diff options
author | rizzo <rizzo@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-04-04 12:59:25 +0000 |
---|---|---|
committer | rizzo <rizzo@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-04-04 12:59:25 +0000 |
commit | 217ea2f1ea680d457c19aac1b563077f1f9ae67c (patch) | |
tree | c2ca953ad9722e235aa0205ca61e6b6a59c5e9e5 /formats/format_wav.c | |
parent | 577bc5cf68911578b9a529d3919a923f5a5b3436 (diff) |
Largely simplify format handlers (for file copy etc.)
collecting common functions in a single place and removing
them from the individual handlers.
The full description is on mantis,
http://bugs.digium.com/view.php?id=6375
and only the ogg_vorbis handler needs to be converted to
the new structure.
As a result of this change, format_au.c and format_pcm_alaw.c
should go away (in a separate commit) as their functionality
(trivial) has been merged in another file.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@17243 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'formats/format_wav.c')
-rw-r--r-- | formats/format_wav.c | 230 |
1 files changed, 85 insertions, 145 deletions
diff --git a/formats/format_wav.c b/formats/format_wav.c index 67df4163b..f46d75b1b 100644 --- a/formats/format_wav.c +++ b/formats/format_wav.c @@ -49,30 +49,16 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") /* Portions of the conversion code are by guido@sienanet.it */ -struct ast_filestream { - void *reserved[AST_RESERVED_POINTERS]; - /* This is what a filestream means to us */ - FILE *f; /* Descriptor */ +#define WAV_BUF_SIZE 320 + +struct wav_desc { /* format-specific parameters */ int bytes; int needsgain; - struct ast_frame fr; /* Frame information */ - char waste[AST_FRIENDLY_OFFSET]; /* Buffer for sending frames, etc */ - char empty; /* Empty character */ - short buf[160]; - int foffset; int lasttimeout; int maxlen; struct timeval last; }; - -AST_MUTEX_DEFINE_STATIC(wav_lock); -static int glistcnt = 0; - -static char *name = "wav"; -static char *desc = "Microsoft WAV format (8000hz Signed Linear)"; -static char *exts = "wav"; - #define BLOCKSIZE 160 #define GAIN 2 /* 2^GAIN is the multiple to increase the volume by */ @@ -165,7 +151,7 @@ static int check_header(FILE *f) ast_log(LOG_WARNING, "Read failed (freq)\n"); return -1; } - if (ltohl(freq) != 8000) { + if (ltohl(freq) != DEFAULT_SAMPLE_RATE) { ast_log(LOG_WARNING, "Unexpected freqency %d\n", ltohl(freq)); return -1; } @@ -233,7 +219,6 @@ static int update_header(FILE *f) off_t cur,end; int datalen,filelen,bytes; - cur = ftello(f); fseek(f, 0, SEEK_END); end = ftello(f); @@ -333,135 +318,90 @@ static int write_header(FILE *f) return 0; } -static struct ast_filestream *wav_open(FILE *f) +static int wav_open(struct ast_filestream *s) { /* 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)))) { - memset(tmp, 0, sizeof(struct ast_filestream)); - if ((tmp->maxlen = check_header(f)) < 0) { - free(tmp); - return NULL; - } - if (ast_mutex_lock(&wav_lock)) { - ast_log(LOG_WARNING, "Unable to lock wav list\n"); - free(tmp); - return NULL; - } - tmp->f = f; - tmp->needsgain = 1; - tmp->fr.data = tmp->buf; - 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->bytes = 0; - glistcnt++; - ast_mutex_unlock(&wav_lock); - ast_update_use_count(); - } - return tmp; + struct wav_desc *tmp = (struct wav_desc *)s->private; + if ((tmp->maxlen = check_header(s->f)) < 0) + return -1; + return 0; } -static struct ast_filestream *wav_rewrite(FILE *f, const char *comment) +static int wav_rewrite(struct ast_filestream *s, const 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)))) { - memset(tmp, 0, sizeof(struct ast_filestream)); - if (write_header(f)) { - free(tmp); - return NULL; - } - if (ast_mutex_lock(&wav_lock)) { - ast_log(LOG_WARNING, "Unable to lock wav list\n"); - free(tmp); - return NULL; - } - tmp->f = f; - glistcnt++; - ast_mutex_unlock(&wav_lock); - ast_update_use_count(); - } else - ast_log(LOG_WARNING, "Out of memory\n"); - return tmp; + + if (write_header(s->f)) + return -1; + return 0; } static void wav_close(struct ast_filestream *s) { char zero = 0; - if (ast_mutex_lock(&wav_lock)) { - ast_log(LOG_WARNING, "Unable to lock wav list\n"); - return; - } - glistcnt--; - ast_mutex_unlock(&wav_lock); - ast_update_use_count(); + struct wav_desc *fs = (struct wav_desc *)s->private; /* Pad to even length */ - if (s->bytes & 0x1) + if (fs->bytes & 0x1) fwrite(&zero, 1, 1, s->f); - fclose(s->f); - free(s); - s = NULL; } static struct ast_frame *wav_read(struct ast_filestream *s, int *whennext) { int res; - int delay; + int samples; /* actual samples read */ int x; - short tmp[sizeof(s->buf) / 2]; - int bytes = sizeof(tmp); + short *tmp; + int bytes = WAV_BUF_SIZE; /* in bytes */ off_t here; /* Send a frame from the file to the appropriate channel */ + struct wav_desc *fs = (struct wav_desc *)s->private; + here = ftello(s->f); - if ((s->maxlen - here) < bytes) - bytes = s->maxlen - here; + if (fs->maxlen - here < bytes) /* truncate if necessary */ + bytes = fs->maxlen - here; if (bytes < 0) bytes = 0; /* ast_log(LOG_DEBUG, "here: %d, maxlen: %d, bytes: %d\n", here, s->maxlen, bytes); */ + s->fr.frametype = AST_FRAME_VOICE; + s->fr.subclass = AST_FORMAT_SLINEAR; + s->fr.mallocd = 0; + FR_SET_BUF(&s->fr, s->buf, AST_FRIENDLY_OFFSET, bytes); - if ( (res = fread(tmp, 1, bytes, s->f)) <= 0 ) { - if (res) { + if ( (res = fread(s->fr.data, 1, s->fr.datalen, s->f)) <= 0 ) { + if (res) ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno)); - } return NULL; } + s->fr.datalen = res; + s->fr.samples = samples = res / 2; + tmp = (short *)(s->fr.data); #if __BYTE_ORDER == __BIG_ENDIAN - for( x = 0; x < sizeof(tmp)/2; x++) tmp[x] = (tmp[x] << 8) | ((tmp[x] & 0xff00) >> 8); + /* file format is little endian so we need to swap */ + for( x = 0; x < samples; x++) + tmp[x] = (tmp[x] << 8) | ((tmp[x] & 0xff00) >> 8); #endif - if (s->needsgain) { - for (x=0;x<sizeof(tmp)/2;x++) + if (fs->needsgain) { + for (x=0; x < samples; x++) { if (tmp[x] & ((1 << GAIN) - 1)) { /* If it has data down low, then it's not something we've artificially increased gain on, so we don't need to gain adjust it */ - s->needsgain = 0; + fs->needsgain = 0; + break; } - } - if (s->needsgain) { - for (x=0;x<sizeof(tmp)/2;x++) { - s->buf[x] = tmp[x] >> GAIN; } - } else { - memcpy(s->buf, tmp, sizeof(s->buf)); + if (fs->needsgain) { + for (x=0; x < samples; x++) + tmp[x] = tmp[x] >> GAIN; + } } - delay = res / 2; - s->fr.frametype = AST_FRAME_VOICE; - s->fr.subclass = AST_FORMAT_SLINEAR; - s->fr.offset = AST_FRIENDLY_OFFSET; - s->fr.datalen = res; - s->fr.data = s->buf; - s->fr.mallocd = 0; - s->fr.samples = delay; - *whennext = delay; + *whennext = samples; return &s->fr; } @@ -470,6 +410,9 @@ static int wav_write(struct ast_filestream *fs, struct ast_frame *f) int x; short tmp[8000], *tmpi; float tmpf; + struct wav_desc *s = (struct wav_desc *)fs->private; + int res; + if (f->frametype != AST_FRAME_VOICE) { ast_log(LOG_WARNING, "Asked to write non-voice frame!\n"); return -1; @@ -489,33 +432,28 @@ static int wav_write(struct ast_filestream *fs, struct ast_frame *f) printf("Data Length: %d\n", f->datalen); #endif - if (fs->buf) { - tmpi = f->data; - /* Volume adjust here to accomodate */ - for (x=0;x<f->datalen/2;x++) { - tmpf = ((float)tmpi[x]) * ((float)(1 << GAIN)); - if (tmpf > 32767.0) - tmpf = 32767.0; - if (tmpf < -32768.0) - tmpf = -32768.0; - tmp[x] = tmpf; - tmp[x] &= ~((1 << GAIN) - 1); + tmpi = f->data; + /* Volume adjust here to accomodate */ + for (x=0;x<f->datalen/2;x++) { + tmpf = ((float)tmpi[x]) * ((float)(1 << GAIN)); + if (tmpf > 32767.0) + tmpf = 32767.0; + if (tmpf < -32768.0) + tmpf = -32768.0; + tmp[x] = tmpf; + tmp[x] &= ~((1 << GAIN) - 1); #if __BYTE_ORDER == __BIG_ENDIAN - tmp[x] = (tmp[x] << 8) | ((tmp[x] & 0xff00) >> 8); + tmp[x] = (tmp[x] << 8) | ((tmp[x] & 0xff00) >> 8); #endif - } - if ((fwrite(tmp, 1, f->datalen, fs->f) != f->datalen) ) { - ast_log(LOG_WARNING, "Bad write (%d): %s\n", errno, strerror(errno)); - return -1; - } - } else { - ast_log(LOG_WARNING, "Cannot write data to file.\n"); + } + if ((res = fwrite(tmp, 1, f->datalen, fs->f)) != f->datalen ) { + ast_log(LOG_WARNING, "Bad write (%d): %s\n", res, strerror(errno)); return -1; } - - fs->bytes += f->datalen; + + s->bytes += f->datalen; update_header(fs->f); return 0; @@ -560,43 +498,45 @@ static off_t wav_tell(struct ast_filestream *fs) return (offset - 44)/2; } -static char *wav_getcomment(struct ast_filestream *s) -{ - return NULL; -} +static struct ast_format_lock me = { .usecnt = -1 }; + +static const struct ast_format wav_f = { + .name = "wav", + .exts = "wav", + .format = AST_FORMAT_SLINEAR, + .open = wav_open, + .rewrite = wav_rewrite, + .write = wav_write, + .seek = wav_seek, + .trunc = wav_trunc, + .tell = wav_tell, + .read = wav_read, + .close = wav_close, + .buf_size = WAV_BUF_SIZE + AST_FRIENDLY_OFFSET, + .desc_size = sizeof(struct wav_desc), + .lockp = &me, +}; int load_module() { - return ast_format_register(name, exts, AST_FORMAT_SLINEAR, - wav_open, - wav_rewrite, - wav_write, - wav_seek, - wav_trunc, - wav_tell, - wav_read, - wav_close, - wav_getcomment); - - + return ast_format_register(&wav_f); } int unload_module() { - return ast_format_unregister(name); + return ast_format_unregister(wav_f.name); } int usecount() { - return glistcnt; + return me.usecnt; } char *description() { - return desc; + return "Microsoft WAV format (8000hz Signed Linear)"; } - char *key() { return ASTERISK_GPL_KEY; |