Split reporting capability to separate file in siprtp sample
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@437 74dad513-b988-da41-8d7b-12977e46ad98
diff --git a/pjsip-apps/build/samples.dsp b/pjsip-apps/build/samples.dsp
index 300920f..189f546 100644
--- a/pjsip-apps/build/samples.dsp
+++ b/pjsip-apps/build/samples.dsp
@@ -118,6 +118,10 @@
# End Source File
# Begin Source File
+SOURCE=..\src\samples\siprtp_report.c
+# End Source File
+# Begin Source File
+
SOURCE=..\src\samples\sndinfo.c
# End Source File
# Begin Source File
diff --git a/pjsip-apps/src/samples/siprtp.c b/pjsip-apps/src/samples/siprtp.c
index e22a67e..24bc6dc 100644
--- a/pjsip-apps/src/samples/siprtp.c
+++ b/pjsip-apps/src/samples/siprtp.c
@@ -1360,208 +1360,7 @@
/*****************************************************************************
* USER INTERFACE STUFFS
*/
-
-static const char *good_number(char *buf, pj_int32_t val)
-{
- if (val < 1000) {
- pj_ansi_sprintf(buf, "%d", val);
- } else if (val < 1000000) {
- pj_ansi_sprintf(buf, "%d.%dK",
- val / 1000,
- (val % 1000) / 100);
- } else {
- pj_ansi_sprintf(buf, "%d.%02dM",
- val / 1000000,
- (val % 1000000) / 10000);
- }
-
- return buf;
-}
-
-
-static void print_call(int call_index)
-{
- struct call *call = &app.call[call_index];
- int len;
- pjsip_inv_session *inv = call->inv;
- pjsip_dialog *dlg = inv->dlg;
- struct media_stream *audio = &call->media[0];
- char userinfo[128];
- char duration[80], last_update[80];
- char bps[16], ipbps[16], packets[16], bytes[16], ipbytes[16];
- pj_time_val now;
-
- pj_gettimeofday(&now);
-
- /* Print duration */
- if (inv->state >= PJSIP_INV_STATE_CONFIRMED) {
-
- PJ_TIME_VAL_SUB(now, call->connect_time);
-
- sprintf(duration, " [duration: %02ld:%02ld:%02ld.%03ld]",
- now.sec / 3600,
- (now.sec % 3600) / 60,
- (now.sec % 60),
- now.msec);
-
- } else {
- duration[0] = '\0';
- }
-
-
-
- /* Call number and state */
- printf("Call #%d: %s%s\n", call_index, pjsip_inv_state_name(inv->state),
- duration);
-
-
-
- /* Call identification */
- len = pjsip_hdr_print_on(dlg->remote.info, userinfo, sizeof(userinfo));
- if (len < 1)
- pj_ansi_strcpy(userinfo, "<--uri too long-->");
- else
- userinfo[len] = '\0';
-
- printf(" %s\n", userinfo);
-
-
- if (call->inv == NULL || call->inv->state < PJSIP_INV_STATE_CONFIRMED ||
- call->connect_time.sec == 0)
- {
- return;
- }
-
-
- /* Signaling quality */
- {
- char pdd[64], connectdelay[64];
- pj_time_val t;
-
- if (call->response_time.sec) {
- t = call->response_time;
- PJ_TIME_VAL_SUB(t, call->start_time);
- sprintf(pdd, "got 1st response in %ld ms", PJ_TIME_VAL_MSEC(t));
- } else {
- pdd[0] = '\0';
- }
-
- if (call->connect_time.sec) {
- t = call->connect_time;
- PJ_TIME_VAL_SUB(t, call->start_time);
- sprintf(connectdelay, ", connected after: %ld ms",
- PJ_TIME_VAL_MSEC(t));
- } else {
- connectdelay[0] = '\0';
- }
-
- printf(" Signaling quality: %s%s\n", pdd, connectdelay);
- }
-
-
- printf(" Stream #0: audio %.*s@%dHz, %dms/frame, %sB/s (%sB/s +IP hdr)\n",
- (int)audio->si.fmt.encoding_name.slen,
- audio->si.fmt.encoding_name.ptr,
- audio->clock_rate,
- audio->samples_per_frame * 1000 / audio->clock_rate,
- good_number(bps, audio->bytes_per_frame * audio->clock_rate / audio->samples_per_frame),
- good_number(ipbps, (audio->bytes_per_frame+32) * audio->clock_rate / audio->samples_per_frame));
-
- if (audio->rtcp.stat.rx.update_cnt == 0)
- strcpy(last_update, "never");
- else {
