Added stream.c example to demonstrate how to stream audio from both sound device or WAV file
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@407 74dad513-b988-da41-8d7b-12977e46ad98
diff --git a/pjsip-apps/build/Samples-vc.mak b/pjsip-apps/build/Samples-vc.mak
index 43ed926..40ee2e9 100644
--- a/pjsip-apps/build/Samples-vc.mak
+++ b/pjsip-apps/build/Samples-vc.mak
@@ -37,10 +37,10 @@
BINDIR = ..\bin\samples
-SAMPLES = $(BINDIR)\simpleua.exe $(BINDIR)\playfile.exe $(BINDIR)\playsine.exe \
+SAMPLES = $(BINDIR)\simpleua.exe $(BINDIR)\playfile.exe $(BINDIR)\playsine.exe\
$(BINDIR)\confsample.exe $(BINDIR)\sndinfo.exe \
$(BINDIR)\level.exe $(BINDIR)\recfile.exe \
- $(BINDIR)\resampleplay.exe $(BINDIR)\siprtp.exe
+ $(BINDIR)\resampleplay.exe $(BINDIR)\siprtp.exe $(BINDIR)\stream.exe
all: $(OBJDIR) $(SAMPLES)
diff --git a/pjsip-apps/build/Samples.mak b/pjsip-apps/build/Samples.mak
index 749c939..d048ec5 100644
--- a/pjsip-apps/build/Samples.mak
+++ b/pjsip-apps/build/Samples.mak
@@ -39,7 +39,7 @@
BINDIR := ../bin/samples
SAMPLES := simpleua playfile playsine confsample sndinfo level recfile resampleplay \
- siprtp
+ siprtp stream
EXES := $(foreach file, $(SAMPLES), $(BINDIR)/$(file)-$(MACHINE_NAME)-$(OS_NAME)-$(CC_NAME)$(HOST_EXE))
diff --git a/pjsip-apps/build/samples.dsp b/pjsip-apps/build/samples.dsp
index be8b213..96289e7 100644
--- a/pjsip-apps/build/samples.dsp
+++ b/pjsip-apps/build/samples.dsp
@@ -120,6 +120,10 @@
SOURCE=..\src\samples\sndinfo.c
# End Source File
+# Begin Source File
+
+SOURCE=..\src\samples\stream.c
+# End Source File
# End Group
# Begin Group "Header Files"
@@ -137,5 +141,9 @@
SOURCE=".\Samples-vc.mak"
# End Source File
+# Begin Source File
+
+SOURCE=.\Samples.mak
+# End Source File
# End Target
# End Project
diff --git a/pjsip-apps/src/samples/debug.c b/pjsip-apps/src/samples/debug.c
index 259a7de..b6c1005 100644
--- a/pjsip-apps/src/samples/debug.c
+++ b/pjsip-apps/src/samples/debug.c
@@ -27,5 +27,5 @@
* E.g.:
* #include "playfile.c"
*/
-#include "siprtp.c"
+#include "stream.c"
diff --git a/pjsip-apps/src/samples/playfile.c b/pjsip-apps/src/samples/playfile.c
index 215ff41..1ae9a6c 100644
--- a/pjsip-apps/src/samples/playfile.c
+++ b/pjsip-apps/src/samples/playfile.c
@@ -176,3 +176,4 @@
/* Done. */
return 0;
}
+
diff --git a/pjsip-apps/src/samples/siprtp.c b/pjsip-apps/src/samples/siprtp.c
index 91cc93d..ca7f07d 100644
--- a/pjsip-apps/src/samples/siprtp.c
+++ b/pjsip-apps/src/samples/siprtp.c
@@ -18,6 +18,43 @@
*/
+
+
+/* Usage */
+static const char *USAGE =
+" PURPOSE: \n"
+" This program establishes SIP INVITE session and media, and calculate \n"
+" the media quality (packet lost, jitter, rtt, etc.). Unlike normal \n"
+" pjmedia applications, this program bypasses all pjmedia stream \n"
+" framework and transmit encoded RTP packets manually using own thread. \n"
+"\n"
+" USAGE:\n"
+" siprtp [options] => to start in server mode\n"
+" siprtp [options] URL => to start in client mode\n"
+"\n"
+" Program options:\n"
+" --count=N, -c Set number of calls to create (default:1) \n"
+"\n"
+" Address and ports options:\n"
+" --local-port=PORT,-p Set local SIP port (default: 5060)\n"
+" --rtp-port=PORT, -r Set start of RTP port (default: 4000)\n"
+" --ip-addr=IP, -i Set local IP address to use (otherwise it will\n"
