From b168261f1030cb00be5fa90d0f9bfd9cfa72fe0f Mon Sep 17 00:00:00 2001 From: arpi Date: Mon, 1 Apr 2002 19:14:14 +0000 Subject: sample git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@5463 b3059339-0415-0410-9bf9-f77b7e298cf2 --- libmpcodecs/ad_sample.c | 129 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 129 insertions(+) create mode 100644 libmpcodecs/ad_sample.c diff --git a/libmpcodecs/ad_sample.c b/libmpcodecs/ad_sample.c new file mode 100644 index 0000000000..f20fbcbb92 --- /dev/null +++ b/libmpcodecs/ad_sample.c @@ -0,0 +1,129 @@ +// SAMPLE audio decoder - you can use this file as template when creating new codec! + +#include +#include +#include + +#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_ + AFM_SAMPLE, // replace with registered AFM number + "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. + // if you don't return CONTROL_TRUE, it will defaults to: + // sh_audio->a_in_buffer_len=0; // clear input buffer + ... + 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; +} -- cgit v1.2.3