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rtc session distinguish play and publish

Browse Source
pull/1753/head
xiaozhihong 5 years ago
parent 1951cc8e5f
commit 775065175a
4 changed files with 82 additions and 39 deletions
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22
trunk/src/app/srs_app_http_api.cpp
Unescape Escape View File

@ -1086,11 +1086,13 @@ srs_error_t SrsGoApiRtcPlay::exchange_sdp(const std::string& app, const std::str
local_media_desc.rtcp_mux_ = true;
local_media_desc.rtcp_rsize_ = true;
SrsSSRCInfo ssrc_info;
ssrc_info.ssrc_ = ++ssrc_num;
// TODO:use formated cname
ssrc_info.cname_ = "test_sdp_cname";
local_media_desc.ssrc_infos_.push_back(ssrc_info);
if (local_media_desc.recvonly_ || local_media_desc.sendrecv_) {
SrsSSRCInfo ssrc_info;
ssrc_info.ssrc_ = ++ssrc_num;
// TODO:use formated cname
ssrc_info.cname_ = "test_sdp_cname";
local_media_desc.ssrc_infos_.push_back(ssrc_info);
}
}
return err;
@ -1381,10 +1383,12 @@ srs_error_t SrsGoApiRtcPublish::exchange_sdp(const std::string& app, const std::
local_media_desc.rtcp_mux_ = true;
SrsSSRCInfo ssrc_info;
ssrc_info.ssrc_ = ++ssrc_num;
ssrc_info.cname_ = "test_sdp_cname";
local_media_desc.ssrc_infos_.push_back(ssrc_info);
if (local_media_desc.recvonly_ || local_media_desc.sendrecv_) {
SrsSSRCInfo ssrc_info;
ssrc_info.ssrc_ = ++ssrc_num;
ssrc_info.cname_ = "test_sdp_cname";
local_media_desc.ssrc_infos_.push_back(ssrc_info);
}
}
return err;

