[svn-commits] oej: branch oej/rana-dtmf-duration-1.8 r364162 - /team/oej/rana-dtmf-duration...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Apr 27 04:32:34 CDT 2012
Author: oej
Date: Fri Apr 27 04:32:29 2012
New Revision: 364162
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=364162
Log:
Now we're up to the standard of the 1.6.0 version of the patch.
All that is required is to activate the DTMF queue
Modified:
team/oej/rana-dtmf-duration-1.8/res/res_rtp_asterisk.c
Modified: team/oej/rana-dtmf-duration-1.8/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/rana-dtmf-duration-1.8/res/res_rtp_asterisk.c?view=diff&rev=364162&r1=364161&r2=364162
==============================================================================
--- team/oej/rana-dtmf-duration-1.8/res/res_rtp_asterisk.c (original)
+++ team/oej/rana-dtmf-duration-1.8/res/res_rtp_asterisk.c Fri Apr 27 04:32:29 2012
@@ -161,21 +161,25 @@
unsigned int dtmf_timeout; /*!< When this timestamp is reached we consider END frame lost and forcibly abort digit */
unsigned int dtmfsamples;
enum ast_rtp_dtmf_mode dtmfmode;/*!< The current DTMF mode of the RTP stream */
+
/* DTMF Transmission Variables */
unsigned int lastdigitts;
enum dtmf_send_states sending_dtmf; /*!< - are we sending dtmf */
- char sending_digit; /*!< boolean - are we sending digits */
char send_digit; /*!< digit we are sending */
+ char send_dtmf_frame; /*!< Number of samples in a frame with the current packetization */
+ AST_LIST_HEAD_NOLOCK(, ast_frame) dtmfqueue; /*!< Queue for DTMF that we receive while occupied with transmitting an outbound DTMF */
+ struct timeval dtmfmute;
+
int send_payload;
int send_duration;
+ int send_endflag:1; /*!< We have received END marker but are in waiting mode */
unsigned int received_duration; /*!< Received duration (according to control frames) */
- int send_endflag; /*!< We have received END marker but are in waiting mode */
+
unsigned int flags;
struct timeval rxcore;
struct timeval txcore;
double drxcore; /*!< The double representation of the first received packet */
struct timeval lastrx; /*!< timeval when we last received a packet */
- struct timeval dtmfmute;
struct ast_smoother *smoother;
int *ioid;
unsigned short seqno; /*!< Sequence number, RFC 3550, page 13. */
@@ -595,6 +599,7 @@
static int ast_rtp_destroy(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ struct ast_frame *f;
/* Destroy the smoother that was smoothing out audio if present */
if (rtp->smoother) {
@@ -623,6 +628,11 @@
ast_free(rtp->red);
}
+ /* Empty the DTMF queue */
+ while ((f = AST_LIST_REMOVE_HEAD(&rtp->dtmfqueue, frame_list))) {
+ ast_frfree(f);
+ }
+
/* Finally destroy ourselves */
ast_free(rtp);
@@ -655,6 +665,16 @@
/* If we have no remote address information bail out now */
if (ast_sockaddr_isnull(&remote_address)) {
return -1;
+ }
+
+ /* If we're sending DTMF already, we will ignore this but raise sending_dtmf with one
+ to mark that we're busy and can't be disturbed. When we receive an END packet, we will
+ act on that - either start playing with some delay or stack it up in a dtmfqueue.
+ */
+ if (rtp->sending_dtmf) {
+ ast_debug(3, "Received DTMF begin while we're playing out DTMF. Ignoring \n");
+ rtp->sending_dtmf = DTMF_SEND_INPROGRESS_WITH_QUEUE; /* Tell the world that there's an ignored DTMF */
+/* OEJ Fix ??? */
}
/* Convert given digit into what we want to transmit */
@@ -700,14 +720,17 @@
payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
}
rtp->seqno++;
- rtp->send_duration += 160;
+ rtp->send_duration += 160; /* OEJ - check what's going on here. */
+
rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
}
- /* Record that we are in the process of sending a digit and information needed to continue doing so */
- rtp->sending_digit = 1;
+ /* Since we received a begin, we can safely store the digit and disable any compensation */
+ rtp->sending_dtmf = DTMF_SEND_INIT;
rtp->send_digit = digit;
rtp->send_payload = payload;
+
+ ast_log(LOG_DEBUG, "DEBUG DTMF BEGIN - Digit %d send-digit %d\n", digit, rtp->send_digit);
return 0;
}
@@ -744,6 +767,19 @@
return 0;
}
+/*! \brief Send continuation frame for DTMF
+
+This is called when we get a frame in ast_rtp_read. To keep the timing, because there may be delays through Asterisk
+feature handling and other code, we need to clock the outbound DTMF with the frame size we have on the stream.