+" try to determine local IP address from hostname)\n"
+"\n"
+" Logging Options:\n"
+" --log-level=N, -l Set log verbosity level (default=5)\n"
+" --app-log-level=N Set app screen log verbosity (default=3)\n"
+" --log-file=FILE Write log to file FILE\n"
+"\n"
+" Codec Options:\n"
+" --a-pt=PT Set audio payload type to PT (default=0)\n"
+" --a-name=NAME Set audio codec name to NAME (default=pcmu)\n"
+" --a-clock=RATE Set audio codec rate to RATE Hz (default=8000Hz)\n"
+" --a-bitrate=BPS Set audio codec bitrate to BPS (default=64000bps)\n"
+" --a-ptime=MS Set audio frame time to MS msec (default=20ms)\n"
+;
+
+
/* Include all headers. */
#include <pjsip.h>
#include <pjmedia.h>
@@ -725,35 +762,6 @@
}
-/* Usage */
-static const char *USAGE =
-"Usage:\n"
-" siprtp [options] => to start in server mode\n"
-" siprtp [options] URL => to start in client mode\n"
-"\n"
-"Program options:\n"
-" --count=N, -c Set number of calls to create (default:1) \n"
-"\n"
-"Address and ports options:\n"
-" --local-port=PORT,-p Set local SIP port (default: 5060)\n"
-" --rtp-port=PORT, -r Set start of RTP port (default: 4000)\n"
-" --ip-addr=IP, -i Set local IP address to use (otherwise it will\n"
-" try to determine local IP address from hostname)\n"
-"\n"
-"Logging Options:\n"
-" --log-level=N, -l Set log verbosity level (default=5)\n"
-" --app-log-level=N Set app screen log verbosity (default=3)\n"
-" --log-file=FILE Write log to file FILE\n"
-"\n"
-"Codec Options:\n"
-" --a-pt=PT Set audio payload type to PT (default=0)\n"
-" --a-name=NAME Set audio codec name to NAME (default=pcmu)\n"
-" --a-clock=RATE Set audio codec rate to RATE Hz (default=8000 Hz)\n"
-" --a-bitrate=BPS Set audio codec bitrate to BPS (default=64000 bps)\n"
-" --a-ptime=MS Set audio frame time to MS msec (default=20 msec)\n"
-;
-
-
/* Init application options */
static pj_status_t init_options(int argc, char *argv[])
{
diff --git a/pjsip-apps/src/samples/stream.c b/pjsip-apps/src/samples/stream.c
new file mode 100644
index 0000000..8503c1b
--- /dev/null
+++ b/pjsip-apps/src/samples/stream.c
@@ -0,0 +1,681 @@
+/* $Id$ */
+/*
+ * 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
+ */
+
+
+static const char *desc =
+ " stream.c \n"
+ " \n"
+ " PURPOSE: \n"
+ " Demonstrate how to use pjmedia stream component to transmit/receive \n"
+ " RTP packets to/from sound device. \n"
+ "\n"
+ "\n"
+ " USAGE: \n"
+ " stream [options] \n"
+ "\n"
+ "\n"
+ " Options:\n"
+ " --codec=CODEC Set the codec name. Valid codec names are: \n"
+ " pcma, pcmu, gsm, speexnb, speexwb, speexuwb. \n"
+ " (default: pcma). \n"
+ " --local-port=PORT Set local RTP port (default=4000) \n"
+ " --remote=IP:PORT Set the remote peer. If this option is set, \n"
+ " the program will transmit RTP audio to the \n"
+ " specified address. (default: recv only) \n"
+ " --play-file=WAV Send audio from the WAV file instead of from \n"
+ " the sound device. \n"
+// " --record-file=WAV Record incoming audio to WAV file instead of \n"
+// " playing it to sound device. \n"
+ " --send-recv Set stream direction to bidirectional. \n"
+ " --send-only Set stream direction to send only \n"
+ " --recv-only Set stream direction to recv only (default) \n"
+ "\n"
+;
+
+
+
+#include <pjlib.h>
+#include <pjlib-util.h>
