Added raw sound to ogg/vorbis converter

3 files changed, 178 insertions, 0 deletions

diff --git a/README.md b/README.md
index 5eaaf57..6e937d5 100644
--- a/README.md
+++ b/README.md
@@ -8,3 +8,4 @@ hex_editor_1k - cli hex editor
 http_show - CGI that outputs what have been sended to it
 mini_irc - can connect and reply on PM,chat messages and join chan
 alsa-record-sound - record sound with alsa, 48kHz,2ch
+raw2vorbus - convert raw audio file 48kHz,2ch recorded with alsa-record-sound to ogg format and then you can play it witn mplayer 
diff --git a/raw2ogg/Makefile b/raw2ogg/Makefile
new file mode 100644
index 0000000..bb2d678
--- /dev/null
+++ b/raw2ogg/Makefile
@@ -0,0 +1,2 @@
+make:
+	gcc raw2vorbis.c -o raw2vorbis -logg -lvorbis -lvorbisenc
diff --git a/raw2ogg/raw2vorbis.c b/raw2ogg/raw2vorbis.c
new file mode 100644
index 0000000..5643695
--- /dev/null
+++ b/raw2ogg/raw2vorbis.c
@@ -0,0 +1,175 @@
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <time.h>
+#include <math.h>
+
+#include <vorbis/vorbisenc.h>
+
+#define READ_SZ 1024
+signed char readbuffer[READ_SZ*4+44];
+
+int main(int argc, char **argv)
+{
+	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;
+	FILE *fin=NULL,*fout=NULL;
+
+	/* get input/output files from input */
+	if (argc != 3)
+	{
+		printf("%s [input] [output]\n",argv[0]);
+		exit(1);
+	}
+
+	fin = fopen(argv[1],"r");
+	if (fin == NULL)
+	{
+		printf("Cant open file %s\n", argv[1]);
+		exit(1);
+	}
+
+	fout = fopen(argv[2],"w");
+	if (fout == NULL)
+	{
+		printf("Cant open file %s\n", argv[2]);
+		exit(1);
+	}
+
+	/* encoding setup */
+	vorbis_info_init(&vi);
+
+	ret=vorbis_encode_init_vbr(&vi,2,44100,-.1);
+	if (ret)
+	{
+		printf("Cant setup encode mode\n");
+		exit(1);
+	}
+
+	/* add 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, fout);
+			fwrite(og.body,   1, og.body_len,   fout);
+		}
+	}
+
+	while (!eos)
+	{
+		long i;
+		long bytes = fread(readbuffer, 1, READ_SZ*4, fin);
+
+		if (bytes == 0)
+		{
+			vorbis_analysis_wrote(&vd, 0);
+		} else
+		{
+			float **buffer = vorbis_analysis_buffer(&vd, READ_SZ);
+			/*do i need interleaver 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;
+			}
+			vorbis_analysis_wrote(&vd, bytes/4);
+		}
+
+		/* 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, fout);
+					fwrite(og.body, 1, og.body_len, fout);
+					
+					/* 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;
+				}
+			}
+
+		}
+	}
+	
+	ogg_stream_clear(&os);
+	vorbis_block_clear(&vb);
+	vorbis_dsp_clear(&vd);
+	vorbis_comment_clear(&vc);
+	vorbis_info_clear(&vi);
+
+	fclose(fout);
+	fclose(fin);
+
+	printf("Done\n");
+
+	return 0;
+}
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