+We should not cut short and send a begin then in the next packet an END with a duration that exceeds the
+framesize (in most cases for audio 20 ms) and number of frames. That will seriously cause issues in gateways
+or phones down the path.
+
+An effect of this is that we may get a new DTMF frame while we're transmitting the previous one. For this case,
+we have implemented an DTMF queue that will queue up the dtmf and play out. The alternative would be to skip
+these, which is no good, or cut them short and cause issues with timing for other devices, while we solve our
+own situation. That's generally considered bad behaviour amongst SIP devices.
+*/
static int ast_rtp_dtmf_cont(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
@@ -759,6 +795,20 @@
return -1;
}
+ ast_debug(4, "---- Send duration %d Received duration %d Endflag %d Send-digit %d\n", rtp->send_duration, rtp->received_duration, rtp->send_endflag, rtp->send_digit);
+
+ /*! \todo XXX This code assumes 160 samples, which is for 20 ms of 8000 samples
+ we need to calculate this based on the current sample rate and the rtp
+ stream packetization. Please help me figure this out :-)
+ */
+ if (rtp->received_duration == 0 || rtp->send_duration + 160 <= rtp->received_duration) {
+ rtp->send_duration += 160;
+ } else if (rtp->send_endflag) {
+ ast_log(LOG_DEBUG, "---- Send duration %d Received duration %d - sending END packet\n", rtp->send_duration, rtp->received_duration);
+ /* We are done, ready to send end flag */
+ rtp->send_endflag = 0;
+ return ast_rtp_dtmf_end_with_duration(instance, 0, rtp->received_duration);
+ }
/* Actually create the packet we will be sending */
rtpheader[0] = htonl((2 << 30) | (1 << 23) | (rtp->send_payload << 16) | (rtp->seqno));
rtpheader[1] = htonl(rtp->lastdigitts);
@@ -782,7 +832,7 @@
/* And now we increment some values for the next time we swing by */
rtp->seqno++;
- rtp->send_duration += 160;
+ rtp->send_duration += 160; /* Again assuming 20 ms packetization and 8000 samples */
return 0;
}
@@ -795,12 +845,25 @@
char data[256];
unsigned int *rtpheader = (unsigned int*)data;
unsigned int measured_samples;
+ unsigned int dursamples;
ast_rtp_instance_get_remote_address(instance, &remote_address);
/* Make sure we know where the remote side is so we can send them the packet we construct */
if (ast_sockaddr_isnull(&remote_address)) {
return -1;
+ }
+ dursamples = duration * (8000 / 1000); /* How do we get the sample rate for the primary media in this call? */
+
+ ast_debug(1, "---- Send duration %d Received duration %d Duration %d Endflag %d Digit %d Send-digit %d\n", rtp->send_duration, rtp->received_duration, duration, rtp->send_endflag, digit, rtp ->send_digit);
+
+ if (!rtp->send_endflag && rtp->send_duration + 160 < rtp->received_duration) {
+ /* We still have to send DTMF continuation, because otherwise we will end prematurely. Set end flag to indicate
+ that we will have to end ourselves when we're done with the actual duration
+ */
+ ast_log(LOG_DEBUG, "---- Send duration %d Received duration %d - Avoiding sending END packet\n", rtp->send_duration, rtp->received_duration);
+ rtp->send_endflag = 1;
+ return ast_rtp_dtmf_cont(instance);
}
/* Convert the given digit to the one we are going to send */
@@ -851,7 +914,7 @@
/* Oh and we can't forget to turn off the stuff that says we are sending DTMF */
rtp->lastts += rtp->send_duration;
- rtp->sending_digit = 0;
+ rtp->sending_dtmf = DTMF_NOT_SENDING;
rtp->send_digit = 0;
return 0;
@@ -1201,7 +1264,7 @@
frame->samples /= 2;
}
- if (rtp->sending_digit) {
+ if (rtp->sending_dtmf) {
return 0;
}
@@ -2171,7 +2234,7 @@
}
/* If we are currently sending DTMF to the remote party send a continuation packet */
- if (rtp->sending_digit) {
+ if (rtp->sending_dtmf) {
ast_rtp_dtmf_cont(instance);
}
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