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// 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=AF_FORMAT_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;
}
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