Undo modification to use timer, and back to using threads as the performance is not good

git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@615 74dad513-b988-da41-8d7b-12977e46ad98
This commit is contained in:
Benny Prijono 2006-07-18 21:12:24 +00:00
parent 399188f44c
commit 513795f798
1 changed files with 185 additions and 161 deletions

View File

@ -122,9 +122,9 @@ struct media_stream
/* RTCP stats: */
pjmedia_rtcp_session rtcp; /* incoming RTCP session. */
/* 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. */
/* Thread: */
pj_bool_t thread_quit_flag; /* Stop media thread. */
pj_thread_t *thread; /* Media thread. */
};
@ -222,15 +222,6 @@ 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);
@ -396,9 +387,7 @@ 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,
pjsip_endpt_get_ioqueue(app.sip_endpt), 0,
&app.med_endpt);
status = pjmedia_endpt_create(&app.cp.factory, NULL, 1, &app.med_endpt);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
@ -418,18 +407,10 @@ 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;
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;
app.call[i].media[j].call_index = i;
app.call[i].media[j].media_index = j;
status = -1;
for (retry=0; retry<100; ++retry,rtp_port+=2) {
@ -587,34 +568,32 @@ static void process_incoming_call(pjsip_rx_data *rdata)
return;
}
call = &app.call[i];
/* Verify that we can handle the request. */
options = 0;
status = pjsip_inv_verify_request(rdata, &options, NULL, NULL,
app.sip_endpt, &tdata);
app.sip_endpt, &tdata);
if (status != PJ_SUCCESS) {
/*
* No we can't handle the incoming INVITE request.
*/
if (tdata) {
pjsip_response_addr res_addr;
pjsip_get_response_addr(tdata->pool, rdata, &res_addr);
pjsip_endpt_send_response(app.sip_endpt, &res_addr, tdata,
NULL, NULL);
pjsip_endpt_send_response(app.sip_endpt, &res_addr, tdata,
NULL, NULL);
} else {
/* Respond with 500 (Internal Server Error) */
pjsip_endpt_respond_stateless(app.sip_endpt, rdata, 500, NULL,
NULL, NULL);
NULL, NULL);
}
return;
}
call = &app.call[i];
}
/* Create UAS dialog */
status = pjsip_dlg_create_uas( pjsip_ua_instance(), rdata,
@ -802,28 +781,13 @@ 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, 1};
pj_time_val timeout = {0, 10};
pjsip_endpt_handle_events(app.sip_endpt, &timeout);
}
@ -1082,6 +1046,19 @@ 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.
*/
@ -1125,62 +1102,6 @@ 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_bzero(packet+hdrlen, 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.
*/
@ -1205,40 +1126,150 @@ 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 RTCP packet. */
static void on_tx_rtcp(pj_timer_heap_t *timer_heap,
struct pj_timer_entry *entry)
/*
* Media thread
*
* This is the thread to send and receive both RTP and RTCP packets.
*/
static int media_thread(void *arg)
{
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;
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;
PJ_UNUSED_ARG(timer_heap);
if (!strm->active)
return;
/* Boost thread priority if necessary */
boost_priority();
/* Build RTCP packet */
pjmedia_rtcp_build_rtcp(&strm->rtcp, &rtcp_pkt, &rtcp_len);
/* 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_bzero(packet+hdrlen, 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);
}
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;
/* 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 */
interval.sec = 5;
interval.msec = (pj_rand() % 500);
pjsip_endpt_schedule_timer(app.sip_endpt, &strm->rtcp_timer, &interval);
/* 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);
}
}
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)
@ -1248,7 +1279,6 @@ 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];
@ -1256,7 +1286,7 @@ 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->active)
if (audio->thread != NULL)
destroy_call_media(call->index);
@ -1311,7 +1341,7 @@ static void call_on_media_update( pjsip_inv_session *inv,
/* Attach media to transport */
status = pjmedia_transport_attach(audio->transport, audio,
&audio->si.rem_addr,
&audio->si.rem_rtcp,
&audio->si.rem_rtcp,
sizeof(pj_sockaddr_in),
&on_rx_rtp,
&on_rx_rtcp);
@ -1320,19 +1350,17 @@ 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);
}
@ -1342,25 +1370,21 @@ static void destroy_call_media(unsigned call_index)
{
struct media_stream *audio = &app.call[call_index].media[0];
if (audio->active) {
if (audio->thread) {
audio->active = PJ_FALSE;
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);
}
audio->thread_quit_flag = 1;
pj_thread_join(audio->thread);
pj_thread_destroy(audio->thread);
audio->thread = NULL;
audio->thread_quit_flag = 0;
pjmedia_transport_detach(audio->transport, audio);
}
}
/*****************************************************************************
* USER INTERFACE STUFFS
*/
@ -1696,8 +1720,8 @@ int main(int argc, char *argv[])
/* Shutting down... */
destroy_media();
destroy_sip();
destroy_media();
if (app.pool) {
pj_pool_release(app.pool);