From c936fc827c95f2a1abb6dd00a477cbe37bf56167 Mon Sep 17 00:00:00 2001
From: Mark Spencer <markster@digium.com>
Date: Sat, 14 May 2005 23:57:44 +0000
Subject: [PATCH] Add remainder of rtp fixes, iax2 patch (bug #3961)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5654 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_iax2.c | 48 +---------
frame.c | 192 +++++++++++++++++++++++++++++++++++++++
include/asterisk/frame.h | 3 +
rtp.c | 121 +-----------------------
4 files changed, 202 insertions(+), 162 deletions(-)
diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index 29cf3cd05b..61e6c3a45c 100755
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -859,48 +859,6 @@ static struct chan_iax2_pvt *new_iax(struct sockaddr_in *sin, int lockpeer, cons
return tmp;
}
-static int get_samples(struct ast_frame *f)
-{
- int samples=0;
- switch(f->subclass) {
- case AST_FORMAT_SPEEX:
- samples = 160; /* XXX Not necessarily true XXX */
- break;
- case AST_FORMAT_G723_1:
- samples = 240 /* XXX Not necessarily true XXX */;
- break;
- case AST_FORMAT_ILBC:
- samples = 240 * (f->datalen / 50);
- break;
- case AST_FORMAT_GSM:
- samples = 160 * (f->datalen / 33);
- break;
- case AST_FORMAT_G729A:
- samples = 160 * (f->datalen / 20);
- break;
- case AST_FORMAT_SLINEAR:
- samples = f->datalen / 2;
- break;
- case AST_FORMAT_LPC10:
- samples = 22 * 8;
- samples += (((char *)(f->data))[7] & 0x1) * 8;
- break;
- case AST_FORMAT_ULAW:
- samples = f->datalen;
- break;
- case AST_FORMAT_ALAW:
- samples = f->datalen;
- break;
- case AST_FORMAT_ADPCM:
- case AST_FORMAT_G726:
- samples = f->datalen *2;
- break;
- default:
- ast_log(LOG_WARNING, "Don't know how to calculate samples on %d packets\n", f->subclass);
- }
- return samples;
-}
-
static struct iax_frame *iaxfrdup2(struct iax_frame *fr)
{
/* Malloc() a copy of a frame */
@@ -2347,7 +2305,7 @@ static int schedule_delivery(struct iax_frame *fr, int reallydeliver, int update
if(fr->af.frametype == AST_FRAME_VOICE) {
type = JB_TYPE_VOICE;
- len = get_samples(&fr->af)/8;
+ len = ast_codec_get_samples(&fr->af) / 8;
} else if(fr->af.frametype == AST_FRAME_CNG) {
type = JB_TYPE_SILENCE;
}
@@ -6240,7 +6198,7 @@ static int socket_read(int *id, int fd, short events, void *cbdata)
f.mallocd = 0;
f.offset = 0;
if (f.datalen && (f.frametype == AST_FRAME_VOICE))
- f.samples = get_samples(&f);
+ f.samples = ast_codec_get_samples(&f);
else
f.samples = 0;
fr.outoforder = 0;
@@ -7355,7 +7313,7 @@ retryowner2:
f.mallocd = 0;
f.offset = 0;
if (f.datalen && (f.frametype == AST_FRAME_VOICE)) {
- f.samples = get_samples(&f);
+ f.samples = ast_codec_get_samples(&f);
/* We need to byteswap incoming slinear samples from network byte order */
if (f.subclass == AST_FORMAT_SLINEAR)
ast_frame_byteswap_be(&f);
diff --git a/frame.c b/frame.c
index 41932f8805..2e68ec4ffb 100755
--- a/frame.c
+++ b/frame.c
@@ -35,6 +35,12 @@ AST_MUTEX_DEFINE_STATIC(framelock);
#define SMOOTHER_SIZE 8000
+#define TYPE_HIGH 0x0
