view libmpcodecs/ad_sample.c @ 10411:80dbdfe86c5b

I attach a fix to the problem described in: http://mplayerhq.hu/pipermail/mplayer-dev-eng/2003-July/019494.html The bug came out to be that sws_rgb2rgb_init was called, but only after the critical step in which ws.c copied the relevant function pointer to wsConvFunc. Someone deserves 1000l for this. Maybe we want to preinit the function pointers so that they will print something like "Call to an rgb2rgb function without calling to sws_rgb2rgb_init first. Please report." - this bug wasn't discovered since the function pointers were NULL, and the rest of the cde uses "wsConvFunc" only if it is not NULL. Raindel Shachar <[e-mail]>

// SAMPLE audio decoder - you can use this file as template when creating new codec!

#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>

#include "config.h"
#include "ad_internal.h"

static ad_info_t info =  {
	"Sample audio decoder",  // name of the driver
	"sample",    // driver name. should be the same as filename without ad_
	"A'rpi",     // writer/maintainer of _this_ file
	"",          // writer/maintainer/site of the _codec_
	""           // comments
};

LIBAD_EXTERN(sample)

#include "libsample/sample.h" // include your codec's .h files here

static int preinit(sh_audio_t *sh){
  // let's check if the driver is available, return 0 if not.
  // (you should do that if you use external lib(s) which is optional)
  ...
  
  // there are default values set for buffering, but you can override them:
  
  // minimum output buffer size (should be the uncompressed max. frame size)
  sh->audio_out_minsize=4*2*1024; // in this sample, we assume max 4 channels,
                                  // 2 bytes/sample and 1024 samples/frame
				  // Default: 8192
  
  // minimum input buffer size (set only if you need input buffering)
  // (should be the max compressed frame size)
  sh->audio_in_minsize=2048; // Default: 0 (no input buffer)
  
  // if you set audio_in_minsize non-zero, the buffer will be allocated
  // before the init() call by the core, and you can access it via
  // pointer: sh->audio_in_buffer
  // it will free'd after uninit(), so you don't have to use malloc/free here!

  // the next few parameters define the audio format (channels, sample type,
  // in/out bitrate etc.). it's OK to move these to init() if you can set
  // them only after some initialization:
  
  sh->samplesize=2;              // bytes (not bits!) per sample per channel
  sh->channels=2;                // number of channels
  sh->samplerate=44100;          // samplerate
  sh->sample_format=AFMT_S16_LE; // sample format, see libao2/afmt.h
  
  sh->i_bps=64000/8; // input data rate (compressed bytes per second)
  // Note: if you have VBR or unknown input rate, set it to some common or
  // average value, instead of zero. it's used to predict time delay of
  // buffered compressed bytes, so it must be more-or-less real!
  
//sh->o_bps=...     // output data rate (uncompressed bytes per second)
  // Note: you DON'T need to set o_bps in most cases, as it defaults to:
  //   sh->samplesize*sh->channels*sh->samplerate;

  // for constant rate compressed QuickTime (.mov files) codecs you MUST
  // set the compressed and uncompressed packet size (used by the demuxer):
  sh->ds->ss_mul = 34; // compressed packet size
  sh->ds->ss_div = 64; // samples per packet
  
  return 1; // return values: 1=OK 0=ERROR
}

static int init(sh_audio_t *sh_audio){
  // initialize the decoder, set tables etc...

  // you can store HANDLE or private struct pointer at sh->context
  // you can access WAVEFORMATEX header at sh->wf
  
  // set sample format/rate parameters if you didn't do it in preinit() yet.

  return 1; // return values: 1=OK 0=ERROR
}

static void uninit(sh_audio_t *sh){
  // uninit the decoder etc...
  // again: you don't have to free() a_in_buffer here! it's done by the core.
}

static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen){

  // audio decoding. the most important thing :)
  // parameters you get:
  //  buf = pointer to the output buffer, you have to store uncompressed 
  //        samples there
  //  minlen = requested minimum size (in bytes!) of output. it's just a
  //        _recommendation_, you can decode more or less, it just tell you that
  //        the caller process needs 'minlen' bytes. if it gets less, it will
  //        call decode_audio() again.
  //  maxlen = maximum size (bytes) of output. you MUST NOT write more to the
  //        buffer, it's the upper-most limit!
  //        note: maxlen will be always greater or equal to sh->audio_out_minsize

  // now, let's decode...  
  
  // you can read the compressed stream using the demux stream functions:
  //  demux_read_data(sh->ds, buffer, length) - read 'length' bytes to 'buffer'
  //  ds_get_packet(sh->ds, &buffer) - set ptr buffer to next data packet
  // (both func return number of bytes or 0 for error)

  return len; // return value: number of _bytes_ written to output buffer,
              // or -1 for EOF (or uncorrectable error)
}

static int control(sh_audio_t *sh,int cmd,void* arg, ...){
    // various optional functions you MAY implement:
    switch(cmd){
      case ADCTRL_RESYNC_STREAM:
        // it is called once after seeking, to resync.
	// Note: sh_audio->a_in_buffer_len=0; is done _before_ this call!
	...
	return CONTROL_TRUE;
      case ADCTRL_SKIP_FRAME:
        // it is called to skip (jump over) small amount (1/10 sec or 1 frame)
	// of audio data - used to sync audio to video after seeking
	// if you don't return CONTROL_TRUE, it will defaults to:
	//      ds_fill_buffer(sh_audio->ds);  // skip 1 demux packet
	...
	return CONTROL_TRUE;
    }
  return CONTROL_UNKNOWN;
}