+#include <pjmedia.h>
+#include <pjmedia-codec.h>
+
+#include <stdlib.h> /* atoi() */
+#include <stdio.h>
+
+#include "util.h"
+
+
+#define THIS_FILE "stream.c"
+
+
+struct codec
+{
+ char *name;
+ char *encoding_name;
+ int pt;
+ int clock_rate;
+} codec[] =
+{
+ { "pcma", "pcma", PJMEDIA_RTP_PT_PCMA, 8000 },
+ { "pcmu", "pcmu", PJMEDIA_RTP_PT_PCMU, 8000 },
+ { "gsm", "gsm", PJMEDIA_RTP_PT_GSM, 8000 },
+ { "speexnb", "speex", 120, 8000 },
+ { "speexwb", "speex", 121, 16000 },
+ { "speexuwb", "speex", 122, 32000 },
+};
+
+
+/* Prototype */
+static void print_stream_stat(pjmedia_stream *stream);
+
+
+/*
+ * Register all codecs.
+ */
+static pj_status_t init_codecs(pjmedia_endpt *med_endpt)
+{
+ pj_status_t status;
+
+ status = pjmedia_codec_g711_init(med_endpt);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
+
+ status = pjmedia_codec_gsm_init(med_endpt);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
+
+ status = pjmedia_codec_speex_init(med_endpt, 0, -1, -1);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
+
+ return PJ_SUCCESS;
+}
+
+
+/*
+ * Create stream based on the codec, dir, remote address, etc.
+ */
+static pj_status_t create_stream( pj_pool_t *pool,
+ pjmedia_endpt *med_endpt,
+ unsigned codec_index,
+ pjmedia_dir dir,
+ pj_uint16_t local_port,
+ const pj_sockaddr_in *rem_addr,
+ pjmedia_stream **p_stream )
+{
+ pjmedia_stream_info info;
+ pj_status_t status;
+
+
+ /* Reset stream info. */
+ pj_memset(&info, 0, sizeof(info));
+
+
+ /* Initialize stream info formats */
+ info.type = PJMEDIA_TYPE_AUDIO;
+ info.dir = dir;
+ info.fmt.encoding_name = pj_str(codec[codec_index].encoding_name);
+ info.fmt.type = PJMEDIA_TYPE_AUDIO;
+ info.fmt.sample_rate = codec[codec_index].clock_rate;
+ info.fmt.pt = codec[codec_index].pt;
+ info.tx_pt = codec[codec_index].pt;
+ info.ssrc = pj_rand();
+
+
+ /* Copy remote address */
+ pj_memcpy(&info.rem_addr, rem_addr, sizeof(pj_sockaddr_in));
+
+
+ /* Create RTP socket */
+ status = pj_sock_socket(PJ_AF_INET, PJ_SOCK_DGRAM, 0,
+ &info.sock_info.rtp_sock);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
+
+
+ /* Bind RTP socket to local port */
+ info.sock_info.rtp_addr_name.sin_family = PJ_AF_INET;
+ info.sock_info.rtp_addr_name.sin_port = pj_htons(local_port);
+
+ status = pj_sock_bind(info.sock_info.rtp_sock,
+ &info.sock_info.rtp_addr_name,
+ sizeof(pj_sockaddr_in));
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Unable to bind RTP socket", status);
+ pj_sock_close(info.sock_info.rtp_sock);
+ return status;
+ }
+
+
+ /* Create RTCP socket */
+ status = pj_sock_socket(PJ_AF_INET, PJ_SOCK_DGRAM, 0,
+ &info.sock_info.rtcp_sock);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
+
+
+ /* Bind RTP socket to local port + 1 */
+ ++local_port;
+ info.sock_info.rtcp_addr_name.sin_family = PJ_AF_INET;
+ info.sock_info.rtcp_addr_name.sin_port = pj_htons(local_port);
+
+ status = pj_sock_bind(info.sock_info.rtcp_sock,
+ &info.sock_info.rtcp_addr_name,
+ sizeof(pj_sockaddr_in));
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Unable to bind RTCP socket", status);
+ pj_sock_close(info.sock_info.rtp_sock);
+ pj_sock_close(info.sock_info.rtcp_sock);
+ return status;
+ }
+
+
+ /* Now that the stream info is initialized, we can create the
+ * stream.
