Add support for VDPAU video out, including hardware decoding.
[mplayer/glamo.git] / libmpcodecs / ad_sample.c
blob848152c68e55c1245dd23d2ef14499c90b1017e9
1 // SAMPLE audio decoder - you can use this file as template when creating new codec!
3 #include <stdio.h>
4 #include <stdlib.h>
5 #include <unistd.h>
7 #include "config.h"
8 #include "ad_internal.h"
10 static ad_info_t info = {
11 "Sample audio decoder", // name of the driver
12 "sample", // driver name. should be the same as filename without ad_
13 "A'rpi", // writer/maintainer of _this_ file
14 "", // writer/maintainer/site of the _codec_
15 "" // comments
18 LIBAD_EXTERN(sample)
20 #include "libsample/sample.h" // include your codec's .h files here
22 static int preinit(sh_audio_t *sh){
23 // let's check if the driver is available, return 0 if not.
24 // (you should do that if you use external lib(s) which is optional)
25 ...
27 // there are default values set for buffering, but you can override them:
29 // minimum output buffer size (should be the uncompressed max. frame size)
30 sh->audio_out_minsize=4*2*1024; // in this sample, we assume max 4 channels,
31 // 2 bytes/sample and 1024 samples/frame
32 // Default: 8192
34 // minimum input buffer size (set only if you need input buffering)
35 // (should be the max compressed frame size)
36 sh->audio_in_minsize=2048; // Default: 0 (no input buffer)
38 // if you set audio_in_minsize non-zero, the buffer will be allocated
39 // before the init() call by the core, and you can access it via
40 // pointer: sh->audio_in_buffer
41 // it will free'd after uninit(), so you don't have to use malloc/free here!
43 // the next few parameters define the audio format (channels, sample type,
44 // in/out bitrate etc.). it's OK to move these to init() if you can set
45 // them only after some initialization:
47 sh->samplesize=2; // bytes (not bits!) per sample per channel
48 sh->channels=2; // number of channels
49 sh->samplerate=44100; // samplerate
50 sh->sample_format=AF_FORMAT_S16_LE; // sample format, see libao2/afmt.h
52 sh->i_bps=64000/8; // input data rate (compressed bytes per second)
53 // Note: if you have VBR or unknown input rate, set it to some common or
54 // average value, instead of zero. it's used to predict time delay of
55 // buffered compressed bytes, so it must be more-or-less real!
57 //sh->o_bps=... // output data rate (uncompressed bytes per second)
58 // Note: you DON'T need to set o_bps in most cases, as it defaults to:
59 // sh->samplesize*sh->channels*sh->samplerate;
61 // for constant rate compressed QuickTime (.mov files) codecs you MUST
62 // set the compressed and uncompressed packet size (used by the demuxer):
63 sh->ds->ss_mul = 34; // compressed packet size
64 sh->ds->ss_div = 64; // samples per packet
66 return 1; // return values: 1=OK 0=ERROR
69 static int init(sh_audio_t *sh_audio){
70 // initialize the decoder, set tables etc...
72 // you can store HANDLE or private struct pointer at sh->context
73 // you can access WAVEFORMATEX header at sh->wf
75 // set sample format/rate parameters if you didn't do it in preinit() yet.
77 return 1; // return values: 1=OK 0=ERROR
80 static void uninit(sh_audio_t *sh){
81 // uninit the decoder etc...
82 // again: you don't have to free() a_in_buffer here! it's done by the core.
85 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen){
87 // audio decoding. the most important thing :)
88 // parameters you get:
89 // buf = pointer to the output buffer, you have to store uncompressed
90 // samples there
91 // minlen = requested minimum size (in bytes!) of output. it's just a
92 // _recommendation_, you can decode more or less, it just tell you that
93 // the caller process needs 'minlen' bytes. if it gets less, it will
94 // call decode_audio() again.
95 // maxlen = maximum size (bytes) of output. you MUST NOT write more to the
96 // buffer, it's the upper-most limit!
97 // note: maxlen will be always greater or equal to sh->audio_out_minsize
99 // now, let's decode...
101 // you can read the compressed stream using the demux stream functions:
102 // demux_read_data(sh->ds, buffer, length) - read 'length' bytes to 'buffer'
103 // ds_get_packet(sh->ds, &buffer) - set ptr buffer to next data packet
104 // (both func return number of bytes or 0 for error)
106 return len; // return value: number of _bytes_ written to output buffer,
107 // or -1 for EOF (or uncorrectable error)
110 static int control(sh_audio_t *sh,int cmd,void* arg, ...){
111 // various optional functions you MAY implement:
112 switch(cmd){
113 case ADCTRL_RESYNC_STREAM:
114 // it is called once after seeking, to resync.
115 // Note: sh_audio->a_in_buffer_len=0; is done _before_ this call!
117 return CONTROL_TRUE;
118 case ADCTRL_SKIP_FRAME:
119 // it is called to skip (jump over) small amount (1/10 sec or 1 frame)
120 // of audio data - used to sync audio to video after seeking
121 // if you don't return CONTROL_TRUE, it will defaults to:
122 // ds_fill_buffer(sh_audio->ds); // skip 1 demux packet
124 return CONTROL_TRUE;
126 return CONTROL_UNKNOWN;