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authorkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2005-04-03 22:57:18 +0000
committerkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2005-04-03 22:57:18 +0000
commit4ce1edc05cec917ad15513344d46fed1e35b6f58 (patch)
tree0377d7238fbc8640a0885f84f0b381aaef09d78c
parentbbbb1d493dfcc49a639c0d6fda114322041a4f2b (diff)
handle AST_FORMAT_SLINEAR endianness properly on big-endian systems (bug #3865)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@5373 f38db490-d61c-443f-a65b-d21fe96a405b
-rwxr-xr-xchannels/chan_iax2.c7
-rwxr-xr-xchannels/chan_phone.c21
-rwxr-xr-xchannels/iax2-parser.c9
-rwxr-xr-xframe.c17
-rwxr-xr-xinclude/asterisk/frame.h23
-rwxr-xr-xrtp.c14
6 files changed, 80 insertions, 11 deletions
diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index 6135ca4a1..6ea03392d 100755
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -7337,9 +7337,12 @@ retryowner2:
f.src = "IAX2";
f.mallocd = 0;
f.offset = 0;
- if (f.datalen && (f.frametype == AST_FRAME_VOICE))
+ if (f.datalen && (f.frametype == AST_FRAME_VOICE)) {
f.samples = get_samples(&f);
- else
+ /* We need to byteswap incoming slinear samples from network byte order */
+ if (f.subclass == AST_FORMAT_SLINEAR)
+ ast_frame_byteswap_be(&f);
+ } else
f.samples = 0;
iax_frame_wrap(&fr, &f);
diff --git a/channels/chan_phone.c b/channels/chan_phone.c
index 5cee02e6f..8e5bdb523 100755
--- a/channels/chan_phone.c
+++ b/channels/chan_phone.c
@@ -544,10 +544,13 @@ static struct ast_frame *phone_read(struct ast_channel *ast)
: AST_FRAME_VIDEO;
p->fr.subclass = p->lastinput;
p->fr.offset = AST_FRIENDLY_OFFSET;
+ /* Byteswap from little-endian to native-endian */
+ if (p->fr.subclass == AST_FORMAT_SLINEAR)
+ ast_frame_byteswap_le(&p->fr);
return &p->fr;
}
-static int phone_write_buf(struct phone_pvt *p, const char *buf, int len, int frlen)
+static int phone_write_buf(struct phone_pvt *p, const char *buf, int len, int frlen, int swap)
{
int res;
/* Store as much of the buffer as we can, then write fixed frames */
@@ -555,7 +558,10 @@ static int phone_write_buf(struct phone_pvt *p, const char *buf, int len, int fr
/* Make sure we have enough buffer space to store the frame */
if (space < len)
len = space;
- memcpy(p->obuf + p->obuflen, buf, len);
+ if (swap)
+ ast_memcpy_byteswap(p->obuf+p->obuflen, buf, len/2);
+ else
+ memcpy(p->obuf + p->obuflen, buf, len);
p->obuflen += len;
while(p->obuflen > frlen) {
res = write(p->fd, p->obuf, frlen);
@@ -581,7 +587,7 @@ static int phone_write_buf(struct phone_pvt *p, const char *buf, int len, int fr
static int phone_send_text(struct ast_channel *ast, const char *text)
{
int length = strlen(text);
- return phone_write_buf(ast->tech_pvt, text, length, length) ==
+ return phone_write_buf(ast->tech_pvt, text, length, length, 0) ==
length ? 0 : -1;
}
@@ -729,12 +735,17 @@ static int phone_write(struct ast_channel *ast, struct ast_frame *frame)
memset(tmpbuf + 4, 0, sizeof(tmpbuf) - 4);
memcpy(tmpbuf, frame->data, 4);
expected = 24;
- res = phone_write_buf(p, tmpbuf, expected, maxfr);
+ res = phone_write_buf(p, tmpbuf, expected, maxfr, 0);
}
res = 4;
expected=4;
} else {
- res = phone_write_buf(p, pos, expected, maxfr);
+ int swap = 0;
+#if __BYTE_ORDER == __BIG_ENDIAN
+ if (frame->subclass == AST_FORMAT_SLINEAR)
+ swap = 1; /* Swap big-endian samples to little-endian as we copy */
+#endif
+ res = phone_write_buf(p, pos, expected, maxfr, swap);
}
if (res != expected) {
if ((errno != EAGAIN) && (errno != EINTR)) {
diff --git a/channels/iax2-parser.c b/channels/iax2-parser.c
index 3472b5008..f83e7413f 100755
--- a/channels/iax2-parser.c
+++ b/channels/iax2-parser.c
@@ -855,8 +855,15 @@ void iax_frame_wrap(struct iax_frame *fr, struct ast_frame *f)
fr->af.delivery.tv_sec = 0;
fr->af.delivery.tv_usec = 0;
fr->af.data = fr->afdata;
- if (fr->af.datalen)
+ if (fr->af.datalen) {
+#if __BYTE_ORDER == __LITTLE_ENDIAN
+ /* We need to byte-swap slinear samples from network byte order */
+ if (fr->af.subclass == AST_FORMAT_SLINEAR) {
+ ast_memcpy_byteswap(fr->af.data, f->data, fr->af.samples);
+ } else
+#endif
memcpy(fr->af.data, f->data, fr->af.datalen);
