diff --git a/codecs/codec_mp3_d.c b/codecs/codec_mp3_d.c
index aad6d75703cd2e08ef640f810d81b51ef1ce439f..d271dfe51242a7ea1e3fedaf3da729bc7c7ee9ad 100755
--- a/codecs/codec_mp3_d.c
+++ b/codecs/codec_mp3_d.c
@@ -17,6 +17,7 @@
#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>
@@ -108,40 +109,30 @@ static struct ast_frame *mp3tolin_sample()
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;
+ tmp->f.datalen = tmp->tail * 2;
/* Assume 8000 Hz */
- tmp->f.timelen = sent / 8;
+ tmp->f.timelen = tmp->tail / 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;
+ tmp->f.data = tmp->buf;
/* Reset tail pointer */
- tmp->tail -= sent;
- if (tmp->tail)
- memmove(tmp->buf, tmp->buf + sent, tmp->tail * 2);
+ tmp->tail = 0;
#if 0
/* Save a sample frame */
- { static int samplefr = 0;
- if (samplefr == 80) {
- int fd;
- fd = open("mp3.example", O_WRONLY | O_CREAT, 0644);
+ {
+ 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);
- close(fd);
}
- samplefr++;
- }
#endif
return &tmp->f;
}
@@ -295,11 +286,11 @@ static struct ast_translator mp3tolin =
int unload_module(void)
{
int res;
- pthread_mutex_lock(&localuser_lock);
+ ast_pthread_mutex_lock(&localuser_lock);
res = ast_unregister_translator(&mp3tolin);
if (localusecnt)
res = -1;
- pthread_mutex_unlock(&localuser_lock);
+ ast_pthread_mutex_unlock(&localuser_lock);
return res;
}
diff --git a/formats/format_mp3.c b/formats/format_mp3.c
index a3e25d45bc202ca3d34b1bd19d65a6d73414a57e..83c8e1c46c2a5ecadcb73d39dbf0c226e65c350b 100755
--- a/formats/format_mp3.c
+++ b/formats/format_mp3.c
@@ -62,7 +62,7 @@ static struct ast_filestream *mp3_open(int fd)
and be sure it's a valid file. */
struct ast_filestream *tmp;
if ((tmp = malloc(sizeof(struct ast_filestream)))) {
- if (pthread_mutex_lock(&mp3_lock)) {
+ if (ast_pthread_mutex_lock(&mp3_lock)) {
ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
free(tmp);
return NULL;
@@ -76,7 +76,7 @@ static struct ast_filestream *mp3_open(int fd)
tmp->last.tv_sec = 0;
tmp->adj = 0;
glistcnt++;
- pthread_mutex_unlock(&mp3_lock);
+ ast_pthread_mutex_unlock(&mp3_lock);
ast_update_use_count();
}
return tmp;
@@ -89,7 +89,7 @@ static struct ast_filestream *mp3_rewrite(int fd, char *comment)
and be sure it's a valid file. */
struct ast_filestream *tmp;
if ((tmp = malloc(sizeof(struct ast_filestream)))) {
- if (pthread_mutex_lock(&mp3_lock)) {
+ if (ast_pthread_mutex_lock(&mp3_lock)) {
ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
free(tmp);
return NULL;
@@ -99,7 +99,7 @@ static struct ast_filestream *mp3_rewrite(int fd, char *comment)
tmp->fd = fd;
tmp->owner = NULL;
glistcnt++;
- pthread_mutex_unlock(&mp3_lock);
+ ast_pthread_mutex_unlock(&mp3_lock);
ast_update_use_count();
} else
ast_log(LOG_WARNING, "Out of memory\n");
@@ -114,7 +114,7 @@ static struct ast_frame *mp3_read(struct ast_filestream *s)
static void mp3_close(struct ast_filestream *s)
{
struct ast_filestream *tmp, *tmpl = NULL;
- if (pthread_mutex_lock(&mp3_lock)) {
+ if (ast_pthread_mutex_lock(&mp3_lock)) {
ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
return;
}
@@ -137,7 +137,7 @@ static void mp3_close(struct ast_filestream *s)
ast_sched_del(s->owner->sched, s->owner->streamid);
s->owner->streamid = -1;
}
