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Remove avi recorder and corresponding enable_video flags.

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R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42099004

Cr-Commit-Position: refs/heads/master@{#8554}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8554 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andresp@webrtc.org
2015-03-02 13:07:02 +00:00
parent f56c162310
commit e8f50df6b9
28 changed files with 61 additions and 4212 deletions
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63
webrtc/modules/media_file/source/media_file_utility.h
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@ -18,7 +18,6 @@
#include "webrtc/modules/media_file/interface/media_file_defines.h"
namespace webrtc {
class AviFile;
class InStream;
class OutStream;
@ -29,61 +28,6 @@ public:
ModuleFileUtility(const int32_t id);
~ModuleFileUtility();
#ifdef WEBRTC_MODULE_UTILITY_VIDEO
// Open the file specified by fileName for reading (relative path is
// allowed). If loop is true the file will be played until StopPlaying() is
// called. When end of file is reached the file is read from the start.
// Only video will be read if videoOnly is true.
int32_t InitAviReading(const char* fileName, bool videoOnly, bool loop);
// Put 10-60ms of audio data from file into the outBuffer depending on
// codec frame size. bufferLengthInBytes indicates the size of outBuffer.
// The return value is the number of bytes written to audioBuffer.
// Note: This API only play mono audio but can be used on file containing
// audio with more channels (in which case the audio will be coverted to
// mono).
int32_t ReadAviAudioData(int8_t* outBuffer,
size_t bufferLengthInBytes);
// Put one video frame into outBuffer. bufferLengthInBytes indicates the
// size of outBuffer.
// The return value is the number of bytes written to videoBuffer.
int32_t ReadAviVideoData(int8_t* videoBuffer,
size_t bufferLengthInBytes);
// Open/create the file specified by fileName for writing audio/video data
// (relative path is allowed). codecInst specifies the encoding of the audio
// data. videoCodecInst specifies the encoding of the video data. Only video
// data will be recorded if videoOnly is true.
int32_t InitAviWriting(const char* filename,
const CodecInst& codecInst,
const VideoCodec& videoCodecInst,
const bool videoOnly);
// Write one audio frame, i.e. the bufferLengthinBytes first bytes of
// audioBuffer, to file. The audio frame size is determined by the
// codecInst.pacsize parameter of the last sucessfull
// InitAviWriting(..) call.
// Note: bufferLength must be exactly one frame.
int32_t WriteAviAudioData(const int8_t* audioBuffer,
size_t bufferLengthInBytes);
// Write one video frame, i.e. the bufferLength first bytes of videoBuffer,
// to file.
// Note: videoBuffer can contain encoded data. The codec used must be the
// same as what was specified by videoCodecInst for the last successfull
// InitAviWriting(..) call. The videoBuffer must contain exactly
// one video frame.
int32_t WriteAviVideoData(const int8_t* videoBuffer,
size_t bufferLengthInBytes);
// Stop recording to file or stream.
int32_t CloseAviFile();
int32_t VideoCodecInst(VideoCodec& codecInst);
#endif // #ifdef WEBRTC_MODULE_UTILITY_VIDEO
// Prepare for playing audio from stream.
// startPointMs and stopPointMs, unless zero, specify what part of the file
// should be read. From startPointMs ms to stopPointMs ms.
@ -335,13 +279,6 @@ private:
// Scratch buffer used for turning stereo audio to mono.
uint8_t _tempData[WAV_MAX_BUFFER_SIZE];
#ifdef WEBRTC_MODULE_UTILITY_VIDEO
AviFile* _aviAudioInFile;
AviFile* _aviVideoInFile;
AviFile* _aviOutFile;
VideoCodec _videoCodec;
#endif
};
} // namespace webrtc
#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_