+#define TYPE_LOW 0x1
+#define TYPE_SILENCE 0x2
+#define TYPE_DONTSEND 0x3
+#define TYPE_MASK 0x3
+
struct ast_format_list {
int visible; /* Can we see this entry */
int bits; /* bitmask value */
@@ -1008,4 +1014,190 @@ void ast_parse_allow_disallow(struct ast_codec_pref *pref, int *mask, char *list
}
}
+static int g723_len(unsigned char buf)
+{
+ switch(buf & TYPE_MASK) {
+ case TYPE_DONTSEND:
+ return 0;
+ break;
+ case TYPE_SILENCE:
+ return 4;
+ break;
+ case TYPE_HIGH:
+ return 24;
+ break;
+ case TYPE_LOW:
+ return 20;
+ break;
+ default:
+ ast_log(LOG_WARNING, "Badly encoded frame (%d)\n", buf & TYPE_MASK);
+ }
+ return -1;
+}
+
+static int g723_samples(unsigned char *buf, int maxlen)
+{
+ int pos = 0;
+ int samples = 0;
+ int res;
+ while(pos < maxlen) {
+ res = g723_len(buf[pos]);
+ if (res <= 0)
+ break;
+ samples += 240;
+ pos += res;
+ }
+ return samples;
+}
+
+static unsigned char get_n_bits_at(unsigned char *data, int n, int bit)
+{
+ int byte = bit / 8; /* byte containing first bit */
+ int rem = 8 - (bit % 8); /* remaining bits in first byte */
+ unsigned char ret = 0;
+
+ if (n <= 0 || n > 8)
+ return 0;
+
+ if (rem < n) {
+ ret = (data[byte] << (n - rem));
+ ret |= (data[byte + 1] >> (8 - n + rem));
+ } else {
+ ret = (data[byte] >> (rem - n));
+ }
+
+ return (ret & (0xff >> (8 - n)));
+}
+
+static int speex_get_wb_sz_at(unsigned char *data, int len, int bit)
+{
+ static int SpeexWBSubModeSz[] = {
+ 0, 36, 112, 192,
+ 352, 0, 0, 0 };
+ int off = bit;
+ unsigned char c;
+
+ /* skip up to two wideband frames */
+ if (((len * 8 - off) >= 5) &&
+ get_n_bits_at(data, 1, off)) {
+ c = get_n_bits_at(data, 3, off + 1);
+ off += SpeexWBSubModeSz[c];
+
+ if (((len * 8 - off) >= 5) &&
+ get_n_bits_at(data, 1, off)) {
+ c = get_n_bits_at(data, 3, off + 1);
+ off += SpeexWBSubModeSz[c];
+
+ if (((len * 8 - off) >= 5) &&
+ get_n_bits_at(data, 1, off)) {
+ ast_log(LOG_WARNING, "Encountered corrupt speex frame; too many wideband frames in a row.\n");
+ return -1;
+ }
+ }
+
+ }
+ return off - bit;
+}
+
+static int speex_samples(unsigned char *data, int len)
+{
+ static int SpeexSubModeSz[] = {
+ 0, 43, 119, 160,
+ 220, 300, 364, 492,
+ 79, 0, 0, 0,
+ 0, 0, 0, 0 };
+ static int SpeexInBandSz[] = {
+ 1, 1, 4, 4,
+ 4, 4, 4, 4,
+ 8, 8, 16, 16,
+ 32, 32, 64, 64 };
+ int bit = 0;
+ int cnt = 0;
+ int off = 0;
+ unsigned char c;
+
+ while ((len * 8 - bit) >= 5) {
+ /* skip wideband frames */
+ off = speex_get_wb_sz_at(data, len, bit);
+ if (off < 0) {
+ ast_log(LOG_WARNING, "Had error while reading wideband frames for speex samples\n");
+ break;
+ }
+ bit += off;
+
+ if ((len * 8 - bit) < 5) {
+ ast_log(LOG_WARNING, "Not enough bits remaining after wide band for speex samples.\n");
+ break;
+ }
+
+ /* get control bits */
+ c = get_n_bits_at(data, 5, bit);
+ bit += 5;
+
+ if (c == 15) {
+ /* terminator */
+ break;
+ } else if (c == 14) {
+ /* in-band signal; next 4 bits contain signal id */
+ c = get_n_bits_at(data, 4, bit);
+ bit += 4;
