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Code Issues Releases Wiki Activity

Free rtc publish. Refactor TAB to spaces

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This commit is contained in:
winlin 2020-04-25 22:30:55 +08:00
parent cd3c15ec4e
commit ebdc03416a
6 changed files with 46 additions and 155 deletions
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107
trunk/research/players/rtc_publisher.html
View file

@ -149,111 +149,4 @@
</body>
</html>
<script>
/*
var PeerConnection = window.RTCPeerConnection || window.mozRTCPeerConnection || window.webkitRTCPeerConnection;
var SessionDescription = window.RTCSessionDescription || window.mozRTCSessionDescription || window.webkitRTCSessionDescription;
var url = document.location.protocol + "//" + document.domain + ":1985/rtc/v1/publish/";
var method = "POST";
var shouldBeAsync = true;
var request = new XMLHttpRequest();
request.open(method, url, shouldBeAsync);
request.setRequestHeader("Content-Type", "application/json;charset=UTF-8");
var pc = new PeerConnection();
var sendViewerOfferFn = function(desc) {
console.log('sendViewerOfferFn:', desc);
pc.setLocalDescription(desc);
var sdp_json = {"sdp":desc.sdp, "app":"live", "stream":"livestream", "streamurl":"webrtc://localhost/live/livestream"};
request.send(JSON.stringify(sdp_json));
};
pc.addTransceiver("audio", {
direction: "sendonly"
}
);
pc.addTransceiver("video", {
direction: "sendonly"
}
);
var constraints = {
"audio": true,
"video": {
"width": {
"min": "480",
"max": "1920"
},
"height": {
"min": "320",
"max": "1080"
},
"frameRate": {
"min": "15",
"max": "60"
}
}
};
navigator.mediaDevices.getUserMedia(constraints).then(
function(mediaStream) {
var video = document.getElementById('rtc_media_player');
video.srcObject = mediaStream;
video.onloadedmetadata = function(e) {
console.log('play');
video.play();
};
pc.addStream(mediaStream);
pc.createOffer(sendViewerOfferFn,
function(error) {
console.log('sendViewerOfferFn error:' + error);
},
);
}).catch(
function(err) {
console.log(err);
console.log(err.name + ": " + err.message);
}
);
pc.onicecandidate = function(event) {
console.log('onicecandidate');
};
pc.onconnectionstatechange = function(event) {
console.log('onconnectionstatechange');
};
pc.onicegatheringstatechange = function(event) {
console.log('onicegatheringstatechange');
};
pc.onsignalingstatechange = function(event) {
console.log('onsignalingstatechange');
};
request.onerror = function(event) {
console.log('http error');
};
request.onload = function () {
console.log('onload,' , request.responseText);
var json = JSON.parse(request.responseText);
console.log('onmessage viewerResponse:', json.sdp);
pc.setRemoteDescription(new SessionDescription({type:'answer', sdp:json.sdp}));
}
/**/
</script>
</html>