- pthread_mutex_unlock(&mp3_lock);
+ ast_pthread_mutex_unlock(&mp3_lock);
ast_update_use_count();
if (!tmp)
ast_log(LOG_WARNING, "Freeing a filestream we don't seem to own\n");
@@ -187,8 +187,9 @@ static int ast_read_callback(void *data)
gettimeofday(&tv, NULL);
ms = ((tv.tv_usec - s->last.tv_usec) / 1000 + (tv.tv_sec - s->last.tv_sec) * 1000);
/* If we're within 2 milliseconds, that's close enough */
- if ((ms - delay) * (ms - delay) > 4)
+ if ((ms - delay) > 0 )
s->adj -= (ms - delay);
+ s->adj -= 2;
}
s->fr.timelen = delay;
#if 0
@@ -260,7 +261,7 @@ int load_module()
int unload_module()
{
struct ast_filestream *tmp, *tmpl;
- if (pthread_mutex_lock(&mp3_lock)) {
+ if (ast_pthread_mutex_lock(&mp3_lock)) {
ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
return -1;
}
@@ -272,19 +273,19 @@ int unload_module()
tmp = tmp->next;
free(tmpl);
}
- pthread_mutex_unlock(&mp3_lock);
+ ast_pthread_mutex_unlock(&mp3_lock);
return ast_format_unregister(name);
}
int usecount()
{
int res;
- if (pthread_mutex_lock(&mp3_lock)) {
+ if (ast_pthread_mutex_lock(&mp3_lock)) {
ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
return -1;
}
res = glistcnt;
- pthread_mutex_unlock(&mp3_lock);
+ ast_pthread_mutex_unlock(&mp3_lock);
return res;
}
diff --git a/formats/format_wav.c b/formats/format_wav.c
index 071c24f26fccfb1e8dd0c63dd9253acc33d7580d..77ac71916d0bf34f4fb1e8260677ebe5aac7dc09 100755
--- a/formats/format_wav.c
+++ b/formats/format_wav.c
@@ -1,7 +1,7 @@
/*
* Asterisk -- A telephony toolkit for Linux.
*
- * Microsoft WAV File Format using libaudiofile
+ * Work with WAV in the proprietary Microsoft format.
*
* Copyright (C) 1999, Mark Spencer
*
@@ -18,34 +18,35 @@
#include <asterisk/module.h>
#include <arpa/inet.h>
#include <stdlib.h>
+#include <sys/time.h>
#include <stdio.h>
#include <unistd.h>
#include <errno.h>
#include <string.h>
#include <pthread.h>
-#include <audiofile.h>
+#include <endian.h>
+/* Some Ideas for this code came from makewave.c by Jeffery Chilton */
-/* Read 320 samples at a time, max */
-#define WAV_MAX_SIZE 320
-
-/* Fudge in milliseconds */
-#define WAV_FUDGE 2
+/* Portions of the conversion code are by guido@sienanet.it */
struct ast_filestream {
- /* First entry MUST be reserved for the channel type */
void *reserved[AST_RESERVED_POINTERS];
+ /* Believe it or not, we must decode/recode to account for the
+ weird MS format */
/* This is what a filestream means to us */
int fd; /* Descriptor */
- /* Audio File */
- AFfilesetup afs;
- AFfilehandle af;
- int lasttimeout;
+ int bytes;
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];
+ char empty; /* Empty character */
+ short buf[160]; /* Two Real GSM Frames */
+ int foffset;
+ int lasttimeout;
+ struct timeval last;
+ int adj;
+ struct ast_filestream *next;
};
@@ -54,68 +55,248 @@ 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 *desc = "Microsoft WAV format (8000hz Signed Linear)";
static char *exts = "wav";
+#define BLOCKSIZE 160
+
+#if __BYTE_ORDER == __LITTLE_ENDIAN
+#define htoll(b) (b)
+#define htols(b) (b)
+#define ltohl(b) (b)
+#define ltohs(b) (b)
+#else
+#if __BYTE_ORDER == __BIG_ENDIAN
+#define htoll(b) \
+ (((((b) ) & 0xFF) << 24) | \
+ ((((b) >> 8) & 0xFF) << 16) | \
+ ((((b) >> 16) & 0xFF) << 8) | \