- pj_gettimeofday(&now);
- PJ_TIME_VAL_SUB(now, audio->rtcp.stat.rx.update);
- sprintf(last_update, "%02ldh:%02ldm:%02ld.%03lds ago",
- now.sec / 3600,
- (now.sec % 3600) / 60,
- now.sec % 60,
- now.msec);
- }
-
- printf(" RX stat last update: %s\n"
- " total %s packets %sB received (%sB +IP hdr)%s\n"
- " pkt loss=%d (%3.1f%%), dup=%d (%3.1f%%), reorder=%d (%3.1f%%)%s\n"
- " (msec) min avg max last\n"
- " loss period: %7.3f %7.3f %7.3f %7.3f%s\n"
- " jitter : %7.3f %7.3f %7.3f %7.3f%s\n",
- last_update,
- good_number(packets, audio->rtcp.stat.rx.pkt),
- good_number(bytes, audio->rtcp.stat.rx.bytes),
- good_number(ipbytes, audio->rtcp.stat.rx.bytes + audio->rtcp.stat.rx.pkt * 32),
- "",
- audio->rtcp.stat.rx.loss,
- audio->rtcp.stat.rx.loss * 100.0 / audio->rtcp.stat.rx.pkt,
- audio->rtcp.stat.rx.dup,
- audio->rtcp.stat.rx.dup * 100.0 / audio->rtcp.stat.rx.pkt,
- audio->rtcp.stat.rx.reorder,
- audio->rtcp.stat.rx.reorder * 100.0 / audio->rtcp.stat.rx.pkt,
- "",
- audio->rtcp.stat.rx.loss_period.min / 1000.0,
- audio->rtcp.stat.rx.loss_period.avg / 1000.0,
- audio->rtcp.stat.rx.loss_period.max / 1000.0,
- audio->rtcp.stat.rx.loss_period.last / 1000.0,
- "",
- audio->rtcp.stat.rx.jitter.min / 1000.0,
- audio->rtcp.stat.rx.jitter.avg / 1000.0,
- audio->rtcp.stat.rx.jitter.max / 1000.0,
- audio->rtcp.stat.rx.jitter.last / 1000.0,
- ""
- );
-
-
- if (audio->rtcp.stat.tx.update_cnt == 0)
- strcpy(last_update, "never");
- else {
- pj_gettimeofday(&now);
- PJ_TIME_VAL_SUB(now, audio->rtcp.stat.tx.update);
- sprintf(last_update, "%02ldh:%02ldm:%02ld.%03lds ago",
- now.sec / 3600,
- (now.sec % 3600) / 60,
- now.sec % 60,
- now.msec);
- }
-
- printf(" TX stat last update: %s\n"
- " total %s packets %sB sent (%sB +IP hdr)%s\n"
- " pkt loss=%d (%3.1f%%), dup=%d (%3.1f%%), reorder=%d (%3.1f%%)%s\n"
- " (msec) min avg max last\n"
- " loss period: %7.3f %7.3f %7.3f %7.3f%s\n"
- " jitter : %7.3f %7.3f %7.3f %7.3f%s\n",
- last_update,
- good_number(packets, audio->rtcp.stat.tx.pkt),
- good_number(bytes, audio->rtcp.stat.tx.bytes),
- good_number(ipbytes, audio->rtcp.stat.tx.bytes + audio->rtcp.stat.tx.pkt * 32),
- "",
- audio->rtcp.stat.tx.loss,
- audio->rtcp.stat.tx.loss * 100.0 / audio->rtcp.stat.tx.pkt,
- audio->rtcp.stat.tx.dup,
- audio->rtcp.stat.tx.dup * 100.0 / audio->rtcp.stat.tx.pkt,
- audio->rtcp.stat.tx.reorder,
- audio->rtcp.stat.tx.reorder * 100.0 / audio->rtcp.stat.tx.pkt,
- "",
- audio->rtcp.stat.tx.loss_period.min / 1000.0,
- audio->rtcp.stat.tx.loss_period.avg / 1000.0,
- audio->rtcp.stat.tx.loss_period.max / 1000.0,
- audio->rtcp.stat.tx.loss_period.last / 1000.0,
- "",
- audio->rtcp.stat.tx.jitter.min / 1000.0,
- audio->rtcp.stat.tx.jitter.avg / 1000.0,
- audio->rtcp.stat.tx.jitter.max / 1000.0,
- audio->rtcp.stat.tx.jitter.last / 1000.0,
- ""
- );
-
-
- printf(" RTT delay : %7.3f %7.3f %7.3f %7.3f%s\n",
- audio->rtcp.stat.rtt.min / 1000.0,
- audio->rtcp.stat.rtt.avg / 1000.0,
- audio->rtcp.stat.rtt.max / 1000.0,
- audio->rtcp.stat.rtt.last / 1000.0,
- ""
- );
-
-}
+#include "siprtp_report.c"
static void list_calls()
diff --git a/pjsip-apps/src/samples/siprtp_report.c b/pjsip-apps/src/samples/siprtp_report.c
new file mode 100644
index 0000000..3b8b1eb
--- /dev/null
+++ b/pjsip-apps/src/samples/siprtp_report.c
@@ -0,0 +1,230 @@
+/* $Id$ */
+/*
+ * Copyright (C) 2003-2006 Benny Prijono <benny@prijono.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+/*
+ * DO NOT COMPILE THIS FILE ON ITS OWN!
