add a fixed jitterbuffer to rtp (settable from a channel variable) (libSTFU)

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@5090 d0543943-73ff-0310-b7d9-9358b9ac24b2

Makefile.am

@@ -57,6 +57,7 @@ src/switch_ivr.c \
 src/switch_stun.c\
 src/switch_log.c\
 src/switch_xml.c\
+libs/stfu/stfu.c\
 libs/libteletone/src/libteletone_detect.c\
 libs/libteletone/src/libteletone_generate.c
 
@@ -95,6 +96,7 @@ libs/libteletone/src/libteletone.h
 
 CORE_CFLAGS     = `$(switch_srcdir)/libs/apr/apr-1-config --cflags --cppflags --includes`
 CORE_CFLAGS    += `$(switch_srcdir)/libs/apr-util/apu-1-config --includes`
+CORE_CFLAGS    += -I$(switch_srcdir)/libs/stfu
 CORE_CFLAGS    += -I$(switch_srcdir)/libs/sqlite
 CORE_CFLAGS    += -I$(switch_srcdir)/libs/pcre
 CORE_CFLAGS    += -I$(switch_srcdir)/libs/srtp/include

conf/default_context.xml

@@ -53,6 +53,7 @@
       <!-- Request a certain tone/file to be played while you wait for the call to be answered-->
       <action application="set" data="ringback=${us-ring}"/>
       <!--<action application="set" data="ringback=/home/ring.wav"/>-->
+      <!--<action application="set" data="jitterbuffer_msec=400"/>-->
       <action application="bridge" data="sofia/$${sip_profile}/1234@conference.freeswitch.org"/>
     </condition>
   </extension>

libs/stfu/stfu.c

@@ -163,7 +163,6 @@ stfu_status_t stfu_n_add_data(stfu_instance_t *i, uint32_t ts, void *data, size_
 		cplen = sizeof(frame->data);
 	}
 
-
 	memcpy(frame->data, data, cplen);
 	frame->ts = ts;
 	frame->dlen = cplen;
@@ -185,6 +184,8 @@ stfu_frame_t *stfu_n_read_a_frame(stfu_instance_t *i)
 
 	if (i->running) {
 		should_have = i->last_ts + i->interval;
+	} else {
+		should_have = i->out_queue->array[0].ts;
 	}
 
 	for(index = 0; index < i->out_queue->array_len; index++) {
@@ -220,11 +221,12 @@ stfu_frame_t *stfu_n_read_a_frame(stfu_instance_t *i)
 
 			i->miss_count++;
 
-			if (i->miss_count > 10) {
+			if (i->miss_count > 10 || i->in_queue->array_len == i->in_queue->array_size ) {
 				i->out_queue->wr_len = i->out_queue->array_len;
 				i->last_ts = should_have = frame->ts;
 				return NULL;
 			}
+
 			i->last_ts = should_have;
 			rframe = &i->out_queue->int_frame;
 			rframe->dlen = i->out_queue->array[i->out_queue->last_index].dlen;
@@ -248,6 +250,7 @@ stfu_frame_t *stfu_n_read_a_frame(stfu_instance_t *i)
 		i->last_ts = rframe->ts;
 		rframe->was_read = 1;
 		i->running = 1;
+		i->miss_count = 0;
 	}
 
 	return rframe;

src/include/switch_rtp.h

@@ -165,6 +165,14 @@ SWITCH_DECLARE(void) switch_rtp_destroy(switch_rtp_t **rtp_session);
 */
 SWITCH_DECLARE(switch_status_t) switch_rtp_activate_ice(switch_rtp_t *rtp_session, char *login, char *rlogin);
 
+/*! 
+  \brief Acvite a jitter buffer on an RTP session
+  \param rtp_session the rtp session
+  \param queue_frames the number of frames to delay
+  \return SWITCH_STATUS_SUCCESS
+*/
+SWITCH_DECLARE(switch_status_t) switch_rtp_activate_jitter_buffer(switch_rtp_t *rtp_session, uint32_t queue_frames);
+
 /*!
   \brief Set an RTP Flag
   \param rtp_session the RTP session

src/include/switch_types.h

@@ -592,13 +592,15 @@ typedef enum {
 SFF_CNG        = (1 <<  0) - Frame represents comfort noise
 SFF_RAW_RTP    = (1 <<  1) - Frame has raw rtp accessible
 SFF_RTP_HEADER = (1 << 2)  - Get the rtp header from the frame header
+SFF_PLC        = (1 << 3)  - Frame has generated PLC data
 </pre>
  */
 typedef enum {
 	SFF_NONE = 0,
 	SFF_CNG = (1 << 0),
 	SFF_RAW_RTP = (1 << 1),
-	SFF_RTP_HEADER = (1 << 2)
+	SFF_RTP_HEADER = (1 << 2),
+	SFF_PLC = (1 << 3)
 } switch_frame_flag_t;
 
