diff options
Diffstat (limited to 'codecs/codec_mp3_d.c')
| -rwxr-xr-x | codecs/codec_mp3_d.c | 323 |
1 files changed, 323 insertions, 0 deletions
diff --git a/codecs/codec_mp3_d.c b/codecs/codec_mp3_d.c new file mode 100755 index 000000000..95e1fb51f --- /dev/null +++ b/codecs/codec_mp3_d.c @@ -0,0 +1,323 @@ +/* + * 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/translate.h> +#include <asterisk/module.h> +#include <asterisk/logger.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 pthread_mutex_t localuser_lock = PTHREAD_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() +{ + 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() +{ + 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.timelen = 30; + f.mallocd = 0; + f.offset = 0; + f.src = __PRETTY_FUNCTION__; + return &f; +} + +static struct ast_frame *mp3tolin_frameout(struct ast_translator_pvt *tmp) +{ + int sent; + if (!tmp->tail) + return NULL; + sent = tmp->tail; + if (sent > MAX_OUT_FRAME/2) + sent = MAX_OUT_FRAME/2; + /* 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 = sent * 2; + /* Assume 8000 Hz */ + tmp->f.timelen = sent / 8; + tmp->f.mallocd = 0; + tmp->f.offset = AST_FRIENDLY_OFFSET; + tmp->f.src = __PRETTY_FUNCTION__; + memcpy(tmp->outbuf, tmp->buf, tmp->tail * 2); + tmp->f.data = tmp->outbuf; + /* Reset tail pointer */ + tmp->tail -= sent; + if (tmp->tail) + memmove(tmp->buf, tmp->buf + sent, tmp->tail * 2); + +#if 0 + /* Save a sample frame */ + { static int samplefr = 0; + if (samplefr == 80) { + int fd; + fd = open("mp3.example", O_WRONLY | O_CREAT, 0644); + write(fd, tmp->f.data, tmp->f.datalen); + close(fd); + } + samplefr++; + } +#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; + pthread_mutex_lock(&localuser_lock); + res = ast_unregister_translator(&mp3tolin); + if (localusecnt) + res = -1; + pthread_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; +} |