diff options
author | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 1999-12-05 07:09:27 +0000 |
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committer | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 1999-12-05 07:09:27 +0000 |
commit | cf38740db3247ad69d6c6ad5a2b9693aadefca02 (patch) | |
tree | de8b284e53e13ef2e2a40b595eac7d961aed9d44 /formats/format_wav.c | |
parent | 8e2d6060131752f269b65bcb7bc5077a772a31c4 (diff) |
Version 0.1.0 from FTP
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'formats/format_wav.c')
-rwxr-xr-x | formats/format_wav.c | 355 |
1 files changed, 355 insertions, 0 deletions
diff --git a/formats/format_wav.c b/formats/format_wav.c new file mode 100755 index 000000000..e3265d379 --- /dev/null +++ b/formats/format_wav.c @@ -0,0 +1,355 @@ +/* + * Asterisk -- A telephony toolkit for Linux. + * + * Microsoft WAV File Format using libaudiofile + * + * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC + * + * Mark Spencer <markster@linux-support.net> + * + * This program is free software, distributed under the terms of + * the GNU General Public License + */ + +#include <asterisk/channel.h> +#include <asterisk/file.h> +#include <asterisk/logger.h> +#include <asterisk/sched.h> +#include <asterisk/module.h> +#include <arpa/inet.h> +#include <stdlib.h> +#include <stdio.h> +#include <unistd.h> +#include <errno.h> +#include <string.h> +#include <pthread.h> +#include <audiofile.h> + + +/* Read 320 samples at a time, max */ +#define WAV_MAX_SIZE 320 + +/* Fudge in milliseconds */ +#define WAV_FUDGE 2 + +struct ast_filestream { + /* First entry MUST be reserved for the channel type */ + void *reserved[AST_RESERVED_POINTERS]; + /* This is what a filestream means to us */ + int fd; /* Descriptor */ + /* Audio File */ + AFfilesetup afs; + AFfilehandle af; + int lasttimeout; + struct ast_channel *owner; + struct ast_filestream *next; + struct ast_frame fr; /* Frame information */ + char waste[AST_FRIENDLY_OFFSET]; /* Buffer for sending frames, etc */ + short samples[WAV_MAX_SIZE]; +}; + + +static struct ast_filestream *glist = NULL; +static pthread_mutex_t wav_lock = PTHREAD_MUTEX_INITIALIZER; +static int glistcnt = 0; + +static char *name = "wav"; +static char *desc = "Microsoft WAV format (PCM/16, 8000Hz mono)"; +static char *exts = "wav"; + +static struct ast_filestream *wav_open(int fd) +{ + /* We don't have any header to read or anything really, but + if we did, it would go here. We also might want to check + and be sure it's a valid file. */ + struct ast_filestream *tmp; + int notok = 0; + int fmt, width; + double rate; + if ((tmp = malloc(sizeof(struct ast_filestream)))) { + tmp->afs = afNewFileSetup(); + if (!tmp->afs) { + ast_log(LOG_WARNING, "Unable to create file setup\n"); + free(tmp); + return NULL; + } + afInitFileFormat(tmp->afs, AF_FILE_WAVE); + tmp->af = afOpenFD(fd, "r", tmp->afs); + if (!tmp->af) { + afFreeFileSetup(tmp->afs); + ast_log(LOG_WARNING, "Unable to open file descriptor\n"); + free(tmp); + return NULL; + } +#if 0 + afGetFileFormat(tmp->af, &version); + if (version != AF_FILE_WAVE) { + ast_log(LOG_WARNING, "This is not a wave file (%d)\n", version); + notok++; + } +#endif + /* Read the format and make sure