diff src/libvorbis-1.3.3/examples/encoder_example.c @ 86:98c1576536ae

Bring in flac, ogg, vorbis
author Chris Cannam <cannam@all-day-breakfast.com>
date Tue, 19 Mar 2013 17:37:49 +0000
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--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/src/libvorbis-1.3.3/examples/encoder_example.c	Tue Mar 19 17:37:49 2013 +0000
@@ -0,0 +1,252 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: simple example encoder
+ last mod: $Id: encoder_example.c 16946 2010-03-03 16:12:40Z xiphmont $
+
+ ********************************************************************/
+
+/* takes a stereo 16bit 44.1kHz WAV file from stdin and encodes it into
+   a Vorbis bitstream */
+
+/* Note that this is POSIX, not ANSI, code */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <time.h>
+#include <math.h>
+#include <vorbis/vorbisenc.h>
+
+#ifdef _WIN32 /* We need the following two to set stdin/stdout to binary */
+#include <io.h>
+#include <fcntl.h>
+#endif
+
+#if defined(__MACOS__) && defined(__MWERKS__)
+#include <console.h>      /* CodeWarrior's Mac "command-line" support */
+#endif
+
+#define READ 1024
+signed char readbuffer[READ*4+44]; /* out of the data segment, not the stack */
+
+int main(){
+  ogg_stream_state os; /* take physical pages, weld into a logical
+                          stream of packets */
+  ogg_page         og; /* one Ogg bitstream page.  Vorbis packets are inside */
+  ogg_packet       op; /* one raw packet of data for decode */
+
+  vorbis_info      vi; /* struct that stores all the static vorbis bitstream
+                          settings */
+  vorbis_comment   vc; /* struct that stores all the user comments */
+
+  vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
+  vorbis_block     vb; /* local working space for packet->PCM decode */
+
+  int eos=0,ret;
+  int i, founddata;
+
+#if defined(macintosh) && defined(__MWERKS__)
+  int argc = 0;
+  char **argv = NULL;
+  argc = ccommand(&argv); /* get a "command line" from the Mac user */
+                          /* this also lets the user set stdin and stdout */
+#endif
+
+  /* we cheat on the WAV header; we just bypass 44 bytes (simplest WAV
+     header is 44 bytes) and assume that the data is 44.1khz, stereo, 16 bit
+     little endian pcm samples. This is just an example, after all. */
+
+#ifdef _WIN32 /* We need to set stdin/stdout to binary mode. Damn windows. */
+  /* if we were reading/writing a file, it would also need to in
+     binary mode, eg, fopen("file.wav","wb"); */
+  /* Beware the evil ifdef. We avoid these where we can, but this one we
+     cannot. Don't add any more, you'll probably go to hell if you do. */
+  _setmode( _fileno( stdin ), _O_BINARY );
+  _setmode( _fileno( stdout ), _O_BINARY );
+#endif
+
+
+  /* we cheat on the WAV header; we just bypass the header and never
+     verify that it matches 16bit/stereo/44.1kHz.  This is just an
+     example, after all. */
+
+  readbuffer[0] = '\0';
+  for (i=0, founddata=0; i<30 && ! feof(stdin) && ! ferror(stdin); i++)
+  {
+    fread(readbuffer,1,2,stdin);
+
+    if ( ! strncmp((char*)readbuffer, "da", 2) ){
+      founddata = 1;
+      fread(readbuffer,1,6,stdin);
+      break;
+    }
+  }
+
+  /********** Encode setup ************/
+
+  vorbis_info_init(&vi);
+
+  /* choose an encoding mode.  A few possibilities commented out, one
+     actually used: */
+
+  /*********************************************************************
+   Encoding using a VBR quality mode.  The usable range is -.1
+   (lowest quality, smallest file) to 1. (highest quality, largest file).
+   Example quality mode .4: 44kHz stereo coupled, roughly 128kbps VBR
+
+   ret = vorbis_encode_init_vbr(&vi,2,44100,.4);
+
+   ---------------------------------------------------------------------
+
+   Encoding using an average bitrate mode (ABR).