+ ((((b) >> 24) & 0xFF) ))
+#define htols(b) \
+ (((((b) ) & 0xFF) << 8) | \
+ ((((b) >> 8) & 0xFF) ))
+#define ltohl(b) htoll(b)
+#define ltohs(b) htols(b)
+#else
+#error "Endianess not defined"
+#endif
+#endif
+
+
+static int check_header(int fd)
+{
+ int type, size, formtype;
+ int fmt, hsize;
+ short format, chans, bysam, bisam;
+ int bysec;
+ int freq;
+ int data;
+ if (read(fd, &type, 4) != 4) {
+ ast_log(LOG_WARNING, "Read failed (type)\n");
+ return -1;
+ }
+ if (read(fd, &size, 4) != 4) {
+ ast_log(LOG_WARNING, "Read failed (size)\n");
+ return -1;
+ }
+ size = ltohl(size);
+ if (read(fd, &formtype, 4) != 4) {
+ ast_log(LOG_WARNING, "Read failed (formtype)\n");
+ return -1;
+ }
+ if (memcmp(&type, "RIFF", 4)) {
+ ast_log(LOG_WARNING, "Does not begin with RIFF\n");
+ return -1;
+ }
+ if (memcmp(&formtype, "WAVE", 4)) {
+ ast_log(LOG_WARNING, "Does not contain WAVE\n");
+ return -1;
+ }
+ if (read(fd, &fmt, 4) != 4) {
+ ast_log(LOG_WARNING, "Read failed (fmt)\n");
+ return -1;
+ }
+ if (memcmp(&fmt, "fmt ", 4)) {
+ ast_log(LOG_WARNING, "Does not say fmt\n");
+ return -1;
+ }
+ if (read(fd, &hsize, 4) != 4) {
+ ast_log(LOG_WARNING, "Read failed (formtype)\n");
+ return -1;
+ }
+ if (ltohl(hsize) != 16) {
+ ast_log(LOG_WARNING, "Unexpected header size %d\n", ltohl(hsize));
+ return -1;
+ }
+ if (read(fd, &format, 2) != 2) {
+ ast_log(LOG_WARNING, "Read failed (format)\n");
+ return -1;
+ }
+ if (ltohs(format) != 1) {
+ ast_log(LOG_WARNING, "Not a wav file %d\n", ltohs(format));
+ return -1;
+ }
+ if (read(fd, &chans, 2) != 2) {
+ ast_log(LOG_WARNING, "Read failed (format)\n");
+ return -1;
+ }
+ if (ltohs(chans) != 1) {
+ ast_log(LOG_WARNING, "Not in mono %d\n", ltohs(chans));
+ return -1;
+ }
+ if (read(fd, &freq, 4) != 4) {
+ ast_log(LOG_WARNING, "Read failed (freq)\n");
+ return -1;
+ }
+ if (ltohl(freq) != 8000) {
+ ast_log(LOG_WARNING, "Unexpected freqency %d\n", ltohl(freq));
+ return -1;
+ }
+ /* Ignore the byte frequency */
+ if (read(fd, &bysec, 4) != 4) {
+ ast_log(LOG_WARNING, "Read failed (BYTES_PER_SECOND)\n");
+ return -1;
+ }
+ /* Check bytes per sample */
+ if (read(fd, &bysam, 2) != 2) {
+ ast_log(LOG_WARNING, "Read failed (BYTES_PER_SAMPLE)\n");
+ return -1;
+ }
+ if (ltohs(bysam) != 2) {
+ ast_log(LOG_WARNING, "Can only handle 16bits per sample: %d\n", ltohs(bysam));
+ return -1;
+ }
+ if (read(fd, &bisam, 2) != 2) {
+ ast_log(LOG_WARNING, "Read failed (Bits Per Sample): &d\n", ltohs(bisam));
+ return -1;
+ }
+ /* Begin data chunk */
+ if (read(fd, &data, 4) != 4) {
+ ast_log(LOG_WARNING, "Read failed (data)\n");
+ return -1;
+ }
+ if (memcmp(&data, "data", 4)) {
+ ast_log(LOG_WARNING, "Does not say data\n");
+ return -1;
+ }
+ /* Ignore the data length */
+ if (read(fd, &data, 4) != 4) {
+ ast_log(LOG_WARNING, "Read failed (data)\n");
+ return -1;
+ }
+ return 0;
+}
+
+static int update_header(int fd, int bytes)
+{
+ int cur;
+ int datalen = htoll(bytes);
+ /* int filelen = htoll(52 + ((bytes + 1) & ~0x1)); */
+ int filelen = htoll(36 + bytes);
+
+ cur = lseek(fd, 0, SEEK_CUR);
+ if (cur < 0) {
+ ast_log(LOG_WARNING, "Unable to find our position\n");
+ return -1;
+ }
+ if (lseek(fd, 4, SEEK_SET) != 4) {
+ ast_log(LOG_WARNING, "Unable to set our position\n");
