Remove really broke MP3 stuff in favor of G.726 in the near future

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1689 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 0 insertions, 320 deletions

diff --git a/codecs/codec_mp3_d.c b/codecs/codec_mp3_d.c
deleted file mode 100755
index 9ff3961aa..000000000
--- a/codecs/codec_mp3_d.c
+++ /dev/null
@@ -1,320 +0,0 @@
-/*
- * Asterisk -- A telephony toolkit for Linux.
- *
- * MP3 Decoder
- *
- * The MP3 code is from freeamp, which in turn is from xingmp3's release
- * which I can't seem to find anywhere
- * 
- * Copyright (C) 1999, Mark Spencer
- *
- * Mark Spencer <markster@linux-support.net>
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License
- */
-
-#include <asterisk/lock.h>
-#include <asterisk/translate.h>
-#include <asterisk/module.h>
-#include <asterisk/logger.h>
-#include <asterisk/channel.h>
-#include <pthread.h>
-#include <fcntl.h>
-#include <errno.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <netinet/in.h>
-#include <string.h>
-#include <stdio.h>
-
-#include "mp3/include/L3.h"
-#include "mp3/include/mhead.h"
-
-#include "mp3anal.h"
-
-/* Sample frame data */
-#include "mp3_slin_ex.h"
-
-#define MAX_OUT_FRAME 320
-
-#define MAX_FRAME_SIZE 1441
-#define MAX_OUTPUT_LEN 2304
-
-static ast_mutex_t localuser_lock = AST_MUTEX_INITIALIZER;
-static int localusecnt=0;
-
-static char *tdesc = "MP3/PCM16 (signed linear) Translator (Decoder only)";
-
-struct ast_translator_pvt {
-	MPEG m;
-	MPEG_HEAD head;
-	DEC_INFO info;
-	struct ast_frame f;
-	/* Space to build offset */
-	char offset[AST_FRIENDLY_OFFSET];
-	/* Mini buffer */
-	char outbuf[MAX_OUT_FRAME];
-	/* Enough to store a full second */
-	short buf[32000];
-	/* Tail of signed linear stuff */
-	int tail;
-	/* Current bitrate */
-	int bitrate;
-	/* XXX What's forward? XXX */
-	int forward;
-	/* Have we called head info yet? */
-	int init;
-	int copy;
-};
-
-#define mp3_coder_pvt ast_translator_pvt
-
-static struct ast_translator_pvt *mp3_new(void)
-{
-	struct mp3_coder_pvt *tmp;
-	tmp = malloc(sizeof(struct mp3_coder_pvt));
-	if (tmp) {
-		tmp->init = 0;
-		tmp->tail = 0;
-		tmp->copy = -1;
-		mpeg_init(&tmp->m);
-	}
-	return tmp;
-}
-
-static struct ast_frame *mp3tolin_sample(void)
-{
-	static struct ast_frame f;
-	int size;
-	if (mp3_badheader(mp3_slin_ex)) {
-		ast_log(LOG_WARNING, "Bad MP3 sample??\n");
-		return NULL;
-	}
-	size = mp3_framelen(mp3_slin_ex);
-	if (size < 1) {
-		ast_log(LOG_WARNING, "Failed to size??\n");
-		return NULL;
-	}
-	f.frametype = AST_FRAME_VOICE;
-	f.subclass = AST_FORMAT_MP3;
-	f.data = mp3_slin_ex;
-	f.datalen = sizeof(mp3_slin_ex);
-	/* Dunno how long an mp3 frame is -- kinda irrelevant anyway */
-	f.samples = 240;
-	f.mallocd = 0;
-	f.offset = 0;
-	f.src = __PRETTY_FUNCTION__;
-	return &f;
-}
-
-static struct ast_frame *mp3tolin_frameout(struct ast_translator_pvt *tmp)
-{
-	if (!tmp->tail)
-		return NULL;
-	/* Signed linear is no particular frame size, so just send whatever
-	   we have in the buffer in one lump sum */
-	tmp->f.frametype = AST_FRAME_VOICE;
-	tmp->f.subclass = AST_FORMAT_SLINEAR;
-	tmp->f.datalen = tmp->tail * 2;
-	/* Assume 8000 Hz */
-	tmp->f.samples = tmp->tail;
-	tmp->f.mallocd = 0;
-	tmp->f.offset = AST_FRIENDLY_OFFSET;
-	tmp->f.src = __PRETTY_FUNCTION__;
-	tmp->f.data = tmp->buf;
-	/* Reset tail pointer */
-	tmp->tail = 0;
-
-#if 0
-	/* Save a sample frame */
-	{
-		static int fd = -1;
-		if (fd < 0) 
-			fd = open("mp3out.raw", O_WRONLY | O_CREAT | O_TRUNC, 0644);
-		write(fd, tmp->f.data, tmp->f.datalen);
-	} 		
-#endif
-	return &tmp->f;	
-}
-
-static int mp3_init(struct ast_translator_pvt *tmp, int len)
-{	
-	if (!audio_decode_init(&tmp->m, &tmp->head, len,0,0,1 /* Convert to mono */,24000)) {
-		ast_log(LOG_WARNING, "audio_decode_init() failed\n");
-		return -1;
-	}
-	audio_decode_info(&tmp->m, &tmp->info);
-#if 0
-	ast_verbose(
-"Channels: %d\nOutValues: %d\nSample Rate: %d\nBits: %d\nFramebytes: %d\nType: %d\n",