+ }
}
struct iax_frame *iax_frame_new(int direction, int datalen)
diff --git a/frame.c b/frame.c
index 673e7b46d..6ae728e9c 100755
--- a/frame.c
+++ b/frame.c
@@ -83,7 +83,7 @@ void ast_smoother_set_flags(struct ast_smoother *s, int flags)
s->flags = flags;
}
-int ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f)
+int __ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f, int swap)
{
if (f->frametype != AST_FRAME_VOICE) {
ast_log(LOG_WARNING, "Huh? Can't smooth a non-voice frame!\n");
@@ -129,7 +129,10 @@ int ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f)
return 0;
}
}
- memcpy(s->data + s->len, f->data, f->datalen);
+ if (swap)
+ ast_swapcopy_samples(s->data+s->len, f->data, f->samples);
+ else
+ memcpy(s->data + s->len, f->data, f->datalen);
/* If either side is empty, reset the delivery time */
if (!s->len || (!f->delivery.tv_sec && !f->delivery.tv_usec) ||
(!s->delivery.tv_sec && !s->delivery.tv_usec))
@@ -399,6 +402,16 @@ int ast_fr_fdhangup(int fd)
return ast_fr_fdwrite(fd, &hangup);
}
+void ast_swapcopy_samples(void *dst, void *src, int samples)
+{
+ int i;
+ unsigned short *dst_s = dst;
+ unsigned short *src_s = src;
+
+ for (i=0; i<samples; i++)
+ dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8);
+}
+
static struct ast_format_list AST_FORMAT_LIST[] = {
{ 1, AST_FORMAT_G723_1 , "g723" , "G.723.1"},
{ 1, AST_FORMAT_GSM, "gsm" , "GSM"},
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index 2edf248a3..88eb6d5c7 100755
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -301,6 +301,19 @@ int ast_fr_fdwrite(int fd, struct ast_frame *frame);
*/
int ast_fr_fdhangup(int fd);
+void ast_swapcopy_samples(void *dst, void *src, int samples);
+
+/* Helpers for byteswapping native samples to/from
+ little-endian and big-endian. */
+#if __BYTE_ORDER == __LITTLE_ENDIAN
+#define ast_frame_byteswap_le(fr) do { ; } while(0)
+#define ast_frame_byteswap_be(fr) do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data, __f->data, __f->samples); } while(0)
+#else
+#define ast_frame_byteswap_le(fr) do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data, __f->data, __f->samples); } while(0)
+#define ast_frame_byteswap_be(fr) do { ; } while(0)
+#endif
+
+
/*! Get the name of a format */
/*!
* \param format id of format
@@ -347,8 +360,16 @@ extern void ast_smoother_set_flags(struct ast_smoother *smoother, int flags);
extern int ast_smoother_get_flags(struct ast_smoother *smoother);
extern void ast_smoother_free(struct ast_smoother *s);
extern void ast_smoother_reset(struct ast_smoother *s, int bytes);
-extern int ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f);
+extern int __ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f, int swap);
extern struct ast_frame *ast_smoother_read(struct ast_smoother *s);
+#define ast_smoother_feed(s,f) do { __ast_smoother_feed(s, f, 0); } while(0)
+#if __BYTE_ORDER == __LITTLE_ENDIAN
+#define ast_smoother_feed_be(s,f) do { __ast_smoother_feed(s, f, 1); } while(0)
+#define ast_smoother_feed_le(s,f) do { __ast_smoother_feed(s, f, 0); } while(0)
+#else
+#define ast_smoother_feed_be(s,f) do { __ast_smoother_feed(s, f, 0); } while(0)
+#define ast_smoother_feed_le(s,f) do { __ast_smoother_feed(s, f, 1); } while(0)
+#endif
extern void ast_frame_dump(char *name, struct ast_frame *f, char *prefix);
diff --git a/rtp.c b/rtp.c
index cfcfc1c3d..22840eb3c 100755
--- a/rtp.c
+++ b/rtp.c
@@ -596,6 +596,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
break;
case AST_FORMAT_SLINEAR:
rtp->f.samples = rtp->f.datalen / 2;
+ ast_frame_byteswap_be(&rtp->f);
break;
case AST_FORMAT_GSM:
rtp->f.samples = 160 * (rtp->f.datalen / 33);
@@ -1320,6 +1321,19 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
switch(subclass) {
+ case AST_FORMAT_SLINEAR:
+ if (!rtp->smoother) {
+ rtp->smoother = ast_smoother_new(320);
+ }
+ if (!rtp->smoother) {
+ ast_log(LOG_WARNING, "Unable to create smoother :(\n");
+ return -1;
+ }
+ ast_smoother_feed_be(rtp->smoother, _f);
+
+ while((f = ast_smoother_read(rtp->smoother)))
+ ast_rtp_raw_write(rtp, f, codec);
+ break;
case AST_FORMAT_ULAW:
case AST_FORMAT_ALAW:
if (!rtp->smoother) {