90
trunk/src/app/srs_app_rtc_conn.cpp
Unescape Escape View File

@ -1614,6 +1614,10 @@ srs_error_t SrsRtcPublisher::on_rtcp_sender_report(char* buf, int nb_buf, SrsUdp
{
srs_error_t err = srs_success;
if (nb_buf < 28) {
return srs_error_new(ERROR_RTC_RTCP_CHECK, "invalid rtp sender report packet, nb_buf=%d", nb_buf);
}
SrsBuffer* stream = new SrsBuffer(buf, nb_buf);
SrsAutoFree(SrsBuffer, stream);
@ -2100,7 +2104,7 @@ SrsRtcSession::SrsRtcSession(SrsRtcServer* rtc_svr, const SrsRequest& req, const
// TODO: FIXME: Support reload.
sessionStunTimeout = _srs_config->get_rtc_stun_timeout(req.vhost);
rtc_publisher = new SrsRtcPublisher(this);
rtc_publisher = NULL;
}
SrsRtcSession::~SrsRtcSession()
@ -2362,6 +2366,10 @@ 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) {
return srs_error_new(ERROR_RTC_RTCP, "rtc publisher null");
}
return rtc_publisher->on_rtcp_xr(buf, nb_buf, skt);
}
@ -2369,8 +2377,8 @@ srs_error_t SrsRtcSession::on_rtcp_sender_report(char* buf, int nb_buf, SrsUdpMu
{
srs_error_t err = srs_success;
if (nb_buf < 28) {
return srs_error_new(ERROR_RTC_RTCP_CHECK, "invalid rtp sender report packet, nb_buf=%d", nb_buf);
if (rtc_publisher == NULL) {
return srs_error_new(ERROR_RTC_RTCP, "rtc publisher null");
}
return rtc_publisher->on_rtcp_sender_report(buf, nb_buf, skt);
@ -2447,34 +2455,25 @@ block +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
srs_error_t SrsRtcSession::on_connection_established(SrsUdpMuxSocket* skt)
{
// FIXME:
if (true)
{
uint32_t video_ssrc = 0;
uint32_t audio_ssrc = 0;
uint16_t video_payload_type = 0;
uint16_t audio_payload_type = 0;
for (size_t i = 0; i < remote_sdp.media_descs_.size(); ++i) {
const SrsMediaDesc& media_desc = remote_sdp.media_descs_[i];
if (media_desc.is_audio()) {
if (! media_desc.ssrc_infos_.empty()) {
audio_ssrc = media_desc.ssrc_infos_[0].ssrc_;
audio_payload_type = media_desc.payload_types_[0].payload_type_;
}
} else if (media_desc.is_video()) {
if (! media_desc.ssrc_infos_.empty()) {
video_ssrc = media_desc.ssrc_infos_[0].ssrc_;
video_payload_type = media_desc.payload_types_[0].payload_type_;
}
}
srs_error_t err = srs_success;
srs_trace("rtc session=%s, to=%dms connection established", id().c_str(), srsu2msi(sessionStunTimeout));
if (! local_sdp.media_descs_.empty() &&
(local_sdp.media_descs_.back().recvonly_ || local_sdp.media_descs_.back().sendrecv_)) {
if ((err = start_publish(skt)) != srs_success) {
return srs_error_wrap(err, "start publish");
}
}
// FIXME: err process.
rtc_publisher->initialize(skt, video_ssrc, audio_ssrc, request);
if (! local_sdp.media_descs_.empty() &&
(local_sdp.media_descs_.back().sendonly_ || local_sdp.media_descs_.back().sendrecv_)) {
if ((err = start_play(skt)) != srs_success) {
return srs_error_wrap(err, "start play");
}
}
srs_trace("rtc session=%s, to=%dms connection established", id().c_str(), srsu2msi(sessionStunTimeout));
return start_play(skt);
return err;
}
srs_error_t SrsRtcSession::start_play(SrsUdpMuxSocket* skt)
@ -2510,6 +2509,39 @@ srs_error_t SrsRtcSession::start_play(SrsUdpMuxSocket* skt)
return err;
}
srs_error_t SrsRtcSession::start_publish(SrsUdpMuxSocket* skt)
{
srs_error_t err = srs_success;
rtc_publisher = new SrsRtcPublisher(this);
uint32_t video_ssrc = 0;
uint32_t audio_ssrc = 0;
uint16_t video_payload_type = 0;
uint16_t audio_payload_type = 0;
for (size_t i = 0; i < remote_sdp.media_descs_.size(); ++i) {
const SrsMediaDesc& media_desc = remote_sdp.media_descs_[i];
if (media_desc.is_audio()) {
if (! media_desc.ssrc_infos_.empty()) {
audio_ssrc = media_desc.ssrc_infos_[0].ssrc_;
audio_payload_type = media_desc.payload_types_[0].payload_type_;
}
} else if (media_desc.is_video()) {
if (! media_desc.ssrc_infos_.empty()) {
video_ssrc = media_desc.ssrc_infos_[0].ssrc_;
video_payload_type = media_desc.payload_types_[0].payload_type_;
}
}
}
// FIXME: err process.
if ((err = rtc_publisher->initialize(skt, video_ssrc, audio_ssrc, request)) != srs_success) {
return srs_error_wrap(err, "rtc publisher init");
}
return err;
}
bool SrsRtcSession::is_stun_timeout()
{
return last_stun_time + sessionStunTimeout < srs_get_system_time();
@ -2599,6 +2631,10 @@ srs_error_t SrsRtcSession::on_rtp(SrsUdpMuxSocket* skt)
{
srs_error_t err = srs_success;
if (rtc_publisher == NULL) {
return srs_error_new(ERROR_RTC_RTCP, "rtc publisher null");
}
if (dtls_session == NULL) {
return srs_error_new(ERROR_RTC_RTCP, "recv unexpect rtp packet before dtls done");
}

1
trunk/src/app/srs_app_rtc_conn.hpp
Unescape Escape View File

@ -358,6 +358,7 @@ public:
srs_error_t send_client_hello(SrsUdpMuxSocket* skt);
srs_error_t on_connection_established(SrsUdpMuxSocket* skt);
srs_error_t start_play(SrsUdpMuxSocket* skt);
srs_error_t start_publish(SrsUdpMuxSocket* skt);
public:
bool is_stun_timeout();
private:

8
trunk/src/app/srs_app_rtp_queue.cpp
Unescape Escape View File

@ -127,8 +127,8 @@ void SrsRtpNackList::get_nack_seqs(vector<uint16_t>& seqs)
void SrsRtpNackList::update_rtt(int rtt)
{
rtt_ = rtt;
srs_verbose("NACK, update rtt from %ld to %d", opts_.nack_interval, rtt);
rtt_ = rtt * SRS_UTIME_MILLISECONDS;
srs_verbose("NACK, update rtt from %ld to %d", opts_.nack_interval, rtt_);
// FIXME: limit min and max value.
opts_.nack_interval = rtt_;
}
@ -182,7 +182,9 @@ srs_error_t SrsRtpQueue::insert(SrsRtpSharedPacket* rtp_pkt)
} else {
SrsRtpNackInfo* nack_info = NULL;
if ((nack_info = nack_.find(seq)) != NULL) {
srs_verbose("seq=%u, alive time=%d, nack count=%d, rtx success", seq, now - nack_info->generate_time_, nack_info->req_nack_count_);
int nack_rtt = nack_info->req_nack_count_ ? ((now - nack_info->pre_req_nack_time_) / SRS_UTIME_MILLISECONDS) : 0;
srs_verbose("seq=%u, alive time=%d, nack count=%d, rtx success, resend use %dms",
seq, now - nack_info->generate_time_, nack_info->req_nack_count_, nack_rtt);
nack_.remove(seq);
} else {
// Calc jitter.

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