+ return -1;
+ }
+ if (write(fd, &filelen, 4) != 4) {
+ ast_log(LOG_WARNING, "Unable to set write file size\n");
+ return -1;
+ }
+ if (lseek(fd, 40, SEEK_SET) != 40) {
+ ast_log(LOG_WARNING, "Unable to set our position\n");
+ return -1;
+ }
+ if (write(fd, &datalen, 4) != 4) {
+ ast_log(LOG_WARNING, "Unable to set write datalen\n");
+ return -1;
+ }
+ if (lseek(fd, cur, SEEK_SET) != cur) {
+ ast_log(LOG_WARNING, "Unable to return to position\n");
+ return -1;
+ }
+ return 0;
+}
+
+static int write_header(int fd)
+{
+ unsigned int hz=htoll(8000);
+ unsigned int bhz = htoll(16000);
+ unsigned int hs = htoll(16);
+ unsigned short fmt = htols(1);
+ unsigned short chans = htols(1);
+ unsigned short bysam = htols(2);
+ unsigned short bisam = htols(16);
+ unsigned int size = htoll(0);
+ /* Write a wav header, ignoring sizes which will be filled in later */
+ if (write(fd, "RIFF", 4) != 4) {
+ ast_log(LOG_WARNING, "Unable to write header\n");
+ return -1;
+ }
+ if (write(fd, &size, 4) != 4) {
+ ast_log(LOG_WARNING, "Unable to write header\n");
+ return -1;
+ }
+ if (write(fd, "WAVEfmt ", 8) != 8) {
+ ast_log(LOG_WARNING, "Unable to write header\n");
+ return -1;
+ }
+ if (write(fd, &hs, 4) != 4) {
+ ast_log(LOG_WARNING, "Unable to write header\n");
+ return -1;
+ }
+ if (write(fd, &fmt, 2) != 2) {
+ ast_log(LOG_WARNING, "Unable to write header\n");
+ return -1;
+ }
+ if (write(fd, &chans, 2) != 2) {
+ ast_log(LOG_WARNING, "Unable to write header\n");
+ return -1;
+ }
+ if (write(fd, &hz, 4) != 4) {
+ ast_log(LOG_WARNING, "Unable to write header\n");
+ return -1;
+ }
+ if (write(fd, &bhz, 4) != 4) {
+ ast_log(LOG_WARNING, "Unable to write header\n");
+ return -1;
+ }
+ if (write(fd, &bysam, 2) != 2) {
+ ast_log(LOG_WARNING, "Unable to write header\n");
+ return -1;
+ }
+ if (write(fd, &bisam, 2) != 2) {
+ ast_log(LOG_WARNING, "Unable to write header\n");
+ return -1;
+ }
+ if (write(fd, "data", 4) != 4) {
+ ast_log(LOG_WARNING, "Unable to write header\n");
+ return -1;
+ }
+ if (write(fd, &size, 4) != 4) {
+ ast_log(LOG_WARNING, "Unable to write header\n");
+ return -1;
+ }
+ return 0;
+}
+
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");
+ memset(tmp, 0, sizeof(struct ast_filestream));
+ if (check_header(fd)) {
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);
+ if (ast_pthread_mutex_lock(&wav_lock)) {
ast_log(LOG_WARNING, "Unable to lock wav list\n");
free(tmp);
return NULL;
@@ -124,15 +305,16 @@ static struct ast_filestream *wav_open(int fd)
glist = tmp;
tmp->fd = fd;
tmp->owner = NULL;
- tmp->fr.data = tmp->samples;
+ 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->lasttimeout = -1;
+ tmp->bytes = 0;
glistcnt++;
- pthread_mutex_unlock(&wav_lock);
+ ast_pthread_mutex_unlock(&wav_lock);
ast_update_use_count();
}
return tmp;
@@ -145,28 +327,12 @@ static struct ast_filestream *wav_rewrite(int fd, char *comment)
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");
+ memset(tmp, 0, sizeof(struct ast_filestream));
+ if (write_header(fd)) {
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)) {
+ if (ast_pthread_mutex_lock(&wav_lock)) {
ast_log(LOG_WARNING, "Unable to lock wav list\n");
free(tmp);
return NULL;