+ */
+
+ status = pjmedia_stream_create( med_endpt, pool, &info, NULL, p_stream);
+
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Error creating stream", status);
+ pj_sock_close(info.sock_info.rtp_sock);
+ pj_sock_close(info.sock_info.rtcp_sock);
+ return status;
+ }
+
+
+ return PJ_SUCCESS;
+}
+
+
+/*
+ * main()
+ */
+int main(int argc, char *argv[])
+{
+ pj_caching_pool cp;
+ pjmedia_endpt *med_endpt;
+ pj_pool_t *pool;
+ pjmedia_port *rec_file_port = NULL, *play_file_port = NULL;
+ pjmedia_master_port *master_port = NULL;
+ pjmedia_snd_port *snd_port = NULL;
+ pjmedia_stream *stream = NULL;
+ pjmedia_port *stream_port;
+ char tmp[10];
+ pj_status_t status;
+
+
+ /* Default values */
+ int codec_index = 0;
+ pjmedia_dir dir = PJMEDIA_DIR_DECODING;
+ pj_sockaddr_in remote_addr;
+ pj_uint16_t local_port = 4000;
+ char *rec_file = NULL;
+ char *play_file = NULL;
+
+ enum {
+ OPT_CODEC = 'c',
+ OPT_LOCAL_PORT = 'p',
+ OPT_REMOTE = 'r',
+ OPT_PLAY_FILE = 'w',
+ OPT_RECORD_FILE = 'R',
+ OPT_SEND_RECV = 'b',
+ OPT_SEND_ONLY = 's',
+ OPT_RECV_ONLY = 'i',
+ };
+
+ struct pj_getopt_option long_options[] = {
+ { "codec", 1, 0, OPT_CODEC },
+ { "local-port", 1, 0, OPT_LOCAL_PORT },
+ { "remote", 1, 0, OPT_REMOTE },
+ { "play-file", 1, 0, OPT_PLAY_FILE },
+ { "record-file", 1, 0, OPT_RECORD_FILE },
+ { "send-recv", 0, 0, OPT_SEND_RECV },
+ { "send-only", 0, 0, OPT_SEND_ONLY },
+ { "recv-only", 0, 0, OPT_RECV_ONLY },
+ { NULL, 0, 0, 0 },
+ };
+
+ int c;
+ int option_index;
+
+
+ pj_memset(&remote_addr, 0, sizeof(remote_addr));
+
+
+ /* init PJLIB : */
+ status = pj_init();
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+
+ /* Parse arguments */
+ pj_optind = 0;
+ while((c=pj_getopt_long(argc,argv, "", long_options, &option_index))!=-1) {
+
+ switch (c) {
+ case OPT_CODEC:
+ {
+ unsigned i;
+ for (i=0; i<PJ_ARRAY_SIZE(codec); ++i) {
+ if (pj_ansi_stricmp(pj_optarg, codec[i].name)==0) {
+ break;
+ }
+ }
+
+ if (i == PJ_ARRAY_SIZE(codec)) {
+ printf("Error: unknown codec %s\n", pj_optarg);
+ return 1;
+ }
+
+ codec_index = i;
+ }
+ break;
+
+ case OPT_LOCAL_PORT:
+ local_port = (pj_uint16_t) atoi(pj_optarg);
+ if (local_port < 1) {
+ printf("Error: invalid local port %s\n", pj_optarg);
+ return 1;
+ }
+ break;
+
+ case OPT_REMOTE:
+ {
+ pj_str_t ip = pj_str(strtok(pj_optarg, ":"));
+ pj_uint16_t port = (pj_uint16_t) atoi(strtok(NULL, ":"));
+
+ status = pj_sockaddr_in_init(&remote_addr, &ip, port);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Invalid remote address", status);
+ return 1;
+ }
+ }
+ break;
+
+ case OPT_PLAY_FILE:
+ play_file = pj_optarg;
+ break;
+
+ case OPT_RECORD_FILE:
+ rec_file = pj_optarg;
+ break;
+
+ case OPT_SEND_RECV:
+ dir = PJMEDIA_DIR_ENCODING_DECODING;
+ break;
+
+ case OPT_SEND_ONLY:
+ dir = PJMEDIA_DIR_ENCODING;
+ break;
+
+ case OPT_RECV_ONLY:
+ dir = PJMEDIA_DIR_DECODING;
+ break;
+
+ default:
+ printf("Invalid options %s\n", argv[pj_optind]);
+ return 1;
+ }
+
+ }
+
+
+ /* Verify arguments. */
+ if (dir & PJMEDIA_DIR_ENCODING) {
+ if (remote_addr.sin_addr.s_addr == 0) {
+ printf("Error: remote address must be set\n");
+ return 1;
+ }
+ }
+
+ if (play_file != NULL && dir != PJMEDIA_DIR_ENCODING) {
+ printf("Direction is set to --send-only because of --play-file\n");
+ dir = PJMEDIA_DIR_ENCODING;
+ }
+
+
+ /* Must create a pool factory before we can allocate any memory. */
+ pj_caching_pool_init(&cp, &pj_pool_factory_default_policy, 0);
+
+ /*
+ * Initialize media endpoint.
+ * This will implicitly initialize PJMEDIA too.
+ */
+ status = pjmedia_endpt_create(&cp.factory, NULL, 1, &med_endpt);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ /* Create memory pool for application purpose */
+ pool = pj_pool_create( &cp.factory, /* pool factory */
+ "app", /* pool name. */
+ 4000, /* init size */
+ 4000, /* increment size */
+ NULL /* callback on error */
+ );
+
+
+ /* Register all supported codecs */
+ status = init_codecs(med_endpt);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+
+ /* Create stream based on program arguments */
+ status = create_stream(pool, med_endpt, codec_index, dir, local_port,
+ &remote_addr, &stream);
+ if (status != PJ_SUCCESS)
+ goto on_exit;
+
+
+ /* Get the port interface of the stream */
+ status = pjmedia_stream_get_port( stream, &stream_port);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+
+ if (play_file) {
+
+ status = pjmedia_file_player_port_create(pool, play_file, 0,
+ -1, NULL, &play_file_port);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Unable to use file", status);
+ goto on_exit;
+ }
+
+ status = pjmedia_master_port_create(pool, play_file_port, stream_port,
+ 0, &master_port);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Unable to create master port", status);
+ goto on_exit;
+ }
+
+ status = pjmedia_master_port_start(master_port);
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Error starting master port", status);
+ goto on_exit;
+ }
+
+
+ } else {
+
+ /* Create sound device port. */
+ if (dir == PJMEDIA_DIR_ENCODING_DECODING)
+ status = pjmedia_snd_port_create(pool, -1, -1,
+ stream_port->info.sample_rate,
+ stream_port->info.channel_count,
+ stream_port->info.samples_per_frame,
+ stream_port->info.bits_per_sample,
+ 0, &snd_port);
+ else if (dir == PJMEDIA_DIR_ENCODING)
+ status = pjmedia_snd_port_create_rec(pool, -1,
+ stream_port->info.sample_rate,
+ stream_port->info.channel_count,
+ stream_port->info.samples_per_frame,
+ stream_port->info.bits_per_sample,
+ 0, &snd_port);
+ else
+ status = pjmedia_snd_port_create_player(pool, -1,
+ stream_port->info.sample_rate,
+ stream_port->info.channel_count,
+ stream_port->info.samples_per_frame,
+ stream_port->info.bits_per_sample,
+ 0, &snd_port);
+
+
+ if (status != PJ_SUCCESS) {
