diff options
author | Benny Prijono <bennylp@teluu.com> | 2006-06-22 22:31:48 +0000 |
---|---|---|
committer | Benny Prijono <bennylp@teluu.com> | 2006-06-22 22:31:48 +0000 |
commit | 449c6e01f345c7859876ab2d0d627150636a9885 (patch) | |
tree | 6c68bad8839244eebaeea54549fa3f02373dc474 /pjsip-apps/src | |
parent | 79e892ee8bced558ae329b4ba70b777427af212f (diff) |
Changed siprtp to use timer to schedule transmissions of RTP/RTCP packets
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@545 74dad513-b988-da41-8d7b-12977e46ad98
Diffstat (limited to 'pjsip-apps/src')
-rw-r--r-- | pjsip-apps/src/samples/siprtp.c | 318 |
1 files changed, 146 insertions, 172 deletions
diff --git a/pjsip-apps/src/samples/siprtp.c b/pjsip-apps/src/samples/siprtp.c index 29447291..6f083168 100644 --- a/pjsip-apps/src/samples/siprtp.c +++ b/pjsip-apps/src/samples/siprtp.c @@ -122,9 +122,9 @@ struct media_stream /* RTCP stats: */ pjmedia_rtcp_session rtcp; /* incoming RTCP session. */ - /* Thread: */ - pj_bool_t thread_quit_flag; /* Stop media thread. */ - pj_thread_t *thread; /* Media thread. */ + /* Timer to send RTP and RTCP: */ + pj_timer_entry rtp_timer; /* timer to send RTP pkt. */ + pj_timer_entry rtcp_timer; /* timer to send RTCP pkt. */ }; @@ -222,6 +222,15 @@ static void on_rx_rtp(void *user_data, const void *pkt, pj_ssize_t size); /* This callback is called by media transport on receipt of RTCP packet. */ static void on_rx_rtcp(void *user_data, const void *pkt, pj_ssize_t size); +/* This callback is called when it's time to send RTP packet */ +static void on_tx_rtp( pj_timer_heap_t *timer_heap, + struct pj_timer_entry *entry); + +/* This callback is called when it's time to send RTCP packet. */ +static void on_tx_rtcp(pj_timer_heap_t *timer_heap, + struct pj_timer_entry *entry); + + /* Display error */ static void app_perror(const char *sender, const char *title, pj_status_t status); @@ -387,7 +396,9 @@ static pj_status_t init_media() /* Initialize media endpoint so that at least error subsystem is properly * initialized. */ - status = pjmedia_endpt_create(&app.cp.factory, NULL, 1, &app.med_endpt); + status = pjmedia_endpt_create(&app.cp.factory, + pjsip_endpt_get_ioqueue(app.sip_endpt), 1, + &app.med_endpt); PJ_ASSERT_RETURN(status == PJ_SUCCESS, status); @@ -407,10 +418,18 @@ static pj_status_t init_media() /* Repeat binding media socket to next port when fails to bind * to current port number. */ + struct media_stream *m = &app.call[i].media[j]; int retry; - app.call[i].media[j].call_index = i; - app.call[i].media[j].media_index = j; + m->call_index = i; + m->media_index = j; + + m->rtp_timer.user_data = m; + m->rtp_timer.cb = &on_tx_rtp; + + m->rtcp_timer.user_data = m; + m->rtcp_timer.cb = &on_tx_rtcp; + status = -1; for (retry=0; retry<100; ++retry,rtp_port+=2) { @@ -756,13 +775,28 @@ static void app_perror(const char *sender, const char *title, } +#if defined(PJ_WIN32) && PJ_WIN32 != 0 +#include <windows.h> +static void boost_priority(void) +{ + SetPriorityClass( GetCurrentProcess(), REALTIME_PRIORITY_CLASS); + SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST); +} + +#else +# define boost_priority() +#endif + + /* Worker thread for SIP */ static int sip_worker_thread(void *arg) { PJ_UNUSED_ARG(arg); + boost_priority(); + while (!app.thread_quit) { - pj_time_val timeout = {0, 10}; + pj_time_val timeout = {0, 1}; pjsip_endpt_handle_events(app.sip_endpt, &timeout); } @@ -1021,19 +1055,6 @@ static pj_status_t create_sdp( pj_pool_t *pool, } -#if defined(PJ_WIN32) && PJ_WIN32 != 0 -#include <windows.h> -static void boost_priority(void) -{ - SetPriorityClass( GetCurrentProcess(), REALTIME_PRIORITY_CLASS); - SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST); -} - -#else -# define boost_priority() -#endif - - /* * This callback is called by media transport on receipt of RTP packet. */ @@ -1077,6 +1098,62 @@ static void on_rx_rtp(void *user_data, const void *pkt, pj_ssize_t size) } +/* This callback is called when it's time to send RTP packet */ +static void