+ *
+ * This file is included by siprtp.c to implement the reporting capability
+ * to a separate file, so that user can implement different reporting
+ * functionality (such as writing to XML file).
+ */
+
+static const char *good_number(char *buf, pj_int32_t val)
+{
+ if (val < 1000) {
+ pj_ansi_sprintf(buf, "%d", val);
+ } else if (val < 1000000) {
+ pj_ansi_sprintf(buf, "%d.%dK",
+ val / 1000,
+ (val % 1000) / 100);
+ } else {
+ pj_ansi_sprintf(buf, "%d.%02dM",
+ val / 1000000,
+ (val % 1000000) / 10000);
+ }
+
+ return buf;
+}
+
+
+static void print_call(int call_index)
+{
+ struct call *call = &app.call[call_index];
+ int len;
+ pjsip_inv_session *inv = call->inv;
+ pjsip_dialog *dlg = inv->dlg;
+ struct media_stream *audio = &call->media[0];
+ char userinfo[128];
+ char duration[80], last_update[80];
+ char bps[16], ipbps[16], packets[16], bytes[16], ipbytes[16];
+ pj_time_val now;
+
+ pj_gettimeofday(&now);
+
+ /* Print duration */
+ if (inv->state >= PJSIP_INV_STATE_CONFIRMED) {
+
+ PJ_TIME_VAL_SUB(now, call->connect_time);
+
+ sprintf(duration, " [duration: %02ld:%02ld:%02ld.%03ld]",
+ now.sec / 3600,
+ (now.sec % 3600) / 60,
+ (now.sec % 60),
+ now.msec);
+
+ } else {
+ duration[0] = '\0';
+ }
+
+
+
+ /* Call number and state */
+ printf("Call #%d: %s%s\n", call_index, pjsip_inv_state_name(inv->state),
+ duration);
+
+
+
+ /* Call identification */
+ len = pjsip_hdr_print_on(dlg->remote.info, userinfo, sizeof(userinfo));
+ if (len < 1)
+ pj_ansi_strcpy(userinfo, "<--uri too long-->");
+ else
+ userinfo[len] = '\0';
+
+ printf(" %s\n", userinfo);
+
+
+ if (call->inv == NULL || call->inv->state < PJSIP_INV_STATE_CONFIRMED ||
+ call->connect_time.sec == 0)
+ {
+ return;
+ }
+
+
+ /* Signaling quality */
+ {
+ char pdd[64], connectdelay[64];
+ pj_time_val t;
+
+ if (call->response_time.sec) {
+ t = call->response_time;
+ PJ_TIME_VAL_SUB(t, call->start_time);
+ sprintf(pdd, "got 1st response in %ld ms", PJ_TIME_VAL_MSEC(t));
+ } else {
+ pdd[0] = '\0';
+ }
+
+ if (call->connect_time.sec) {
+ t = call->connect_time;
+ PJ_TIME_VAL_SUB(t, call->start_time);
+ sprintf(connectdelay, ", connected after: %ld ms",
+ PJ_TIME_VAL_MSEC(t));
+ } else {
+ connectdelay[0] = '\0';
+ }
+
+ printf(" Signaling quality: %s%s\n", pdd, connectdelay);
+ }
+
+
+ printf(" Stream #0: audio %.*s@%dHz, %dms/frame, %sB/s (%sB/s +IP hdr)\n",
+ (int)audio->si.fmt.encoding_name.slen,
+ audio->si.fmt.encoding_name.ptr,
+ audio->clock_rate,
+ audio->samples_per_frame * 1000 / audio->clock_rate,
+ good_number(bps, audio->bytes_per_frame * audio->clock_rate / audio->samples_per_frame),
+ good_number(ipbps, (audio->bytes_per_frame+32) * audio->clock_rate / audio->samples_per_frame));
+
+ if (audio->rtcp.stat.rx.update_cnt == 0)
+ strcpy(last_update, "never");
+ else {
+ pj_gettimeofday(&now);
+ PJ_TIME_VAL_SUB(now, audio->rtcp.stat.rx.update);
+ sprintf(last_update, "%02ldh:%02ldm:%02ld.%03lds ago",
