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for #133, alloc and free rtp port.

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This commit is contained in:
winlin 2015-02-17 21:10:06 +08:00
parent d4ceff649f
commit f14af45413
9 changed files with 183 additions and 38 deletions
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101
trunk/src/app/srs_app_rtsp.cpp
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@ -36,21 +36,41 @@ using namespace std;
#ifdef SRS_AUTO_STREAM_CASTER
ISrsRtspHandler::ISrsRtspHandler()
SrsRtpConn::SrsRtpConn(SrsRtspConn* r, int p)
{
rtsp = r;
_port = p;
listener = new SrsUdpListener(this, p);
}
ISrsRtspHandler::~ISrsRtspHandler()
SrsRtpConn::~SrsRtpConn()
{
srs_freep(listener);
}
SrsRtspConn::SrsRtspConn(SrsRtspCaster* c, st_netfd_t fd, std::string o, int lpmin, int lpmax)
int SrsRtpConn::port()
{
return _port;
}
int SrsRtpConn::listen()
{
return listener->listen();
}
int SrsRtpConn::on_udp_packet(sockaddr_in* from, char* buf, int nb_buf)
{
int ret = ERROR_SUCCESS;
return ret;
}
SrsRtspConn::SrsRtspConn(SrsRtspCaster* c, st_netfd_t fd, std::string o)
{
output = o;
local_port_min = lpmin;
local_port_max = lpmax;
session = "O9EaZ4bf"; // TODO: FIXME: generate session id.
video_rtp = NULL;
audio_rtp = NULL;
caster = c;
stfd = fd;
@ -64,6 +84,9 @@ SrsRtspConn::~SrsRtspConn()
srs_close_stfd(stfd);
trd->stop();
srs_freep(video_rtp);
srs_freep(audio_rtp);
srs_freep(trd);
srs_freep(skt);
srs_freep(rtsp);
@ -103,6 +126,8 @@ int SrsRtspConn::do_cycle()
}
} else if (req->is_announce()) {
srs_assert(req->sdp);
video_id = req->sdp->video_stream_id;
audio_id = req->sdp->audio_stream_id;
sps = req->sdp->video_sps;
pps = req->sdp->video_pps;
asc = req->sdp->audio_sh;
@ -119,11 +144,31 @@ int SrsRtspConn::do_cycle()
}
} else if (req->is_setup()) {
srs_assert(req->transport);
int lpm = 0;
if ((ret = caster->alloc_port(&lpm)) != ERROR_SUCCESS) {
srs_error("rtsp: alloc port failed. ret=%d", ret);
return ret;
}
SrsRtpConn* rtp = NULL;
if (req->stream_id == video_id) {
srs_freep(video_rtp);
rtp = video_rtp = new SrsRtpConn(this, lpm);
} else {
srs_freep(audio_rtp);
rtp = audio_rtp = new SrsRtpConn(this, lpm);
}
if ((ret = rtp->listen()) != ERROR_SUCCESS) {
srs_error("rtsp: rtp listen at port=%d failed. ret=%d", lpm, ret);
return ret;
}
srs_trace("rtsp: rtp listen at port=%d ok.", lpm);
SrsRtspSetupResponse* res = new SrsRtspSetupResponse(req->seq);
res->client_port_min = req->transport->client_port_min;
res->client_port_max = req->transport->client_port_max;
res->local_port_min = local_port_min;
res->local_port_max = local_port_max;
res->local_port_min = lpm;
res->local_port_max = lpm + 1;
res->session = session;
if ((ret = rtsp->send_message(res)) != ERROR_SUCCESS) {
if (!srs_is_client_gracefully_close(ret)) {
@ -165,6 +210,14 @@ int SrsRtspConn::cycle()
void SrsRtspConn::on_thread_stop()
{
if (video_rtp) {
caster->free_port(video_rtp->port(), video_rtp->port() + 1);
}
if (audio_rtp) {
caster->free_port(audio_rtp->port(), audio_rtp->port() + 1);
}
caster->remove(this);
}
@ -184,16 +237,40 @@ SrsRtspCaster::~SrsRtspCaster()
srs_freep(conn);
}
clients.clear();
used_ports.clear();
}
int SrsRtspCaster::serve_client(st_netfd_t stfd)
int SrsRtspCaster::alloc_port(int* pport)
{
int ret = ERROR_SUCCESS;
SrsRtspConn* conn = new SrsRtspConn(
this, stfd,
output, local_port_min, local_port_max
);
// use a pair of port.
for (int i = local_port_min; i < local_port_max - 1; i += 2) {
if (!used_ports[i]) {
used_ports[i] = true;
used_ports[i + 1] = true;
*pport = i;
break;
}
}
srs_info("rtsp: alloc port=%d-%d", *pport, *pport + 1);
return ret;
}
void SrsRtspCaster::free_port(int lpmin, int lpmax)
{
for (int i = lpmin; i < lpmax; i++) {
used_ports[i] = false;
}
srs_trace("rtsp: free rtp port=%d-%d", lpmin, lpmax);
}
int SrsRtspCaster::on_tcp_client(st_netfd_t stfd)
{
int ret = ERROR_SUCCESS;
SrsRtspConn* conn = new SrsRtspConn(this, stfd, output);
if ((ret = conn->serve()) != ERROR_SUCCESS) {
srs_error("rtsp: serve client failed. ret=%d", ret);