sample

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@5463 b3059339-0415-0410-9bf9-f77b7e298cf2

1 files changed, 129 insertions, 0 deletions

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 <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_
+	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;
+}