@@ -177,7 +343,7 @@ static struct ast_filestream *wav_rewrite(int fd, char *comment)
tmp->owner = NULL;
tmp->lasttimeout = -1;
glistcnt++;
- pthread_mutex_unlock(&wav_lock);
+ ast_pthread_mutex_unlock(&wav_lock);
ast_update_use_count();
} else
ast_log(LOG_WARNING, "Out of memory\n");
@@ -192,7 +358,8 @@ static struct ast_frame *wav_read(struct ast_filestream *s)
static void wav_close(struct ast_filestream *s)
{
struct ast_filestream *tmp, *tmpl = NULL;
- if (pthread_mutex_lock(&wav_lock)) {
+ char zero = 0;
+ if (ast_pthread_mutex_lock(&wav_lock)) {
ast_log(LOG_WARNING, "Unable to lock wav list\n");
return;
}
@@ -215,54 +382,54 @@ static void wav_close(struct ast_filestream *s)
ast_sched_del(s->owner->sched, s->owner->streamid);
s->owner->streamid = -1;
}
- pthread_mutex_unlock(&wav_lock);
+ ast_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);
+ /* Pad to even length */
+ if (s->bytes & 0x1)
+ write(s->fd, &zero, 1);
close(s->fd);
free(s);
+#if 0
+ printf("bytes = %d\n", s->bytes);
+#endif
}
static int ast_read_callback(void *data)
{
- u_int32_t delay = -1;
int retval = 0;
int res;
+ int delay;
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)
+
+ if ( (res = read(s->fd, s->buf, sizeof(s->buf))) <= 0 ) {
+ 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;
+
+ delay = res / 16;
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.datalen = res;
+ s->fr.data = s->buf;
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 */
+
+ /* Lastly, process the frame */
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->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;
@@ -282,20 +449,38 @@ static int wav_apply(struct ast_channel *c, struct ast_filestream *s)
static int wav_write(struct ast_filestream *fs, struct ast_frame *f)
{
- int res;
+ int res = 0;
+#if 0
+ int size = 0;
+#endif
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);
+ ast_log(LOG_WARNING, "Asked to write non-GSM 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));
+
+#if 0
+ printf("Data Length: %d\n", f->datalen);
+#endif
+
+ if (fs->buf) {
+ if ((write (fs->fd, f->data, f->datalen) != f->datalen) ) {
+ ast_log(LOG_WARNING, "Bad write (%d): %s\n", res, strerror(errno));
+ return -1;
+ }
+ } else {
+ ast_log(LOG_WARNING, "Cannot write data to file.\n");
return -1;
- }
+ }
+
+ fs->bytes += f->datalen;
+ update_header(fs->fd, fs->bytes);
+
return 0;
+
}
static char *wav_getcomment(struct ast_filestream *s)
@@ -312,7 +497,7 @@ int load_module()
wav_write,
wav_read,
wav_close,
- wav_getcomment);
+ wav_getcomment);
}
@@ -320,7 +505,7 @@ int load_module()
int unload_module()
{
struct ast_filestream *tmp, *tmpl;
- if (pthread_mutex_lock(&wav_lock)) {
+ if (ast_pthread_mutex_lock(&wav_lock)) {
ast_log(LOG_WARNING, "Unable to lock wav list\n");
return -1;
}
@@ -332,19 +517,19 @@ int unload_module()
tmp = tmp->next;
free(tmpl);
}
- pthread_mutex_unlock(&wav_lock);
+ ast_pthread_mutex_unlock(&wav_lock);
return ast_format_unregister(name);
}
int usecount()
{
int res;
- if (pthread_mutex_lock(&wav_lock)) {
+ if (ast_pthread_mutex_lock(&wav_lock)) {
ast_log(LOG_WARNING, "Unable to lock wav list\n");
return -1;
}
res = glistcnt;
- pthread_mutex_unlock(&wav_lock);
+ ast_pthread_mutex_unlock(&wav_lock);
return res;
}