48
trunk/src/app/srs_app_rtc_conn.cpp
View file

@ -2150,7 +2150,7 @@ SrsRtcSession::SrsRtcSession(SrsRtcServer* rtc_svr, const SrsRequest& req, const
dtls_session = new SrsDtlsSession(this);
// TODO: FIXME: Check error.
dtls_session->initialize(req);
strd = NULL;
sender = NULL;
username = un;
@ -2165,17 +2165,14 @@ SrsRtcSession::SrsRtcSession(SrsRtcServer* rtc_svr, const SrsRequest& req, const
// TODO: FIXME: Support reload.
sessionStunTimeout = _srs_config->get_rtc_stun_timeout(req.vhost);
rtc_publisher = NULL;
publisher = NULL;
}
SrsRtcSession::~SrsRtcSession()
{
srs_freep(sender);
srs_freep(publisher);
srs_freep(dtls_session);
if (strd) {
strd->stop();
}
srs_freep(strd);
}
void SrsRtcSession::set_local_sdp(const SrsSdp& sdp)
@ -2199,19 +2196,19 @@ srs_error_t SrsRtcSession::on_stun(SrsUdpMuxSocket* skt, SrsStunPacket* stun_req
last_stun_time = srs_get_system_time();
if (strd && strd->sendonly_ukt) {
if (sender && sender->sendonly_ukt) {
// We are running in the ice-lite(server) mode. If client have multi network interface,
// we only choose one candidate pair which is determined by client.
if (stun_req->get_use_candidate() && strd->sendonly_ukt->get_peer_id() != skt->get_peer_id()) {
strd->update_sendonly_socket(skt);
if (stun_req->get_use_candidate() && sender->sendonly_ukt->get_peer_id() != skt->get_peer_id()) {
sender->update_sendonly_socket(skt);
}
}
if (rtc_publisher && rtc_publisher->sendonly_ukt) {
if (publisher && publisher->sendonly_ukt) {
// We are running in the ice-lite(server) mode. If client have multi network interface,
// we only choose one candidate pair which is determined by client.
if (stun_req->get_use_candidate() && rtc_publisher->sendonly_ukt->get_peer_id() != skt->get_peer_id()) {
rtc_publisher->update_sendonly_socket(skt);
if (stun_req->get_use_candidate() && publisher->sendonly_ukt->get_peer_id() != skt->get_peer_id()) {
publisher->update_sendonly_socket(skt);
}
}
}
@ -2428,20 +2425,20 @@ srs_error_t SrsRtcSession::on_rtcp_ps_feedback(char* buf, int nb_buf, SrsUdpMuxS
srs_error_t SrsRtcSession::on_rtcp_xr(char* buf, int nb_buf, SrsUdpMuxSocket* skt)
{
if (rtc_publisher == NULL) {
if (publisher == NULL) {
return srs_error_new(ERROR_RTC_RTCP, "rtc publisher null");
}
return rtc_publisher->on_rtcp_xr(buf, nb_buf, skt);
return publisher->on_rtcp_xr(buf, nb_buf, skt);
}
srs_error_t SrsRtcSession::on_rtcp_sender_report(char* buf, int nb_buf, SrsUdpMuxSocket* skt)
{
if (rtc_publisher == NULL) {
if (publisher == NULL) {
return srs_error_new(ERROR_RTC_RTCP, "rtc publisher null");
}
return rtc_publisher->on_rtcp_sender_report(buf, nb_buf, skt);
return publisher->on_rtcp_sender_report(buf, nb_buf, skt);
}
srs_error_t SrsRtcSession::on_rtcp_receiver_report(char* buf, int nb_buf, SrsUdpMuxSocket* skt)
@ -2540,8 +2537,8 @@ srs_error_t SrsRtcSession::start_play(SrsUdpMuxSocket* skt)
{
srs_error_t err = srs_success;
srs_freep(strd);
strd = new SrsRtcSenderThread(this, skt, _srs_context->get_id());
srs_freep(sender);
sender = new SrsRtcSenderThread(this, skt, _srs_context->get_id());
uint32_t video_ssrc = 0;
uint32_t audio_ssrc = 0;
@ -2558,11 +2555,11 @@ srs_error_t SrsRtcSession::start_play(SrsUdpMuxSocket* skt)
}
}
if ((err =strd->initialize(video_ssrc, audio_ssrc, video_payload_type, audio_payload_type)) != srs_success) {
if ((err = sender->initialize(video_ssrc, audio_ssrc, video_payload_type, audio_payload_type)) != srs_success) {
return srs_error_wrap(err, "SrsRtcSenderThread init");
}
if ((err = strd->start()) != srs_success) {
if ((err = sender->start()) != srs_success) {
return srs_error_wrap(err, "start SrsRtcSenderThread");
}
@ -2573,7 +2570,8 @@ srs_error_t SrsRtcSession::start_publish(SrsUdpMuxSocket* skt)
{
srs_error_t err = srs_success;
rtc_publisher = new SrsRtcPublisher(this);
srs_freep(publisher);
publisher = new SrsRtcPublisher(this);
uint32_t video_ssrc = 0;
uint32_t audio_ssrc = 0;
@ -2591,7 +2589,7 @@ srs_error_t SrsRtcSession::start_publish(SrsUdpMuxSocket* skt)
}
// FIXME: err process.
if ((err = rtc_publisher->initialize(skt, video_ssrc, audio_ssrc, request)) != srs_success) {
if ((err = publisher->initialize(skt, video_ssrc, audio_ssrc, request)) != srs_success) {
return srs_error_wrap(err, "rtc publisher init");
}
@ -2687,7 +2685,7 @@ srs_error_t SrsRtcSession::on_rtp(SrsUdpMuxSocket* skt)
{
srs_error_t err = srs_success;
if (rtc_publisher == NULL) {
if (publisher == NULL) {
return srs_error_new(ERROR_RTC_RTCP, "rtc publisher null");
}
@ -2701,7 +2699,7 @@ srs_error_t SrsRtcSession::on_rtp(SrsUdpMuxSocket* skt)
return srs_error_wrap(err, "rtp unprotect failed");
}
return rtc_publisher->on_rtp(skt, unprotected_buf, nb_unprotected_buf);
return publisher->on_rtp(skt, unprotected_buf, nb_unprotected_buf);
}
SrsUdpMuxSender::SrsUdpMuxSender(SrsRtcServer* s)

4
trunk/src/app/srs_app_rtc_conn.hpp
View file

@ -312,7 +312,7 @@ private:
SrsSdp local_sdp;
SrsRtcSessionStateType session_state;
SrsDtlsSession* dtls_session;
SrsRtcSenderThread* strd;
SrsRtcSenderThread* sender;
std::string username;
std::string peer_id;
srs_utime_t last_stun_time;
@ -330,7 +330,7 @@ public:
SrsRequest request;
SrsSource* source;
private:
SrsRtcPublisher* rtc_publisher;
SrsRtcPublisher* publisher;
public:
SrsRtcSession(SrsRtcServer* rtc_svr, const SrsRequest& req, const std::string& un, int context_id);
virtual ~SrsRtcSession();