-	tmp->info.channels, tmp->info.outvalues, tmp->info.samprate, tmp->info.bits,tmp->info.framebytes,tmp->info.type);
-#endif
-	return 0;
-}
-
-#ifndef MIN
-#define MIN(a,b) (((a) < (b)) ? (a) : (b))
-#endif
-
-#if 1
-static int add_to_buf(short *dst, int maxdst, short *src, int srclen, int samprate)
-{
-	float inc, cur, sum=0;
-	int cnt=0, pos, ptr, lastpos = -1;
-	/* Resample source to destination converting from its sampling rate to 8000 Hz */
-	if (samprate == 8000) {
-		/* Quickly, all we have to do is copy */
-		memcpy(dst, src, 2 * MIN(maxdst, srclen));
-		return MIN(maxdst, srclen);
-	}
-	if (samprate < 8000) {
-		ast_log(LOG_WARNING, "Don't know how to resample a source less than 8000 Hz!\n");
-		/* XXX Wrong thing to do XXX */
-		memcpy(dst, src, 2 * MIN(maxdst, srclen));
-		return MIN(maxdst, srclen);
-	}
-	/* Ugh, we actually *have* to resample */
-	inc = 8000.0 / (float)samprate;
-	cur = 0;
-	ptr = 0;
-	pos = 0;
-#if 0
-	ast_verbose("Incrementing by %f, in = %d bytes, out = %d bytes\n", inc, srclen, maxdst);
-#endif
-	while((pos < maxdst) && (ptr < srclen)) {
-		if (pos != lastpos) {
-			if (lastpos > -1) {
-				sum = sum / (float)cnt;
-				dst[pos - 1] = (int) sum;
-#if 0
-				ast_verbose("dst[%d] = %d\n", pos - 1, dst[pos - 1]);
-#endif
-			}
-			/* Each time we have a first pass */
-			sum = 0;
-			cnt = 0;
-		} else {
-			sum += src[ptr];
-		}
-		ptr++;
-		cur += inc;
-		cnt++;
-		lastpos = pos;
-		pos = (int)cur;
-	}
-	return pos;
-}
-#endif
-
-static int mp3tolin_framein(struct ast_translator_pvt *tmp, struct ast_frame *f)
-{
-	/* Assuming there's space left, decode into the current buffer at
-	   the tail location */
-	int framelen;
-	short tmpbuf[8000];
-	IN_OUT x;
-#if 0
-	if (tmp->copy < 0) {
-		tmp->copy = open("sample.out", O_WRONLY | O_CREAT | O_TRUNC, 0700);
-	}
-	if (tmp->copy > -1)
-		write(tmp->copy, f->data, f->datalen);
-#endif
-	/* Check if it's a valid frame */
-	if (mp3_badheader((unsigned char *)f->data)) {
-		ast_log(LOG_WARNING, "Invalid MP3 header\n");
-		return -1;
-	}
-	if ((framelen = mp3_framelen((unsigned char *)f->data) != f->datalen)) {
-		ast_log(LOG_WARNING, "Calculated length %d does not match real length %d\n", framelen, f->datalen);
-		return -1;
-	}
-	/* Start by putting this in the mp3 buffer */
-	if((framelen = head_info3(f->data, 
-			f->datalen, &tmp->head, &tmp->bitrate, &tmp->forward)) > 0) {
-		if (!tmp->init) {
-			if (mp3_init(tmp, framelen))
-				return -1;
-			else
-				tmp->init++;
-		}
-		if (tmp->tail + MAX_OUTPUT_LEN/2  < sizeof(tmp->buf)/2) {	
-			x = audio_decode(&tmp->m, f->data, tmpbuf);
-			audio_decode_info(&tmp->m, &tmp->info);
-			if (!x.in_bytes) {
-				ast_log(LOG_WARNING, "Invalid MP3 data\n");
-			} else {
-#if 1
-				/* Resample to 8000 Hz */
-				tmp->tail += add_to_buf(tmp->buf + tmp->tail, 
-			           sizeof(tmp->buf) / 2 - tmp->tail, 
-					   tmpbuf,
-					   x.out_bytes/2,
-					   tmp->info.samprate);
-#else
-				memcpy(tmp->buf + tmp->tail, tmpbuf, x.out_bytes);
-				/* Signed linear output */
-				tmp->tail+=x.out_bytes/2;
-#endif
-			}
-		} else {
-			ast_log(LOG_WARNING, "Out of buffer space\n");
-			return -1;
-		}
-	} else {
-		ast_log(LOG_WARNING, "Not a valid MP3 frame\n");
-	}
-	return 0;
-}
-
-static void mp3_destroy_stuff(struct ast_translator_pvt *pvt)
-{
-	close(pvt->copy);
-	free(pvt);
-}
-
-static struct ast_translator mp3tolin =
-	{ "mp3tolin", 
-	   AST_FORMAT_MP3, AST_FORMAT_SLINEAR,
-	   mp3_new,
-	   mp3tolin_framein,
-	   mp3tolin_frameout,
-	   mp3_destroy_stuff,
-	   mp3tolin_sample
-	   };
-
-int unload_module(void)
-{
-	int res;
-	ast_mutex_lock(&localuser_lock);
-	res = ast_unregister_translator(&mp3tolin);
-	if (localusecnt)
-		res = -1;
-	ast_mutex_unlock(&localuser_lock);
-	return res;
-}
-
-int load_module(void)
-{
-	int res;
-	res=ast_register_translator(&mp3tolin);
-	return res;
-}
-
-char *description(void)
-{
-	return tdesc;
-}
-
-int usecount(void)
-{
-	int res;
-	STANDARD_USECOUNT(res);
-	return res;
-}
-
-char *key()
-{
-	return ASTERISK_GPL_KEY;
-}