Mercurial > mplayer.hg
view tremor/ivorbiscodec.h @ 30201:786d035899d2
mark ffqclp as working in codecs.conf, works on all my samples
author | compn |
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date | Fri, 08 Jan 2010 18:44:34 +0000 |
parents | e83eef58b30a |
children |
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/******************************************************************** * * * THIS FILE IS PART OF THE OggVorbis 'TREMOR' CODEC SOURCE CODE. * * * * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * * * * THE OggVorbis 'TREMOR' SOURCE CODE IS (C) COPYRIGHT 1994-2002 * * BY THE Xiph.Org FOUNDATION http://www.xiph.org/ * * * ******************************************************************** function: libvorbis codec headers ********************************************************************/ #ifndef _vorbis_codec_h_ #define _vorbis_codec_h_ #ifdef __cplusplus extern "C" { #endif /* __cplusplus */ #include "ogg.h" typedef struct vorbis_info{ int version; int channels; long rate; /* The below bitrate declarations are *hints*. Combinations of the three values carry the following implications: all three set to the same value: implies a fixed rate bitstream only nominal set: implies a VBR stream that averages the nominal bitrate. No hard upper/lower limit upper and or lower set: implies a VBR bitstream that obeys the bitrate limits. nominal may also be set to give a nominal rate. none set: the coder does not care to speculate. */ long bitrate_upper; long bitrate_nominal; long bitrate_lower; long bitrate_window; void *codec_setup; } vorbis_info; /* vorbis_dsp_state buffers the current vorbis audio analysis/synthesis state. The DSP state belongs to a specific logical bitstream ****************************************************/ typedef struct vorbis_dsp_state{ int analysisp; vorbis_info *vi; ogg_int32_t **pcm; ogg_int32_t **pcmret; int pcm_storage; int pcm_current; int pcm_returned; int preextrapolate; int eofflag; long lW; long W; long nW; long centerW; ogg_int64_t granulepos; ogg_int64_t sequence; void *backend_state; } vorbis_dsp_state; typedef struct vorbis_block{ /* necessary stream state for linking to the framing abstraction */ ogg_int32_t **pcm; /* this is a pointer into local storage */ oggpack_buffer opb; long lW; long W; long nW; int pcmend; int mode; int eofflag; ogg_int64_t granulepos; ogg_int64_t sequence; vorbis_dsp_state *vd; /* For read-only access of configuration */ /* local storage to avoid remallocing; it's up to the mapping to structure it */ void *localstore; long localtop; long localalloc; long totaluse; struct alloc_chain *reap; } vorbis_block; /* vorbis_block is a single block of data to be processed as part of the analysis/synthesis stream; it belongs to a specific logical bitstream, but is independant from other vorbis_blocks belonging to that logical bitstream. *************************************************/ struct alloc_chain{ void *ptr; struct alloc_chain *next; }; /* vorbis_info contains all the setup information specific to the specific compression/decompression mode in progress (eg, psychoacoustic settings, channel setup, options, codebook etc). vorbis_info and substructures are in backends.h. *********************************************************************/ /* the comments are not part of vorbis_info so that vorbis_info can be static storage */ typedef struct vorbis_comment{ /* unlimited user comment fields. libvorbis writes 'libvorbis' whatever vendor is set to in encode */ char **user_comments; int *comment_lengths; int comments; char *vendor; } vorbis_comment; /* libvorbis encodes in two abstraction layers; first we perform DSP and produce a packet (see docs/analysis.txt). The packet is then coded into a framed OggSquish bitstream by the second layer (see docs/framing.txt). Decode is the reverse process; we sync/frame the bitstream and extract individual packets, then decode the packet back into PCM audio. The extra framing/packetizing is used in streaming formats, such as files. Over the net (such as with UDP), the framing and packetization aren't necessary as they're provided by the transport and the streaming layer is not used */ /* Vorbis PRIMITIVES: general ***************************************/ extern void vorbis_info_init(vorbis_info *vi); extern void vorbis_info_clear(vorbis_info *vi); extern int vorbis_info_blocksize(vorbis_info *vi,int zo); extern void vorbis_comment_init(vorbis_comment *vc); extern void vorbis_comment_add(vorbis_comment *vc, char *comment); extern void vorbis_comment_add_tag(vorbis_comment *vc, char *tag, char *contents); extern char *vorbis_comment_query(vorbis_comment *vc, char *tag, int count); extern int vorbis_comment_query_count(vorbis_comment *vc, char *tag); extern void vorbis_comment_clear(vorbis_comment *vc); extern int vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb); extern int vorbis_block_clear(vorbis_block *vb); extern void vorbis_dsp_clear(vorbis_dsp_state *v); /* Vorbis PRIMITIVES: synthesis layer *******************************/ extern int vorbis_synthesis_headerin(vorbis_info *vi,vorbis_comment *vc, ogg_packet *op); extern int vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi); extern int vorbis_synthesis(vorbis_block *vb,ogg_packet *op); extern int vorbis_synthesis_blockin(vorbis_dsp_state *v,vorbis_block *vb); extern int vorbis_synthesis_pcmout(vorbis_dsp_state *v,ogg_int32_t ***pcm); extern int vorbis_synthesis_read(vorbis_dsp_state *v,int samples); extern long vorbis_packet_blocksize(vorbis_info *vi,ogg_packet *op); /* Vorbis ERRORS and return codes ***********************************/ #define OV_FALSE -1 #define OV_EOF -2 #define OV_HOLE -3 #define OV_EREAD -128 #define OV_EFAULT -129 #define OV_EIMPL -130 #define OV_EINVAL -131 #define OV_ENOTVORBIS -132 #define OV_EBADHEADER -133 #define OV_EVERSION -134 #define OV_ENOTAUDIO -135 #define OV_EBADPACKET -136 #define OV_EBADLINK -137 #define OV_ENOSEEK -138 #ifdef __cplusplus } #endif /* __cplusplus */ #endif