 

src/mod/endpoints/mod_sofia/sofia_glue.c

@@ -949,6 +949,23 @@ switch_status_t sofia_glue_activate_rtp(private_object_t *tech_pvt)
 							  switch_channel_get_name(switch_core_session_get_channel(tech_pvt->session)), vad_in ? "in" : "", vad_out ? "out" : "");
 		}
 
+		if ((val = switch_channel_get_variable(channel, "jitterbuffer_msec"))) {
+			int len = atoi(val);
+			
+			if (len < 100 || len > 1000) {
+				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invalid Jitterbuffer spec [%d] myst be between 100 and 1000\n", len);
+			} else {
+				int qlen;
+
+				qlen = len / (tech_pvt->read_codec.implementation->microseconds_per_frame / 1000);
+				
+				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Setting Jitterbuffer to %dms (%d frames)\n", len, qlen);
+				switch_rtp_activate_jitter_buffer(tech_pvt->rtp_session, qlen);
+			}
+		}
+
+
+
 		if (tech_pvt->te) {
 			switch_rtp_set_telephony_event(tech_pvt->rtp_session, tech_pvt->te);
 		}

src/switch_rtp.c

@@ -42,7 +42,7 @@
 #undef inline
 #include <datatypes.h>
 #include <srtp.h>
-
+#include "stfu.h"
 #define MAX_KEY_LEN      64
 #define rtp_header_len 12
 #define RTP_START_PORT 16384
@@ -164,6 +164,7 @@ struct switch_rtp {
 	uint8_t ready;
 	uint8_t cn;
 	switch_time_t last_time;
+	stfu_instance_t *jb;
 };
 
 static int global_init = 0;
@@ -586,6 +587,13 @@ SWITCH_DECLARE(void) switch_rtp_set_cng_pt(switch_rtp_t *rtp_session, switch_pay
 
 }
 
+SWITCH_DECLARE(switch_status_t) switch_rtp_activate_jitter_buffer(switch_rtp_t *rtp_session, uint32_t queue_frames)
+{
+	rtp_session->jb = stfu_n_init(queue_frames);	
+
+	return SWITCH_STATUS_SUCCESS;
+}
+
 SWITCH_DECLARE(switch_status_t) switch_rtp_activate_ice(switch_rtp_t *rtp_session, char *login, char *rlogin)
 {
 	char ice_user[80];
@@ -632,6 +640,10 @@ SWITCH_DECLARE(void) switch_rtp_destroy(switch_rtp_t **rtp_session)
 
 	switch_mutex_lock((*rtp_session)->flag_mutex);
 
+	if ((*rtp_session)->jb) {
+		stfu_n_destroy(&(*rtp_session)->jb);
+	}
+
 	if ((*rtp_session)->dtmf_data.dtmf_buffer) {
 		switch_buffer_destroy(&(*rtp_session)->dtmf_data.dtmf_buffer);
 	}
@@ -821,6 +833,24 @@ static int rtp_common_read(switch_rtp_t *rtp_session, switch_payload_t *payload_
 			return -1;
 		}
 
+		if (rtp_session->jb && bytes) {
+			stfu_frame_t *frame;
+			
+			stfu_n_eat(rtp_session->jb, ntohl(rtp_session->recv_msg.header.ts), rtp_session->recv_msg.body, bytes - rtp_header_len);
+			if ((frame = stfu_n_read_a_frame(rtp_session->jb))) {
+				memcpy(rtp_session->recv_msg.body, frame->data, frame->dlen);
+				if (frame->plc) {
+					*flags |= SFF_PLC;
+				}
+				bytes = frame->dlen + rtp_header_len;
+				rtp_session->recv_msg.header.ts = htonl(frame->ts);
+			} else {
+				bytes = 0;
+				continue;
+			}
+			
+		}
+
 		if (!bytes && switch_test_flag(rtp_session, SWITCH_RTP_FLAG_BREAK)) {
 			switch_clear_flag_locked(rtp_session, SWITCH_RTP_FLAG_BREAK);
 
@@ -907,6 +937,7 @@ static int rtp_common_read(switch_rtp_t *rtp_session, switch_payload_t *payload_
 			bytes = sbytes;
 		}
 
+
 		if (status == SWITCH_STATUS_BREAK || bytes == 0) {
 			if (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_DATAWAIT)) {
 				if (rtp_session->ms_per_packet) {
@@ -997,6 +1028,7 @@ static int rtp_common_read(switch_rtp_t *rtp_session, switch_payload_t *payload_
 			continue;
 		}
 	
+
 		break;
 	}