From f8c39a08f6ff651221ff896f8a174abf635f3a0a Mon Sep 17 00:00:00 2001 From: markster Date: Tue, 4 Nov 2003 02:40:09 +0000 Subject: 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 --- codecs/codec_mp3_d.c | 320 --------------------------------------------------- 1 file changed, 320 deletions(-) delete mode 100755 codecs/codec_mp3_d.c (limited to 'codecs/codec_mp3_d.c') 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 - * - * This program is free software, distributed under the terms of - * the GNU General Public License - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#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; -} -- cgit v1.2.3