*fileName*/,
const char* /*fileName*/,
bool /*loop*/,
bool /*videoOnly*/) { return -1;}

6
src/modules/utility/interface/file_recorder.h
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@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 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
@ -38,7 +38,7 @@ public:
virtual FileFormats RecordingFileFormat() const = 0;
virtual WebRtc_Word32 StartRecordingAudioFile(
const WebRtc_Word8* fileName,
const char* fileName,
const CodecInst& codecInst,
WebRtc_UWord32 notification,
ACMAMRPackingFormat amrFormat = AMRFileStorage) = 0;
@ -71,7 +71,7 @@ public:
// specifies the amr/amrwb storage format.
// Note: the file format is AVI.
virtual WebRtc_Word32 StartRecordingVideoFile(
const WebRtc_Word8* fileName,
const char* fileName,
const CodecInst& audioCodecInst,
const VideoCodec& videoCodecInst,
ACMAMRPackingFormat amrFormat = AMRFileStorage,

4
src/modules/utility/interface/rtp_dump.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 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
@ -31,7 +31,7 @@ public:
// Open the file fileNameUTF8 for writing RTP/RTCP packets.
// Note: this API also adds the rtpplay header.
virtual WebRtc_Word32 Start(const WebRtc_Word8* fileNameUTF8) = 0;
virtual WebRtc_Word32 Start(const char* fileNameUTF8) = 0;
// Close the existing file. No more packets will be recorded.
virtual WebRtc_Word32 Stop() = 0;

4
src/modules/utility/source/file_player_impl.cc
View File

@ -228,7 +228,7 @@ WebRtc_Word32 FilePlayerImpl::SetAudioScaling(float scaleFactor)
return -1;
}
WebRtc_Word32 FilePlayerImpl::StartPlayingFile(const WebRtc_Word8* fileName,
WebRtc_Word32 FilePlayerImpl::StartPlayingFile(const char* fileName,
bool loop,
WebRtc_UWord32 startPosition,
float volumeScaling,
@ -491,7 +491,7 @@ VideoFilePlayerImpl::~VideoFilePlayerImpl()
}
WebRtc_Word32 VideoFilePlayerImpl::StartPlayingVideoFile(
const WebRtc_Word8* fileName,
const char* fileName,
bool loop,
bool videoOnly)
{

6
src/modules/utility/source/file_player_impl.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 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
@ -39,7 +39,7 @@ public:
const WebRtc_UWord32 frequencyInHz);
virtual WebRtc_Word32 RegisterModuleFileCallback(FileCallback* callback);
virtual WebRtc_Word32 StartPlayingFile(
const WebRtc_Word8* fileName,
const char* fileName,
bool loop,
WebRtc_UWord32 startPosition,
float volumeScaling,
@ -90,7 +90,7 @@ public:
// FilePlayer functions.
virtual WebRtc_Word32 TimeUntilNextVideoFrame();
virtual WebRtc_Word32 StartPlayingVideoFile(const WebRtc_Word8* fileName,
virtual WebRtc_Word32 StartPlayingVideoFile(const char* fileName,
bool loop,
bool videoOnly);
virtual WebRtc_Word32 StopPlayingFile();

4
src/modules/utility/source/file_recorder_impl.cc
View File

@ -95,7 +95,7 @@ WebRtc_Word32 FileRecorderImpl::RegisterModuleFileCallback(
}
WebRtc_Word32 FileRecorderImpl::StartRecordingAudioFile(
const WebRtc_Word8* fileName,
const char* fileName,
const CodecInst& codecInst,
WebRtc_UWord32 notificationTimeMs,
ACMAMRPackingFormat amrFormat)
@ -405,7 +405,7 @@ AviRecorder::~AviRecorder( )
}
WebRtc_Word32 AviRecorder::StartRecordingVideoFile(
const WebRtc_Word8* fileName,
const char* fileName,
const CodecInst& audioCodecInst,
const VideoCodec& videoCodecInst,
ACMAMRPackingFormat amrFormat,

8
src/modules/utility/source/file_recorder_impl.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 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
@ -50,7 +50,7 @@ public:
virtual WebRtc_Word32 RegisterModuleFileCallback(FileCallback* callback);
virtual FileFormats RecordingFileFormat() const;
virtual WebRtc_Word32 StartRecordingAudioFile(
const WebRtc_Word8* fileName,
const char* fileName,
const CodecInst& codecInst,
WebRtc_UWord32 notificationTimeMs,
ACMAMRPackingFormat amrFormat = AMRFileStorage);
@ -66,7 +66,7 @@ public:
const AudioFrame& frame,
const TickTime* playoutTS = NULL);
virtual WebRtc_Word32 StartRecordingVideoFile(
const WebRtc_Word8* fileName,
const char* fileName,
const CodecInst& audioCodecInst,
const VideoCodec& videoCodecInst,
ACMAMRPackingFormat amrFormat = AMRFileStorage,
@ -112,7 +112,7 @@ public:
// FileRecorder functions.
virtual WebRtc_Word32 StartRecordingVideoFile(
const WebRtc_Word8* fileName,
const char* fileName,
const CodecInst& audioCodecInst,
const VideoCodec& videoCodecInst,
ACMAMRPackingFormat amrFormat = AMRFileStorage,

8
src/modules/utility/source/rtp_dump_impl.cc
View File

@ -39,7 +39,7 @@
#endif // defined(_DEBUG) && defined(_WIN32)
namespace webrtc {
const WebRtc_Word8 RTPFILE_VERSION[] = "1.0";
const char RTPFILE_VERSION[] = "1.0";
const WebRtc_UWord32 MAX_UWORD32 = 0xffffffff;
// This stucture is specified in the rtpdump documentation.
@ -87,7 +87,7 @@ RtpDumpImpl::~RtpDumpImpl()
WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s deleted", __FUNCTION__);
}
WebRtc_Word32 RtpDumpImpl::Start(const WebRtc_Word8* fileNameUTF8)
WebRtc_Word32 RtpDumpImpl::Start(const char* fileNameUTF8)
{
if (fileNameUTF8 == NULL)
@ -109,7 +109,7 @@ WebRtc_Word32 RtpDumpImpl::Start(const WebRtc_Word8* fileNameUTF8)
_startTime = GetTimeInMS();
// All rtp dump files start with #!rtpplay.
WebRtc_Word8 magic[16];
char magic[16];
sprintf(magic, "#!rtpplay%s \n", RTPFILE_VERSION);
if (_file.WriteText(magic) == -1)
{
@ -125,7 +125,7 @@ WebRtc_Word32 RtpDumpImpl::Start(const WebRtc_Word8* fileNameUTF8)
// as Wireshark since it makes more sense.
// http://wiki.wireshark.org/rtpdump explains that an additional 2 bytes
// of padding should be added to the header.
WebRtc_Word8 dummyHdr[16];
char dummyHdr[16];
memset(dummyHdr, 0, 16);
if (!_file.Write(dummyHdr, sizeof(dummyHdr)))
{

4
src/modules/utility/source/rtp_dump_impl.h
View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 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
@ -22,7 +22,7 @@ public:
RtpDumpImpl();
virtual ~RtpDumpImpl();
virtual WebRtc_Word32 Start(const WebRtc_Word8* fileNameUTF8);
virtual WebRtc_Word32 Start(const char* fileNameUTF8);
virtual WebRtc_Word32 Stop();
virtual bool IsActive() const;
virtual WebRtc_Word32 DumpPacket(const WebRtc_UWord8* packet,