+   example: 44kHz stereo coupled, average 128kbps VBR
+
+   ret = vorbis_encode_init(&vi,2,44100,-1,128000,-1);
+
+   ---------------------------------------------------------------------
+
+   Encode using a quality mode, but select that quality mode by asking for
+   an approximate bitrate.  This is not ABR, it is true VBR, but selected
+   using the bitrate interface, and then turning bitrate management off:
+
+   ret = ( vorbis_encode_setup_managed(&vi,2,44100,-1,128000,-1) ||
+           vorbis_encode_ctl(&vi,OV_ECTL_RATEMANAGE2_SET,NULL) ||
+           vorbis_encode_setup_init(&vi));
+
+   *********************************************************************/
+
+  ret=vorbis_encode_init_vbr(&vi,2,44100,0.1);
+
+  /* do not continue if setup failed; this can happen if we ask for a
+     mode that libVorbis does not support (eg, too low a bitrate, etc,
+     will return 'OV_EIMPL') */
+
+  if(ret)exit(1);
+
+  /* add a comment */
+  vorbis_comment_init(&vc);
+  vorbis_comment_add_tag(&vc,"ENCODER","encoder_example.c");
+
+  /* set up the analysis state and auxiliary encoding storage */
+  vorbis_analysis_init(&vd,&vi);
+  vorbis_block_init(&vd,&vb);
+
+  /* set up our packet->stream encoder */
+  /* pick a random serial number; that way we can more likely build
+     chained streams just by concatenation */
+  srand(time(NULL));
+  ogg_stream_init(&os,rand());
+
+  /* Vorbis streams begin with three headers; the initial header (with
+     most of the codec setup parameters) which is mandated by the Ogg
+     bitstream spec.  The second header holds any comment fields.  The
+     third header holds the bitstream codebook.  We merely need to
+     make the headers, then pass them to libvorbis one at a time;
+     libvorbis handles the additional Ogg bitstream constraints */
+
+  {
+    ogg_packet header;
+    ogg_packet header_comm;
+    ogg_packet header_code;
+
+    vorbis_analysis_headerout(&vd,&vc,&header,&header_comm,&header_code);
+    ogg_stream_packetin(&os,&header); /* automatically placed in its own
+                                         page */
+    ogg_stream_packetin(&os,&header_comm);
+    ogg_stream_packetin(&os,&header_code);
+
+    /* This ensures the actual
+     * audio data will start on a new page, as per spec
+     */
+    while(!eos){
+      int result=ogg_stream_flush(&os,&og);
+      if(result==0)break;
+      fwrite(og.header,1,og.header_len,stdout);
+      fwrite(og.body,1,og.body_len,stdout);
+    }
+
+  }
+
+  while(!eos){
+    long i;
+    long bytes=fread(readbuffer,1,READ*4,stdin); /* stereo hardwired here */
+
+    if(bytes==0){
+      /* end of file.  this can be done implicitly in the mainline,
+         but it's easier to see here in non-clever fashion.
+         Tell the library we're at end of stream so that it can handle
+         the last frame and mark end of stream in the output properly */
+      vorbis_analysis_wrote(&vd,0);
+
+    }else{
+      /* data to encode */
+
+      /* expose the buffer to submit data */
+      float **buffer=vorbis_analysis_buffer(&vd,READ);
+
+      /* uninterleave samples */
+      for(i=0;i<bytes/4;i++){
+        buffer[0][i]=((readbuffer[i*4+1]<<8)|
+                      (0x00ff&(int)readbuffer[i*4]))/32768.f;
+        buffer[1][i]=((readbuffer[i*4+3]<<8)|
+                      (0x00ff&(int)readbuffer[i*4+2]))/32768.f;
+      }
+
+      /* tell the library how much we actually submitted */
+      vorbis_analysis_wrote(&vd,i);
+    }
+
+    /* vorbis does some data preanalysis, then divvies up blocks for
+       more involved (potentially parallel) processing.  Get a single
+       block for encoding now */
+    while(vorbis_analysis_blockout(&vd,&vb)==1){
+
+      /* analysis, assume we want to use bitrate management */
+      vorbis_analysis(&vb,NULL);
+      vorbis_bitrate_addblock(&vb);
+
+      while(vorbis_bitrate_flushpacket(&vd,&op)){
+
+        /* weld the packet into the bitstream */
+        ogg_stream_packetin(&os,&op);
+
+        /* write out pages (if any) */
+        while(!eos){
+          int result=ogg_stream_pageout(&os,&og);
+          if(result==0)break;
+          fwrite(og.header,1,og.header_len,stdout);
+          fwrite(og.body,1,og.body_len,stdout);
+
+          /* this could be set above, but for illustrative purposes, I do
+             it here (to show that vorbis does know where the stream ends) */
+
+          if(ogg_page_eos(&og))eos=1;
+        }
+      }
+    }
+  }
+
+  /* clean up and exit.  vorbis_info_clear() must be called last */
+
+  ogg_stream_clear(&os);
+  vorbis_block_clear(&vb);
+  vorbis_dsp_clear(&vd);
+  vorbis_comment_clear(&vc);
+  vorbis_info_clear(&vi);
+
+  /* ogg_page and ogg_packet structs always point to storage in
+     libvorbis.  They're never freed or manipulated directly */
+
+  fprintf(stderr,"Done.\n");
+  return(0);
+}