+ bit += SpeexInBandSz[c];
+ } else if (c == 13) {
+ /* user in-band; next 5 bits contain msg len */
+ c = get_n_bits_at(data, 5, bit);
+ bit += 5;
+ bit += c * 8;
+ } else if (c > 8) {
+ /* unknown */
+ break;
+ } else {
+ /* skip number bits for submode (less the 5 control bits) */
+ bit += SpeexSubModeSz[c] - 5;
+ cnt += 160; /* new frame */
+ }
+ }
+ return cnt;
+}
+
+
+int ast_codec_get_samples(struct ast_frame *f)
+{
+ int samples=0;
+ switch(f->subclass) {
+ case AST_FORMAT_SPEEX:
+ samples = speex_samples(f->data, f->datalen);
+ break;
+ case AST_FORMAT_G723_1:
+ samples = g723_samples(f->data, f->datalen);
+ break;
+ case AST_FORMAT_ILBC:
+ samples = 240 * (f->datalen / 50);
+ break;
+ case AST_FORMAT_GSM:
+ samples = 160 * (f->datalen / 33);
+ break;
+ case AST_FORMAT_G729A:
+ samples = f->datalen * 8;
+ break;
+ case AST_FORMAT_SLINEAR:
+ samples = f->datalen / 2;
+ break;
+ case AST_FORMAT_LPC10:
+ /* assumes that the RTP packet contains one LPC10 frame */
+ samples = 22 * 8;
+ samples += (((char *)(f->data))[7] & 0x1) * 8;
+ break;
+ case AST_FORMAT_ULAW:
+ case AST_FORMAT_ALAW:
+ samples = f->datalen;
+ break;
+ case AST_FORMAT_ADPCM:
+ case AST_FORMAT_G726:
+ samples = f->datalen * 2;
+ break;
+ default:
+ ast_log(LOG_WARNING, "Unable to calculate samples for format %s\n", ast_getformatname(f->subclass));
+ }
+ return samples;
+}
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index 695333514b..d8a715a8f4 100755
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -399,6 +399,9 @@ extern int ast_codec_pref_string(struct ast_codec_pref *pref, char *buf, size_t
/* Shift a codec preference list up or down 65 bytes so that it becomes an ASCII string */
extern void ast_codec_pref_convert(struct ast_codec_pref *pref, char *buf, size_t size, int right);
+/* Returns the number of samples contained in the frame */
+extern int ast_codec_get_samples(struct ast_frame *f);
+
/* Gets duration in ms of interpolation frame for a format */
static inline int ast_codec_interp_len(int format)
{
diff --git a/rtp.c b/rtp.c
index 9fde8b857b..b0bf75e425 100755
--- a/rtp.c
+++ b/rtp.c
@@ -43,12 +43,6 @@
#define RTP_MTU 1200
-#define TYPE_HIGH 0x0
-#define TYPE_LOW 0x1
-#define TYPE_SILENCE 0x2
-#define TYPE_DONTSEND 0x3
-#define TYPE_MASK 0x3
-
static int dtmftimeout = 3000; /* 3000 samples */
static int rtpstart = 0;
@@ -128,42 +122,6 @@ int ast_rtcp_fd(struct ast_rtp *rtp)
return -1;
}
-static int g723_len(unsigned char buf)
-{
- switch(buf & TYPE_MASK) {
- case TYPE_DONTSEND:
- return 0;
- break;
- case TYPE_SILENCE:
- return 4;
- break;
- case TYPE_HIGH:
- return 24;
- break;
- case TYPE_LOW:
- return 20;
- break;
- default:
- ast_log(LOG_WARNING, "Badly encoded frame (%d)\n", buf & TYPE_MASK);
- }
- return -1;
-}
-
-static int g723_samples(unsigned char *buf, int maxlen)
-{
- int pos = 0;
- int samples = 0;
- int res;
- while(pos < maxlen) {
- res = g723_len(buf[pos]);
- if (res <= 0)
- break;
- samples += 240;
- pos += res;
