view tremor/ivorbiscodec.h @ 23978:ef6e50c3c172

Revert setting audio output channel count for FFmpeg The FFmpeg API needs to be fixed before this can be done sanely. ffdca wants the desired output channel count to be set in avctx->channels. Unfortunately it also completely fails if the requested number of channels is not available rather than returning a different amount (if 6 channels are requested we'd probably rather use stereo than fail completely). ffvorbis ignores caller-set values in avctx->channels. It writes the channel count there once during init. This means the caller can only set the count before init because later there would be no indication whether the channel count in avctx reflects real output. ffwma requires the caller to supply the encoded channel count in avctx->channels during init or it fails. So it is not possible to set a different number of desired output channels there before init either.

/********************************************************************
 *                                                                  *
 * 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