+ app_perror(THIS_FILE, "Unable to create sound port", status);
+ goto on_exit;
+ }
+
+ /* Connect sound port to stream */
+ status = pjmedia_snd_port_connect( snd_port, stream_port );
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ }
+
+
+ /* Done */
+
+ if (dir == PJMEDIA_DIR_DECODING)
+ printf("Stream is active, dir is recv-only, local port is %d\n",
+ local_port);
+ else if (dir == PJMEDIA_DIR_ENCODING)
+ printf("Stream is active, dir is send-only, sending to %s:%d\n",
+ pj_inet_ntoa(remote_addr.sin_addr),
+ pj_ntohs(remote_addr.sin_port));
+ else
+ printf("Stream is active, send/recv, local port is %d, "
+ "sending to %s:%d\n",
+ local_port,
+ pj_inet_ntoa(remote_addr.sin_addr),
+ pj_ntohs(remote_addr.sin_port));
+
+
+ for (;;) {
+
+ puts("");
+ puts("Commands:");
+ puts(" s Display media statistics");
+ puts(" q Quit");
+ puts("");
+
+ printf("Command: "); fflush(stdout);
+
+ fgets(tmp, sizeof(tmp), stdin);
+
+ if (tmp[0] == 's')
+ print_stream_stat(stream);
+ else if (tmp[0] == 'q')
+ break;
+
+ }
+
+
+
+ /* Start deinitialization: */
+on_exit:
+
+ /* Destroy sound device */
+ if (snd_port) {
+ pjmedia_snd_port_destroy( snd_port );
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ }
+
+ /* If there is master port, then we just need to destroy master port
+ * (it will recursively destroy upstream and downstream ports, which
+ * in this case are file_port and stream_port).
+ */
+ if (master_port) {
+ pjmedia_master_port_destroy(master_port);
+ play_file_port = NULL;
+ stream = NULL;
+ }
+
+ /* Destroy stream */
+ if (stream) {
+ pjmedia_stream_destroy(stream);
+ }
+
+ /* Destroy file ports */
+ if (play_file_port)
+ pjmedia_port_destroy( play_file_port );
+ if (rec_file_port)
+ pjmedia_port_destroy( rec_file_port );
+
+
+ /* Release application pool */
+ pj_pool_release( pool );
+
+ /* Destroy media endpoint. */
+ pjmedia_endpt_destroy( med_endpt );
+
+ /* Destroy pool factory */
+ pj_caching_pool_destroy( &cp );
+
+
+ return (status == PJ_SUCCESS) ? 0 : 1;
+}
+
+
+
+
+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;
+}
+
+
+
+/*
+ * Print stream statistics
+ */
+static void print_stream_stat(pjmedia_stream *stream)
+{
+ char duration[80], last_update[80];
+ char bps[16], ipbps[16], packets[16], bytes[16], ipbytes[16];
+ pjmedia_port *port;
+ pjmedia_rtcp_stat stat;
+ pj_time_val now;
+
+
+ pj_gettimeofday(&now);
+ pjmedia_stream_get_stat(stream, &stat);
+ pjmedia_stream_get_port(stream, &port);
+
+ puts("Stream statistics:");
+
+ /* Print duration */
+ PJ_TIME_VAL_SUB(now, stat.start);
+ sprintf(duration, " Duration: %02ld:%02ld:%02ld.%03ld",
+ now.sec / 3600,
+ (now.sec % 3600) / 60,
+ (now.sec % 60),
+ now.msec);
+
+
+ printf(" Info: audio %.*s@%dHz, %dms/frame, %sB/s (%sB/s +IP hdr)\n",
+ (int)port->info.encoding_name.slen,
+ port->info.encoding_name.ptr,
+ port->info.sample_rate,
+ port->info.samples_per_frame * 1000 / port->info.sample_rate,