- }
- return samples;
-}
-
void ast_rtp_set_data(struct ast_rtp *rtp, void *data)
{
rtp->data = data;
@@ -594,43 +552,9 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) {
- switch(rtp->f.subclass) {
- case AST_FORMAT_ULAW:
- case AST_FORMAT_ALAW:
- rtp->f.samples = rtp->f.datalen;
- break;
- case AST_FORMAT_SLINEAR:
- rtp->f.samples = rtp->f.datalen / 2;
+ rtp->f.samples = ast_codec_get_samples(&rtp->f);
+ if (rtp->f.subclass == AST_FORMAT_SLINEAR)
ast_frame_byteswap_be(&rtp->f);
- break;
- case AST_FORMAT_GSM:
- rtp->f.samples = 160 * (rtp->f.datalen / 33);
- break;
- case AST_FORMAT_ILBC:
- rtp->f.samples = 240 * (rtp->f.datalen / 50);
- break;
- case AST_FORMAT_ADPCM:
- case AST_FORMAT_G726:
- rtp->f.samples = rtp->f.datalen * 2;
- break;
- case AST_FORMAT_G729A:
- rtp->f.samples = rtp->f.datalen * 8;
- break;
- case AST_FORMAT_G723_1:
- rtp->f.samples = g723_samples(rtp->f.data, rtp->f.datalen);
- break;
- case AST_FORMAT_SPEEX:
- /* assumes that the RTP packet contained one Speex frame */
- rtp->f.samples = 160;
- break;
- case AST_FORMAT_LPC10:
- rtp->f.samples = 22 * 8;
- rtp->f.samples += (((char *)(rtp->f.data))[7] & 0x1) * 8;
- break;
- default:
- ast_log(LOG_NOTICE, "Unable to calculate samples for format %s\n", ast_getformatname(rtp->f.subclass));
- break;
- }
calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
} else {
/* Video -- samples is # of samples vs. 90000 */
@@ -1233,45 +1157,8 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
ms = calc_txstamp(rtp, &f->delivery);
/* Default prediction */
if (f->subclass < AST_FORMAT_MAX_AUDIO) {
- pred = rtp->lastts + ms * 8;
-
- switch(f->subclass) {
- case AST_FORMAT_ULAW:
- case AST_FORMAT_ALAW:
- /* If we're within +/- 20ms from when where we
- predict we should be, use that */
- pred = rtp->lastts + f->datalen;
- break;
- case AST_FORMAT_ADPCM:
- case AST_FORMAT_G726:
- /* If we're within +/- 20ms from when where we
- predict we should be, use that */
- pred = rtp->lastts + f->datalen * 2;
- break;
- case AST_FORMAT_G729A:
- pred = rtp->lastts + f->datalen * 8;
- break;
- case AST_FORMAT_GSM:
- pred = rtp->lastts + (f->datalen * 160 / 33);
- break;
- case AST_FORMAT_ILBC:
- pred = rtp->lastts + (f->datalen * 240 / 50);
- break;
- case AST_FORMAT_G723_1:
- pred = rtp->lastts + g723_samples(f->data, f->datalen);
- break;
- case AST_FORMAT_SPEEX:
- pred = rtp->lastts + 160;
- /* assumes that the RTP packet contains one Speex frame */
- break;
- case AST_FORMAT_LPC10:
- /* assumes that the RTP packet contains one LPC10 frame */
- pred = rtp->lastts + 22 * 8;
- pred += (((char *)(f->data))[7] & 0x1) * 8;
- break;
- default:
- ast_log(LOG_WARNING, "Not sure about timestamp format for codec format %s\n", ast_getformatname(f->subclass));
- }
+ pred = rtp->lastts + ast_codec_get_samples(f);
+
/* Re-calculate last TS */
rtp->lastts = rtp->lastts + ms * 8;
if (!f->delivery.tv_sec && !f->delivery.tv_usec) {
--
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