Mercurial > mplayer.hg
view libmpcodecs/ad_sample.c @ 17921:baa1df31e661
insert linebreak, \-playing-msg option
author | kraymer |
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date | Wed, 22 Mar 2006 18:19:15 +0000 |
parents | 815f03b7cee5 |
children | 0f1b5b68af32 |
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line source
// 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; }