+ good_number(bps, port->info.bytes_per_frame * port->info.sample_rate /
+ port->info.sample_rate),
+ good_number(ipbps, (port->info.bytes_per_frame+32) *
+ port->info.sample_rate / port->info.sample_rate));
+
+ if (stat.rx.update_cnt == 0)
+ strcpy(last_update, "never");
+ else {
+ pj_gettimeofday(&now);
+ PJ_TIME_VAL_SUB(now, 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, stat.rx.pkt),
+ good_number(bytes, stat.rx.bytes),
+ good_number(ipbytes, stat.rx.bytes + stat.rx.pkt * 32),
+ "",
+ stat.rx.loss,
+ stat.rx.loss * 100.0 / stat.rx.pkt,
+ stat.rx.dup,
+ stat.rx.dup * 100.0 / stat.rx.pkt,
+ stat.rx.reorder,
+ stat.rx.reorder * 100.0 / stat.rx.pkt,
+ "",
+ stat.rx.loss_period.min / 1000.0,
+ stat.rx.loss_period.avg / 1000.0,
+ stat.rx.loss_period.max / 1000.0,
+ stat.rx.loss_period.last / 1000.0,
+ "",
+ stat.rx.jitter.min / 1000.0,
+ stat.rx.jitter.avg / 1000.0,
+ stat.rx.jitter.max / 1000.0,
+ stat.rx.jitter.last / 1000.0,
+ ""
+ );
+
+
+ if (stat.tx.update_cnt == 0)
+ strcpy(last_update, "never");
+ else {
+ pj_gettimeofday(&now);
+ PJ_TIME_VAL_SUB(now, 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, stat.tx.pkt),
+ good_number(bytes, stat.tx.bytes),
+ good_number(ipbytes, stat.tx.bytes + stat.tx.pkt * 32),
+ "",
+ stat.tx.loss,
+ stat.tx.loss * 100.0 / stat.tx.pkt,
+ stat.tx.dup,
+ stat.tx.dup * 100.0 / stat.tx.pkt,
+ stat.tx.reorder,
+ stat.tx.reorder * 100.0 / stat.tx.pkt,
+ "",
+ stat.tx.loss_period.min / 1000.0,
+ stat.tx.loss_period.avg / 1000.0,
+ stat.tx.loss_period.max / 1000.0,
+ stat.tx.loss_period.last / 1000.0,
+ "",
+ stat.tx.jitter.min / 1000.0,
+ stat.tx.jitter.avg / 1000.0,
+ stat.tx.jitter.max / 1000.0,
+ stat.tx.jitter.last / 1000.0,
+ ""
+ );
+
+
+ printf(" RTT delay : %7.3f %7.3f %7.3f %7.3f%s\n",
+ stat.rtt.min / 1000.0,
+ stat.rtt.avg / 1000.0,
+ stat.rtt.max / 1000.0,
+ stat.rtt.last / 1000.0,
+ ""
+ );
+
+}
+
diff --git a/pjsip-apps/src/samples/util.h b/pjsip-apps/src/samples/util.h
index 2e2a32c..02ff702 100644
--- a/pjsip-apps/src/samples/util.h
+++ b/pjsip-apps/src/samples/util.h
@@ -36,14 +36,14 @@
* This utility function parses the command line and look for
* common sound options.
*/
-static pj_status_t get_snd_options( const char *app_name,
- int argc,
- char *argv[],
- int *dev_id,
- int *clock_rate,
- int *channel_count,
- int *samples_per_frame,
- int *bits_per_sample)
+pj_status_t get_snd_options(const char *app_name,
+ int argc,
+ char *argv[],
+ int *dev_id,
+ int *clock_rate,
+ int *channel_count,
+ int *samples_per_frame,
+ int *bits_per_sample)
{
struct pj_getopt_option long_options[] = {
{ "dev", 1, 0, 'd' },
@@ -133,7 +133,7 @@
/* Dump memory pool usage. */
-static void dump_pool_usage( const char *app_name, pj_caching_pool *cp )
+void dump_pool_usage( const char *app_name, pj_caching_pool *cp )
{
#if !defined(PJ_HAS_POOL_ALT_API) || PJ_HAS_POOL_ALT_API==0
pj_pool_t *p;