it's exactly what we seek. */ + if (afGetChannels(tmp->af, AF_DEFAULT_TRACK) != 1) { + ast_log(LOG_WARNING, "Invalid number of channels %d. Should be mono (1)\n", afGetChannels(tmp->af, AF_DEFAULT_TRACK)); + notok++; + } + afGetSampleFormat(tmp->af, AF_DEFAULT_TRACK, &fmt, &width); + if (fmt != AF_SAMPFMT_TWOSCOMP) { + ast_log(LOG_WARNING, "Input file is not signed\n"); + notok++; + } + rate = afGetRate(tmp->af, AF_DEFAULT_TRACK); + if ((rate < 7900) || (rate > 8100)) { + ast_log(LOG_WARNING, "Rate %f is not close enough to 8000 Hz\n", rate); + notok++; + } + if (width != 16) { + ast_log(LOG_WARNING, "Input file is not 16-bit\n"); + notok++; + } + if (notok) { + afCloseFile(tmp->af); + afFreeFileSetup(tmp->afs); + free(tmp); + return NULL; + } + if (pthread_mutex_lock(&wav_lock)) { + afCloseFile(tmp->af); + afFreeFileSetup(tmp->afs); + ast_log(LOG_WARNING, "Unable to lock wav list\n"); + free(tmp); + return NULL; + } + tmp->next = glist; + glist = tmp; + tmp->fd = fd; + tmp->owner = NULL; + tmp->fr.data = tmp->samples; + tmp->fr.frametype = AST_FRAME_VOICE; + tmp->fr.subclass = AST_FORMAT_SLINEAR; + /* datalen will vary for each frame */ + tmp->fr.src = name; + tmp->fr.mallocd = 0; + tmp->lasttimeout = -1; + glistcnt++; + pthread_mutex_unlock(&wav_lock); + ast_update_use_count(); + } + return tmp; +} + +static struct ast_filestream *wav_rewrite(int fd, char *comment) +{ + /* We don't have any header to read or anything really, but + if we did, it would go here. We also might want to check + and be sure it's a valid file. */ + struct ast_filestream *tmp; + if ((tmp = malloc(sizeof(struct ast_filestream)))) { + tmp->afs = afNewFileSetup(); + if (!tmp->afs) { + ast_log(LOG_WARNING, "Unable to create file setup\n"); + free(tmp); + return NULL; + } + /* WAV format */ + afInitFileFormat(tmp->afs, AF_FILE_WAVE); + /* Mono */ + afInitChannels(tmp->afs, AF_DEFAULT_TRACK, 1); + /* Signed linear, 16-bit */ + afInitSampleFormat(tmp->afs, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16); + /* 8000 Hz */ + afInitRate(tmp->afs, AF_DEFAULT_TRACK, (double)8000.0); + tmp->af = afOpenFD(fd, "w", tmp->afs); + if (!tmp->af) { + afFreeFileSetup(tmp->afs); + ast_log(LOG_WARNING, "Unable to open file descriptor\n"); + free(tmp); + return NULL; + } + if (pthread_mutex_lock(&wav_lock)) { + ast_log(LOG_WARNING, "Unable to lock wav list\n"); + free(tmp); + return NULL; + } + tmp->next = glist; + glist = tmp; + tmp->fd = fd; + tmp->owner = NULL; + tmp->lasttimeout = -1; + glistcnt++; + pthread_mutex_unlock(&wav_lock); + ast_update_use_count(); + } else + ast_log(LOG_WARNING, "Out of memory\n"); + return tmp; +} + +static struct ast_frame *wav_read(struct ast_filestream *s) +{ + return NULL; +} + +static void wav_close(struct ast_filestream *s) +{ + struct ast_filestream *tmp, *tmpl = NULL; + if (pthread_mutex_lock(&wav_lock)) { + ast_log(LOG_WARNING, "Unable to lock wav list\n"); + return; + } + tmp = glist; + while(tmp) { + if (tmp == s) { + if (tmpl) + tmpl->next = tmp->next; + else + glist = tmp->next; + break; + } + tmpl = tmp; + tmp = tmp->next; + } + glistcnt--; + if (s->owner) { + s->owner->stream = NULL; + if (s->owner->streamid > -1) + ast_sched_del(s->owner->sched, s->owner->streamid); + s->owner->streamid = -1; + } + pthread_mutex_unlock(&wav_lock); + ast_update_use_count(); + if (!tmp) + ast_log(LOG_WARNING, "Freeing a filestream we don't seem to own\n"); + afCloseFile(tmp->af); + afFreeFileSetup(tmp->afs); + close(s->fd); + free(s); +} + +static int ast_read_callback(void *data) +{ + u_int32_t delay = -1; + int retval = 0; + int res; + struct ast_filestream *s = data; + /* Send a frame from the file to the appropriate channel */ + + if ((res = afReadFrames(s->af, AF_DEFAULT_TRACK, s->samples, sizeof(s->samples)/2)) < 1) { + if (res) + ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno)); + s->owner->streamid = -1; + return 0; + } + /* Per 8 samples, one milisecond */ + delay = res / 8; + s->fr.frametype = AST_FRAME_VOICE; + s->fr.subclass = AST_FORMAT_SLINEAR; + s->fr.offset = AST_FRIENDLY_OFFSET; + s->fr.datalen = res * 2; + s->fr.data = s->samples; + s->fr.mallocd = 0; + s->fr.timelen = delay; + /* Unless there is no delay, we're going to exit out as soon as we + have processed the current frame. */ + /* If there is a delay, lets schedule the next event */ + if (delay != s->lasttimeout) { + /* We'll install the next timeout now. */ + s->owner->streamid = ast_sched_add(s->owner->sched, + delay, + ast_read_callback, s); + + s->lasttimeout = delay; + } else { + /* Just come back again at the same time */ + retval = -1; + } + /* Lastly, process the frame */ + if (ast_write(s->owner, &s->fr)) { + ast_log(LOG_WARNING, "Failed to write frame\n"); + s->owner->streamid = -1; + return 0; + } + + return retval; +} + +static int wav_apply(struct ast_channel *c, struct ast_filestream *s) +{ + /* Select our owner for this stream, and get the ball rolling. */ + s->owner = c; + ast_read_callback(s); + return 0; +} + +static int wav_write(struct ast_filestream *fs, struct ast_frame *f) +{ + int res; + if (f->frametype != AST_FRAME_VOICE) { + ast_log(LOG_WARNING, "Asked to write non-voice frame!\n"); + return -1; + } + if (f->subclass != AST_FORMAT_SLINEAR) { + ast_log(LOG_WARNING, "Asked to write non-signed linear frame (%d)!\n", f->subclass); + return -1; + } + if ((res = afWriteFrames(fs->af, AF_DEFAULT_TRACK, f->data, f->datalen/2)) != f->datalen/2) { + ast_log(LOG_WARNING, "Unable to write frame: res=%d (%s)\n", res, strerror(errno)); + return -1; + } + return 0; +} + +char *wav_getcomment(struct ast_filestream *s) +{ + return NULL; +} + +int load_module() +{ + return ast_format_register(name, exts, AST_FORMAT_SLINEAR, + wav_open, + wav_rewrite, + wav_apply, + wav_write, + wav_read, + wav_close, + wav_getcomment); + + +} + +int unload_module() +{ + struct ast_filestream *tmp, *tmpl; + if (pthread_mutex_lock(&wav_lock)) { + ast_log(LOG_WARNING, "Unable to lock wav list\n"); + return -1; + } + tmp = glist; + while(tmp) { + if (tmp->owner) + ast_softhangup(tmp->owner); + tmpl = tmp; + tmp = tmp->next; + free(tmpl); + } + pthread_mutex_unlock(&wav_lock); + return ast_format_unregister(name); +} + +int usecount() +{ + int res; + if (pthread_mutex_lock(&wav_lock)) { + ast_log(LOG_WARNING, "Unable to lock wav list\n"); + return -1; + } + res = glistcnt; + pthread_mutex_unlock(&wav_lock); + return res; +} + +char *description() +{ + return desc; +} + |