+ now.sec / 3600,
+ (now.sec % 3600) / 60,
+ now.sec % 60,
+ now.msec);
+ }
+
+ printf(" RX stat last update: %s\n"
+ " total %s packets %sB received (%sB +IP hdr)%s\n"
+ " pkt loss=%d (%3.1f%%), dup=%d (%3.1f%%), reorder=%d (%3.1f%%)%s\n"
+ " (msec) min avg max last\n"
+ " loss period: %7.3f %7.3f %7.3f %7.3f%s\n"
+ " jitter : %7.3f %7.3f %7.3f %7.3f%s\n",
+ last_update,
+ good_number(packets, audio->rtcp.stat.rx.pkt),
+ good_number(bytes, audio->rtcp.stat.rx.bytes),
+ good_number(ipbytes, audio->rtcp.stat.rx.bytes + audio->rtcp.stat.rx.pkt * 32),
+ "",
+ audio->rtcp.stat.rx.loss,
+ audio->rtcp.stat.rx.loss * 100.0 / audio->rtcp.stat.rx.pkt,
+ audio->rtcp.stat.rx.dup,
+ audio->rtcp.stat.rx.dup * 100.0 / audio->rtcp.stat.rx.pkt,
+ audio->rtcp.stat.rx.reorder,
+ audio->rtcp.stat.rx.reorder * 100.0 / audio->rtcp.stat.rx.pkt,
+ "",
+ audio->rtcp.stat.rx.loss_period.min / 1000.0,
+ audio->rtcp.stat.rx.loss_period.avg / 1000.0,
+ audio->rtcp.stat.rx.loss_period.max / 1000.0,
+ audio->rtcp.stat.rx.loss_period.last / 1000.0,
+ "",
+ audio->rtcp.stat.rx.jitter.min / 1000.0,
+ audio->rtcp.stat.rx.jitter.avg / 1000.0,
+ audio->rtcp.stat.rx.jitter.max / 1000.0,
+ audio->rtcp.stat.rx.jitter.last / 1000.0,
+ ""
+ );
+
+
+ if (audio->rtcp.stat.tx.update_cnt == 0)
+ strcpy(last_update, "never");
+ else {
+ pj_gettimeofday(&now);
+ PJ_TIME_VAL_SUB(now, audio->rtcp.stat.tx.update);
+ sprintf(last_update, "%02ldh:%02ldm:%02ld.%03lds ago",
+ now.sec / 3600,
+ (now.sec % 3600) / 60,
+ now.sec % 60,
+ now.msec);
+ }
+
+ printf(" TX stat last update: %s\n"
+ " total %s packets %sB sent (%sB +IP hdr)%s\n"
+ " pkt loss=%d (%3.1f%%), dup=%d (%3.1f%%), reorder=%d (%3.1f%%)%s\n"
+ " (msec) min avg max last\n"
+ " loss period: %7.3f %7.3f %7.3f %7.3f%s\n"
+ " jitter : %7.3f %7.3f %7.3f %7.3f%s\n",
+ last_update,
+ good_number(packets, audio->rtcp.stat.tx.pkt),
+ good_number(bytes, audio->rtcp.stat.tx.bytes),
+ good_number(ipbytes, audio->rtcp.stat.tx.bytes + audio->rtcp.stat.tx.pkt * 32),
+ "",
+ audio->rtcp.stat.tx.loss,
+ audio->rtcp.stat.tx.loss * 100.0 / audio->rtcp.stat.tx.pkt,
+ audio->rtcp.stat.tx.dup,
+ audio->rtcp.stat.tx.dup * 100.0 / audio->rtcp.stat.tx.pkt,
+ audio->rtcp.stat.tx.reorder,
+ audio->rtcp.stat.tx.reorder * 100.0 / audio->rtcp.stat.tx.pkt,
+ "",
+ audio->rtcp.stat.tx.loss_period.min / 1000.0,
+ audio->rtcp.stat.tx.loss_period.avg / 1000.0,
+ audio->rtcp.stat.tx.loss_period.max / 1000.0,
+ audio->rtcp.stat.tx.loss_period.last / 1000.0,
+ "",
+ audio->rtcp.stat.tx.jitter.min / 1000.0,
+ audio->rtcp.stat.tx.jitter.avg / 1000.0,
+ audio->rtcp.stat.tx.jitter.max / 1000.0,
+ audio->rtcp.stat.tx.jitter.last / 1000.0,
+ ""
+ );
+
+
+ printf(" RTT delay : %7.3f %7.3f %7.3f %7.3f%s\n",
+ audio->rtcp.stat.rtt.min / 1000.0,
+ audio->rtcp.stat.rtt.avg / 1000.0,
+ audio->rtcp.stat.rtt.max / 1000.0,
+ audio->rtcp.stat.rtt.last / 1000.0,
+ ""
+ );
+
+}
+