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authorwm4 <wm4@nowhere>2012-08-18 11:17:35 +0200
committerwm4 <wm4@nowhere>2012-08-20 15:36:04 +0200
commit6a26b4a66504f701baf35e58467e55aea28c0ad5 (patch)
tree8b09b91d63926543eaa8ec95c90a2532bde71dd6 /libmpcodecs/ad_sample.c
parent6f7ba66817b5cd3761b802930dc7ba62464e3c6a (diff)
downloadmpv-6a26b4a66504f701baf35e58467e55aea28c0ad5.tar.bz2
mpv-6a26b4a66504f701baf35e58467e55aea28c0ad5.tar.xz
libmpcodecs: remove redundant audio and video decoders
Probably all of these are supported by libavcodec. Missing things can be added back. Also remove qtpalette.h. It was used by demux_mov.c, and should have been deleted with commit 1fde09db6f4ce.
Diffstat (limited to 'libmpcodecs/ad_sample.c')
-rw-r--r--libmpcodecs/ad_sample.c145
1 files changed, 0 insertions, 145 deletions
diff --git a/libmpcodecs/ad_sample.c b/libmpcodecs/ad_sample.c
deleted file mode 100644
index 69f4b20dfc..0000000000
--- a/libmpcodecs/ad_sample.c
+++ /dev/null
@@ -1,145 +0,0 @@
-// SAMPLE audio decoder - you can use this file as template when creating new codec!
-
-/*
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <unistd.h>
-
-#include "config.h"
-#include "ad_internal.h"
-
-static const 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;
-}