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author | Aki <please@ignore.pl> | 2021-10-01 23:46:47 +0200 |
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committer | Aki <please@ignore.pl> | 2021-10-01 23:46:47 +0200 |
commit | ddd76aa4e1571c8d5965709db5c3cd47a89c9a38 (patch) | |
tree | 4e07bfaebede6531a70984fdebc855bca52f394a /vorbis/examples/encoder_example.c | |
parent | 3a507e08b1d4e5970b27401a7e6517570d529400 (diff) | |
parent | c2d2c3551501110fddd78674d5435bfaa70382a3 (diff) | |
download | starshatter-ddd76aa4e1571c8d5965709db5c3cd47a89c9a38.zip starshatter-ddd76aa4e1571c8d5965709db5c3cd47a89c9a38.tar.gz starshatter-ddd76aa4e1571c8d5965709db5c3cd47a89c9a38.tar.bz2 |
Project is now built with CMake
Diffstat (limited to 'vorbis/examples/encoder_example.c')
-rw-r--r-- | vorbis/examples/encoder_example.c | 251 |
1 files changed, 251 insertions, 0 deletions
diff --git a/vorbis/examples/encoder_example.c b/vorbis/examples/encoder_example.c new file mode 100644 index 0000000..d46a051 --- /dev/null +++ b/vorbis/examples/encoder_example.c @@ -0,0 +1,251 @@ +/******************************************************************** + * * + * 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 + + ********************************************************************/ + +/* 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); +} |