mirror of
https://github.com/mpv-player/mpv
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b0986b3760
Note that r30455 is wrong, that commit does not in fact change the default behavior as claimed in the commit message. It only breaks "-af-adv force=0", which was already pretty much useless though.
146 lines
5.7 KiB
C
146 lines
5.7 KiB
C
// SAMPLE audio decoder - you can use this file as template when creating new codec!
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/*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include "config.h"
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#include "ad_internal.h"
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static const ad_info_t info = {
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"Sample audio decoder", // name of the driver
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"sample", // driver name. should be the same as filename without ad_
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"A'rpi", // writer/maintainer of _this_ file
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"", // writer/maintainer/site of the _codec_
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"" // comments
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};
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LIBAD_EXTERN(sample)
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#include "libsample/sample.h" // include your codec's .h files here
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static int preinit(sh_audio_t *sh){
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// let's check if the driver is available, return 0 if not.
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// (you should do that if you use external lib(s) which is optional)
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...
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// there are default values set for buffering, but you can override them:
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// minimum output buffer size (should be the uncompressed max. frame size)
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sh->audio_out_minsize=4*2*1024; // in this sample, we assume max 4 channels,
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// 2 bytes/sample and 1024 samples/frame
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// Default: 8192
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// minimum input buffer size (set only if you need input buffering)
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// (should be the max compressed frame size)
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sh->audio_in_minsize=2048; // Default: 0 (no input buffer)
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// if you set audio_in_minsize non-zero, the buffer will be allocated
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// before the init() call by the core, and you can access it via
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// pointer: sh->audio_in_buffer
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// it will free'd after uninit(), so you don't have to use malloc/free here!
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// the next few parameters define the audio format (channels, sample type,
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// in/out bitrate etc.). it's OK to move these to init() if you can set
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// them only after some initialization:
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sh->samplesize=2; // bytes (not bits!) per sample per channel
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sh->channels=2; // number of channels
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sh->samplerate=44100; // samplerate
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sh->sample_format=AF_FORMAT_S16_LE; // sample format, see libao2/afmt.h
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sh->i_bps=64000/8; // input data rate (compressed bytes per second)
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// Note: if you have VBR or unknown input rate, set it to some common or
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// average value, instead of zero. it's used to predict time delay of
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// buffered compressed bytes, so it must be more-or-less real!
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//sh->o_bps=... // output data rate (uncompressed bytes per second)
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// Note: you DON'T need to set o_bps in most cases, as it defaults to:
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// sh->samplesize*sh->channels*sh->samplerate;
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// for constant rate compressed QuickTime (.mov files) codecs you MUST
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// set the compressed and uncompressed packet size (used by the demuxer):
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sh->ds->ss_mul = 34; // compressed packet size
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sh->ds->ss_div = 64; // samples per packet
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return 1; // return values: 1=OK 0=ERROR
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}
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static int init(sh_audio_t *sh_audio){
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// initialize the decoder, set tables etc...
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// you can store HANDLE or private struct pointer at sh->context
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// you can access WAVEFORMATEX header at sh->wf
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// set sample format/rate parameters if you didn't do it in preinit() yet.
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return 1; // return values: 1=OK 0=ERROR
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}
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static void uninit(sh_audio_t *sh){
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// uninit the decoder etc...
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// again: you don't have to free() a_in_buffer here! it's done by the core.
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}
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static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen){
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// audio decoding. the most important thing :)
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// parameters you get:
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// buf = pointer to the output buffer, you have to store uncompressed
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// samples there
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// minlen = requested minimum size (in bytes!) of output. it's just a
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// _recommendation_, you can decode more or less, it just tell you that
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// the caller process needs 'minlen' bytes. if it gets less, it will
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// call decode_audio() again.
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// maxlen = maximum size (bytes) of output. you MUST NOT write more to the
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// buffer, it's the upper-most limit!
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// note: maxlen will be always greater or equal to sh->audio_out_minsize
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// now, let's decode...
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// you can read the compressed stream using the demux stream functions:
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// demux_read_data(sh->ds, buffer, length) - read 'length' bytes to 'buffer'
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// ds_get_packet(sh->ds, &buffer) - set ptr buffer to next data packet
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// (both func return number of bytes or 0 for error)
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return len; // return value: number of _bytes_ written to output buffer,
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// or -1 for EOF (or uncorrectable error)
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}
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static int control(sh_audio_t *sh,int cmd,void* arg, ...){
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// various optional functions you MAY implement:
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switch(cmd){
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case ADCTRL_RESYNC_STREAM:
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// it is called once after seeking, to resync.
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// Note: sh_audio->a_in_buffer_len=0; is done _before_ this call!
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...
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return CONTROL_TRUE;
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case ADCTRL_SKIP_FRAME:
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// it is called to skip (jump over) small amount (1/10 sec or 1 frame)
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// of audio data - used to sync audio to video after seeking
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// if you don't return CONTROL_TRUE, it will defaults to:
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// ds_fill_buffer(sh_audio->ds); // skip 1 demux packet
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...
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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}
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