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try to improve autoflush and other silly audio glitches from edge cases and help FSCORE-416

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git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@14800 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Anthony Minessale committed 2009-09-10 00:46:24 +00:00
1 parent 56a8eb4091
commit 7392fa5807
5 files changed
+158 -151
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src/switch_rtp.c
+28 -9
View File
@@ -182,6 +182,7 @@ struct switch_rtp {
char *remote_host_str;
switch_time_t last_stun;
uint32_t samples_per_interval;
uint32_t samples_per_second;
uint32_t conf_samples_per_interval;
uint32_t rsamples_per_interval;
uint32_t ms_per_packet;
@@ -213,6 +214,7 @@ struct switch_rtp {
switch_rtp_bug_flag_t rtp_bugs;
switch_rtp_stats_t stats;
int hot_hits;
int sync_packets;
#ifdef ENABLE_ZRTP
zrtp_session_t *zrtp_session;
@@ -1038,12 +1040,21 @@ SWITCH_DECLARE(switch_status_t) switch_rtp_add_crypto_key(switch_rtp_t *rtp_sess
return SWITCH_STATUS_SUCCESS;
}
SWITCH_DECLARE(switch_status_t) switch_rtp_change_interval(switch_rtp_t *rtp_session, uint32_t ms_per_packet, uint32_t samples_per_interval)
SWITCH_DECLARE(switch_status_t) switch_rtp_set_interval(switch_rtp_t *rtp_session, uint32_t ms_per_packet, uint32_t samples_per_interval)
{
rtp_session->ms_per_packet = ms_per_packet;
rtp_session->samples_per_interval = rtp_session->conf_samples_per_interval = samples_per_interval;
rtp_session->missed_count = 0;
rtp_session->samples_per_second = (uint32_t)((double)(1000.0f / (double)(rtp_session->ms_per_packet / 1000)) * (double)rtp_session->samples_per_interval);
return SWITCH_STATUS_SUCCESS;
}
SWITCH_DECLARE(switch_status_t) switch_rtp_change_interval(switch_rtp_t *rtp_session, uint32_t ms_per_packet, uint32_t samples_per_interval)
{
switch_rtp_set_interval(rtp_session, ms_per_packet, samples_per_interval);
if (rtp_session->timer_name) {
if (rtp_session->timer.timer_interface) {
switch_core_timer_destroy(&rtp_session->timer);
@@ -1123,8 +1134,9 @@ SWITCH_DECLARE(switch_status_t) switch_rtp_create(switch_rtp_t **new_rtp_session
rtp_session->recv_msg.header.cc = 0;
rtp_session->payload = payload;
rtp_session->ms_per_packet = ms_per_packet;
rtp_session->samples_per_interval = rtp_session->conf_samples_per_interval = samples_per_interval;
switch_rtp_set_interval(rtp_session, ms_per_packet, samples_per_interval);
rtp_session->conf_samples_per_interval = samples_per_interval;
if (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER) && switch_strlen_zero(timer_name)) {
timer_name = "soft";
@@ -1490,11 +1502,6 @@ SWITCH_DECLARE(switch_socket_t *) switch_rtp_get_rtp_socket(switch_rtp_t *rtp_se
return rtp_session->sock_input;
}
SWITCH_DECLARE(void) switch_rtp_set_default_samples_per_interval(switch_rtp_t *rtp_session, uint32_t samples_per_interval)
{
rtp_session->samples_per_interval = samples_per_interval;
}
SWITCH_DECLARE(uint32_t) switch_rtp_get_default_samples_per_interval(switch_rtp_t *rtp_session)
{
return rtp_session->samples_per_interval;
@@ -1783,7 +1790,9 @@ static int rtp_common_read(switch_rtp_t *rtp_session, switch_payload_t *payload_
if ((switch_test_flag(rtp_session, SWITCH_RTP_FLAG_AUTOFLUSH) || switch_test_flag(rtp_session, SWITCH_RTP_FLAG_STICKY_FLUSH)) &&
rtp_session->read_pollfd) {
if (switch_poll(rtp_session->read_pollfd, 1, &fdr, 1) == SWITCH_STATUS_SUCCESS) {
if (++rtp_session->hot_hits >= 10) {
rtp_session->hot_hits += rtp_session->samples_per_interval;
if (rtp_session->hot_hits >= rtp_session->samples_per_second * 60) {
hot_socket = 1;
}
} else {
@@ -1792,8 +1801,17 @@ static int rtp_common_read(switch_rtp_t *rtp_session, switch_payload_t *payload_
}
if (hot_socket) {
rtp_session->sync_packets++;
switch_core_timer_sync(&rtp_session->timer);
} else {
if (rtp_session->sync_packets) {
#if 0
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,
"Auto-Flush catching up %d packets (%d)ms.\n",
rtp_session->sync_packets, rtp_session->ms_per_packet * rtp_session->sync_packets);
#endif
rtp_session->sync_packets = 0;
}
switch_core_timer_next(&rtp_session->timer);
}
}
@@ -2440,6 +2458,7 @@ static int rtp_common_write(switch_rtp_t *rtp_session,
rtp_session->ts = (uint32_t) timestamp;
} else if (rtp_session->timer.timer_interface) {
rtp_session->ts = rtp_session->timer.samplecount;
if (rtp_session->ts <= rtp_session->last_write_ts) {
rtp_session->ts = rtp_session->last_write_ts + rtp_session->samples_per_interval;
}