on_tx_rtp( pj_timer_heap_t *timer_heap, + struct pj_timer_entry *entry) +{ + pj_status_t status; + const pjmedia_rtp_hdr *hdr; + pj_ssize_t size; + int hdrlen; + pj_time_val interval; + char packet[512]; + struct media_stream *strm = entry->user_data; + + PJ_UNUSED_ARG(timer_heap); + + if (!strm->active) + return; + + /* Format RTP header */ + status = pjmedia_rtp_encode_rtp( &strm->out_sess, strm->si.tx_pt, + 0, /* marker bit */ + strm->bytes_per_frame, + strm->samples_per_frame, + (const void**)&hdr, &hdrlen); + if (status == PJ_SUCCESS) { + + //PJ_LOG(4,(THIS_FILE, "\t\tTx seq=%d", pj_ntohs(hdr->seq))); + + /* Copy RTP header to packet */ + pj_memcpy(packet, hdr, hdrlen); + + /* Zero the payload */ + pj_memset(packet+hdrlen, 0, strm->bytes_per_frame); + + /* Send RTP packet */ + size = hdrlen + strm->bytes_per_frame; + status = pjmedia_transport_send_rtp(strm->transport, + packet, size); + if (status != PJ_SUCCESS) + app_perror(THIS_FILE, "Error sending RTP packet", status); + + } else { + pj_assert(!"RTP encode() error"); + } + + /* Update RTCP SR */ + pjmedia_rtcp_tx_rtp( &strm->rtcp, (pj_uint16_t)strm->bytes_per_frame); + + /* Schedule next send */ + interval.sec = 0; + interval.msec = strm->samples_per_frame * 1000 / strm->clock_rate; + pj_time_val_normalize(&interval); + + pjsip_endpt_schedule_timer(app.sip_endpt, &strm->rtp_timer, &interval); +} + + /* * This callback is called by media transport on receipt of RTCP packet. */ @@ -1101,150 +1178,40 @@ static void on_rx_rtcp(void *user_data, const void *pkt, pj_ssize_t size) } -/* - * Media thread - * - * This is the thread to send and receive both RTP and RTCP packets. - */ -static int media_thread(void *arg) +/* This callback is called when it's time to send RTCP packet. */ +static void on_tx_rtcp(pj_timer_heap_t *timer_heap, + struct pj_timer_entry *entry) { - enum { RTCP_INTERVAL = 5000, RTCP_RAND = 2000 }; - struct media_stream *strm = arg; - char packet[1500]; - unsigned msec_interval; - pj_timestamp freq, next_rtp, next_rtcp; - - - /* Boost thread priority if necessary */ - boost_priority(); - - /* Let things settle */ - pj_thread_sleep(1000); - - msec_interval = strm->samples_per_frame * 1000 / strm->clock_rate; - pj_get_timestamp_freq(&freq); - - pj_get_timestamp(&next_rtp); - next_rtp.u64 += (freq.u64 * msec_interval / 1000); - - next_rtcp = next_rtp; - next_rtcp.u64 += (freq.u64 * (RTCP_INTERVAL+(pj_rand()%RTCP_RAND)) / 1000); - - - while (!strm->thread_quit_flag) { - pj_timestamp now, lesser; - pj_time_val timeout; - pj_bool_t send_rtp, send_rtcp; - - send_rtp = send_rtcp = PJ_FALSE; - - /* Determine how long to sleep */ - if (next_rtp.u64 < next_rtcp.u64) { - lesser = next_rtp; - send_rtp = PJ_TRUE; - } else { - lesser = next_rtcp; - send_rtcp = PJ_TRUE; - } - - pj_get_timestamp(&now); - if (lesser.u64 <= now.u64) { - timeout.sec = timeout.msec = 0; - //printf("immediate "); fflush(stdout); - } else { - pj_uint64_t tick_delay; - tick_delay = lesser.u64 - now.u64; - timeout.sec = 0; - timeout.msec = (pj_uint32_t)(tick_delay * 1000 / freq.u64); - pj_time_val_normalize(&timeout); - - //printf("%d:%03d ", timeout.sec, timeout.msec); fflush(stdout); - } - - /* Wait for next interval */ - //if (timeout.sec!=0 && timeout.msec!=0) { - pj_thread_sleep(PJ_TIME_VAL_MSEC(timeout)); - if (strm->thread_quit_flag) - break; - //} - - pj_get_timestamp(&now); - - if (send_rtp || next_rtp.u64 <= now.u64) { - /* - * Time to send RTP packet. - */ - pj_status_t status; - const pjmedia_rtp_hdr *hdr; - pj_ssize_t size; - int hdrlen; - - /* Format RTP header */ - status = pjmedia_rtp_encode_rtp( &strm->out_sess, strm->si.tx_pt, - 0, /* marker bit */ - strm->bytes_per_frame, - strm->samples_per_frame, - (const void**)&hdr, &hdrlen); - if (status == PJ_SUCCESS) { - - //PJ_LOG(4,(THIS_FILE, "\t\tTx seq=%d", pj_ntohs(hdr->seq))); - - /* Copy RTP header to packet */ - pj_memcpy(packet, hdr, hdrlen); - - /* Zero the payload */ - pj_memset(packet+hdrlen, 0, strm->bytes_per_frame); - - /* Send RTP packet */ - size = hdrlen + strm->bytes_per_frame; - status = pjmedia_transport_send_rtp(strm->transport, - packet, size); - if (status != PJ_SUCCESS) - app_perror(THIS_FILE, "Error sending RTP packet", status); - - } else { - pj_assert(!"RTP encode() error"); - } - - /* Update RTCP SR */ - pjmedia_rtcp_tx_rtp( &strm->rtcp, (pj_uint16_t)strm->bytes_per_frame); - - /* Schedule next send */ - next_rtp.u64 += (msec_interval * freq.u64 / 1000); - } + pjmedia_rtcp_pkt *rtcp_pkt; + int rtcp_len; + pj_ssize_t size; + pj_status_t status; + pj_time_val interval; + struct media_stream *strm = entry->user_data; + PJ_UNUSED_ARG(timer_heap); - if (send_rtcp || next_rtcp.u64 <= now.u64) { - /* - * Time to send RTCP packet. - */ - pjmedia_rtcp_pkt *rtcp_pkt; - int rtcp_len; - pj_ssize_t size; - pj_status_t status; + if (!strm->active) + return; - /* Build RTCP packet */ - pjmedia_rtcp_build_rtcp(&strm->rtcp, &rtcp_pkt, &rtcp_len); + /* Build RTCP packet */ + pjmedia_rtcp_build_rtcp(&strm->rtcp, &rtcp_pkt, &rtcp_len); - - /* Send packet */ - size = rtcp_len; - status = pjmedia_transport_send_rtcp(strm->transport, - rtcp_pkt, size); - if (status != PJ_SUCCESS) { - app_perror(THIS_FILE, "Error sending RTCP packet", status); - } - - /* Schedule next send */ - next_rtcp.u64 += (freq.u64 * (RTCP_INTERVAL+(pj_rand()%RTCP_RAND)) / - 1000); - } + /* Send packet */ + size = rtcp_len; + status = pjmedia_transport_send_rtcp(strm->transport, + rtcp_pkt, size); + if (status != PJ_SUCCESS) { + app_perror(THIS_FILE, "Error sending RTCP packet", status); } + + /* Schedule next send */ + interval.sec = 5; + interval.msec = (pj_rand() % 500); + pjsip_endpt_schedule_timer(app.sip_endpt, &strm->rtcp_timer, &interval); - return 0; } - /* Callback to be called when SDP negotiation is done in the call: */ static void call_on_media_update( pjsip_inv_session *inv, pj_status_t status) @@ -1254,6 +1221,7 @@ static void call_on_media_update( pjsip_inv_session *inv, struct media_stream *audio; const pjmedia_sdp_session *local_sdp, *remote_sdp; struct codec *codec_desc = NULL; + pj_time_val interval; unsigned i; call = inv->mod_data[mod_siprtp.id]; @@ -1261,7 +1229,7 @@ static void call_on_media_update( pjsip_inv_session *inv, audio = &call->media[0]; /* If this is a mid-call media update, then destroy existing media */ - if (audio->thread != NULL) + if (audio->active) destroy_call_media(call->index); @@ -1324,17 +1292,19 @@ static void call_on_media_update( pjsip_inv_session *inv, return; } - /* Start media thread. */ - audio->thread_quit_flag = 0; - status = pj_thread_create( inv->pool, "media", &media_thread, audio, - 0, 0, &audio->thread); - if (status != PJ_SUCCESS) { - app_perror(THIS_FILE, "Error creating media thread", status); - return; - } - /* Set the media as active */ audio->active = PJ_TRUE; + + /* Immediately schedule to send the first RTP packet. */ + audio->rtp_timer.id = 1; + interval.sec = interval.msec = 0; + pjsip_endpt_schedule_timer(app.sip_endpt, &audio->rtp_timer, &interval); + + /* And schedule the first RTCP packet */ + audio->rtcp_timer.id = 1; + interval.sec = 4; + interval.msec = (pj_rand() % 1000); + pjsip_endpt_schedule_timer(app.sip_endpt, &audio->rtcp_timer, &interval); } @@ -1344,15 +1314,19 @@ static void destroy_call_media(unsigned call_index) { struct media_stream *audio = &app.call[call_index].media[0]; - if (audio->thread) { + if (audio->active) { audio->active = PJ_FALSE; - audio->thread_quit_flag = 1; - pj_thread_join(audio->thread); - pj_thread_destroy(audio->thread); - audio->thread = NULL; - audio->thread_quit_flag = 0; + if (audio->rtp_timer.id) { + audio->rtp_timer.id = 0; + pjsip_endpt_cancel_timer(app.sip_endpt, &audio->rtp_timer); + } + + if (audio->rtcp_timer.id) { + audio->rtcp_timer.id = 0; + pjsip_endpt_cancel_timer(app.sip_endpt, &audio->rtcp_timer); + } pjmedia_transport_detach(audio->transport, audio); } @@ -1694,8 +1668,8 @@ int main(int argc, char *argv[]) /* Shutting down... */ - destroy_sip(); destroy_media(); + destroy_sip(); if (app.pool) { pj_pool_release(app.pool); |