Changeset 407
- Timestamp:
- Apr 20, 2006 11:13:32 AM (19 years ago)
- Location:
- pjproject/trunk/pjsip-apps
- Files:
-
- 1 added
- 7 edited
Legend:
- Unmodified
- Added
- Removed
-
pjproject/trunk/pjsip-apps/build/Samples-vc.mak
r379 r407 38 38 39 39 40 SAMPLES = $(BINDIR)\simpleua.exe $(BINDIR)\playfile.exe $(BINDIR)\playsine.exe 40 SAMPLES = $(BINDIR)\simpleua.exe $(BINDIR)\playfile.exe $(BINDIR)\playsine.exe\ 41 41 $(BINDIR)\confsample.exe $(BINDIR)\sndinfo.exe \ 42 42 $(BINDIR)\level.exe $(BINDIR)\recfile.exe \ 43 $(BINDIR)\resampleplay.exe $(BINDIR)\siprtp.exe 43 $(BINDIR)\resampleplay.exe $(BINDIR)\siprtp.exe $(BINDIR)\stream.exe 44 44 45 45 -
pjproject/trunk/pjsip-apps/build/Samples.mak
r379 r407 40 40 41 41 SAMPLES := simpleua playfile playsine confsample sndinfo level recfile resampleplay \ 42 siprtp 42 siprtp stream 43 43 44 44 EXES := $(foreach file, $(SAMPLES), $(BINDIR)/$(file)-$(MACHINE_NAME)-$(OS_NAME)-$(CC_NAME)$(HOST_EXE)) -
pjproject/trunk/pjsip-apps/build/samples.dsp
r375 r407 121 121 SOURCE=..\src\samples\sndinfo.c 122 122 # End Source File 123 # Begin Source File 124 125 SOURCE=..\src\samples\stream.c 126 # End Source File 123 127 # End Group 124 128 # Begin Group "Header Files" … … 138 142 SOURCE=".\Samples-vc.mak" 139 143 # End Source File 144 # Begin Source File 145 146 SOURCE=.\Samples.mak 147 # End Source File 140 148 # End Target 141 149 # End Project -
pjproject/trunk/pjsip-apps/src/samples/debug.c
r375 r407 28 28 * #include "playfile.c" 29 29 */ 30 #include "s iprtp.c"30 #include "stream.c" 31 31 -
pjproject/trunk/pjsip-apps/src/samples/playfile.c
r361 r407 177 177 return 0; 178 178 } 179 -
pjproject/trunk/pjsip-apps/src/samples/siprtp.c
r395 r407 19 19 20 20 21 22 23 /* Usage */ 24 static const char *USAGE = 25 " PURPOSE: \n" 26 " This program establishes SIP INVITE session and media, and calculate \n" 27 " the media quality (packet lost, jitter, rtt, etc.). Unlike normal \n" 28 " pjmedia applications, this program bypasses all pjmedia stream \n" 29 " framework and transmit encoded RTP packets manually using own thread. \n" 30 "\n" 31 " USAGE:\n" 32 " siprtp [options] => to start in server mode\n" 33 " siprtp [options] URL => to start in client mode\n" 34 "\n" 35 " Program options:\n" 36 " --count=N, -c Set number of calls to create (default:1) \n" 37 "\n" 38 " Address and ports options:\n" 39 " --local-port=PORT,-p Set local SIP port (default: 5060)\n" 40 " --rtp-port=PORT, -r Set start of RTP port (default: 4000)\n" 41 " --ip-addr=IP, -i Set local IP address to use (otherwise it will\n" 42 " try to determine local IP address from hostname)\n" 43 "\n" 44 " Logging Options:\n" 45 " --log-level=N, -l Set log verbosity level (default=5)\n" 46 " --app-log-level=N Set app screen log verbosity (default=3)\n" 47 " --log-file=FILE Write log to file FILE\n" 48 "\n" 49 " Codec Options:\n" 50 " --a-pt=PT Set audio payload type to PT (default=0)\n" 51 " --a-name=NAME Set audio codec name to NAME (default=pcmu)\n" 52 " --a-clock=RATE Set audio codec rate to RATE Hz (default=8000Hz)\n" 53 " --a-bitrate=BPS Set audio codec bitrate to BPS (default=64000bps)\n" 54 " --a-ptime=MS Set audio frame time to MS msec (default=20ms)\n" 55 ; 56 57 21 58 /* Include all headers. */ 22 59 #include <pjsip.h> … … 724 761 return 0; 725 762 } 726 727 728 /* Usage */729 static const char *USAGE =730 "Usage:\n"731 " siprtp [options] => to start in server mode\n"732 " siprtp [options] URL => to start in client mode\n"733 "\n"734 "Program options:\n"735 " --count=N, -c Set number of calls to create (default:1) \n"736 "\n"737 "Address and ports options:\n"738 " --local-port=PORT,-p Set local SIP port (default: 5060)\n"739 " --rtp-port=PORT, -r Set start of RTP port (default: 4000)\n"740 " --ip-addr=IP, -i Set local IP address to use (otherwise it will\n"741 " try to determine local IP address from hostname)\n"742 "\n"743 "Logging Options:\n"744 " --log-level=N, -l Set log verbosity level (default=5)\n"745 " --app-log-level=N Set app screen log verbosity (default=3)\n"746 " --log-file=FILE Write log to file FILE\n"747 "\n"748 "Codec Options:\n"749 " --a-pt=PT Set audio payload type to PT (default=0)\n"750 " --a-name=NAME Set audio codec name to NAME (default=pcmu)\n"751 " --a-clock=RATE Set audio codec rate to RATE Hz (default=8000 Hz)\n"752 " --a-bitrate=BPS Set audio codec bitrate to BPS (default=64000 bps)\n"753 " --a-ptime=MS Set audio frame time to MS msec (default=20 msec)\n"754 ;755 763 756 764 -
pjproject/trunk/pjsip-apps/src/samples/util.h
r393 r407 37 37 * common sound options. 38 38 */ 39 static pj_status_t get_snd_options(const char *app_name,40 41 42 43 44 45 46 39 pj_status_t get_snd_options(const char *app_name, 40 int argc, 41 char *argv[], 42 int *dev_id, 43 int *clock_rate, 44 int *channel_count, 45 int *samples_per_frame, 46 int *bits_per_sample) 47 47 { 48 48 struct pj_getopt_option long_options[] = { … … 134 134 135 135 /* Dump memory pool usage. */ 136 staticvoid dump_pool_usage( const char *app_name, pj_caching_pool *cp )136 void dump_pool_usage( const char *app_name, pj_caching_pool *cp ) 137 137 { 138 138 #if !defined(PJ_HAS_POOL_ALT_API) || PJ_HAS_POOL_ALT_API==0
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