external/srs
1
0
Fork 0
mirror of https://github.com/ossrs/srs.git synced 2025-02-13 20:01:56 +00:00
Code Issues Releases Wiki Activity

Merge branch 'feature/gb28181' into develop

Browse source
This commit is contained in:
winlin 2020-08-30 12:15:58 +08:00
commit c8eb667a04
4 changed files with 85 additions and 25 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

75
trunk/src/app/srs_app_gb28181.cpp
View file

@ -412,7 +412,7 @@ srs_error_t SrsGb28181PsRtpProcessor::rtmpmuxer_enqueue_data(SrsGb28181RtmpMuxer
muxer->get_channel_id().c_str(), ssrc, muxer->channel_peer_port(), peer_port);
}else {
//muxer->ps_packet_enqueue(pkt);
muxer->insert_jitterbuffer(pkt);
muxer->insert_jitterbuffer(pkt);
}//end if (muxer->channel_peer_port() != peer_port)
}//end if (muxer)
@ -593,19 +593,6 @@ int64_t SrsPsStreamDemixer::parse_ps_timestamp(const uint8_t* p)
return val;
}
bool SrsPsStreamDemixer::is_aac(){
// SrsBuffer *avs = new SrsBuffer(stream->bytes(), stream->length());
// SrsAutoFree(SrsBuffer, avs);
// if (!avs->empty()) {
// char* frame = NULL;
// int frame_size = 0;
// SrsRawAacStreamCodec codec;
// if ((err = aac->adts_demux(avs, &frame, &frame_size, codec)) != srs_success) {
// return srs_error_wrap(err, "demux adts");
// }
return true;
}
srs_error_t SrsPsStreamDemixer::on_ps_stream(char* ps_data, int ps_size, uint32_t timestamp, uint32_t ssrc)
{
srs_error_t err = srs_success;
@ -976,8 +963,13 @@ SrsGb28181RtmpMuxer::SrsGb28181RtmpMuxer(SrsGb28181Manger* c, std::string id, bo
source_publish = true;
jitter_buffer = new SrsPsJitterBuffer(id);
jitter_buffer_audio = new SrsPsJitterBuffer(id);
ps_buflen = 0;
ps_buffer = NULL;
ps_buflen_auido = 0;
ps_buffer_audio = NULL;
}
SrsGb28181RtmpMuxer::~SrsGb28181RtmpMuxer()
@ -988,7 +980,10 @@ SrsGb28181RtmpMuxer::~SrsGb28181RtmpMuxer()
srs_cond_destroy(wait_ps_queue);
srs_freep(jitter_buffer);
srs_freep(jitter_buffer_audio);
srs_freepa(ps_buffer);
srs_freepa(ps_buffer_audio);
srs_freep(channel);
srs_freep(ps_demixer);
srs_freep(trd);
@ -1094,6 +1089,14 @@ srs_error_t SrsGb28181RtmpMuxer::initialize(SrsServer *s, SrsRequest* r)
jitter_buffer->SetNackMode(kNack, -1, -1);
jitter_buffer->SetNackSettings(250, 450, 0);
if (!jitter_buffer_audio) {
jitter_buffer_audio = new SrsPsJitterBuffer(channel_id);
}
jitter_buffer_audio->SetDecodeErrorMode(kSelectiveErrors);
jitter_buffer_audio->SetNackMode(kNack, -1, -1);
jitter_buffer_audio->SetNackSettings(250, 450, 0);
if (!source_publish) return err;
req = r;
@ -1146,6 +1149,17 @@ srs_error_t SrsGb28181RtmpMuxer::do_cycle()
};
}
}
if(jitter_buffer_audio->FoundFrame(cur_timestamp)){
jitter_buffer_audio->GetPsFrame(&ps_buffer_audio, ps_buflen_auido, buffer_size, cur_timestamp);
if (buffer_size > 0){
if ((err = ps_demixer->on_ps_stream(ps_buffer_audio, buffer_size, cur_timestamp, 0)) != srs_success){
srs_warn("gb28181: demix ps stream error:%s", srs_error_desc(err).c_str());
srs_freep(err);
};
}
}
}else {
//demix ps to h264/aac, to rtmp
while(!ps_queue.empty()){
@ -1224,8 +1238,39 @@ void SrsGb28181RtmpMuxer::stop()
void SrsGb28181RtmpMuxer::insert_jitterbuffer(SrsPsRtpPacket *pkt)
{
if (!pkt){
return;
}
recv_rtp_stream_time = srs_get_system_time();
jitter_buffer->InsertPacket(*pkt, pkt->payload->bytes(), pkt->payload->length(), NULL);
char *payload = pkt->payload->bytes();
uint8_t p1 = (uint8_t)(payload[0]);
uint8_t p2 = (uint8_t)(payload[1]);
uint8_t p3 = (uint8_t)(payload[2]);
uint8_t p4 = (uint8_t)(payload[3]);
//check for rtp ps audio streaming
bool av_same_ts = true;
if (p1 == 0x00 && p2 == 0x00 && p3 == 0x01 && p4 == 0xC0 &&
ps_rtp_video_ts != pkt->timestamp) {
av_same_ts = false;
}
//if audio and video are the same clock,
//if both audio and video use jitter_buffer,
//otherwise audio uses jitter_buffer_audio, and video uses jitter_buffer
if (av_same_ts){
pkt->marker = false;
jitter_buffer->InsertPacket(*pkt, pkt->payload->bytes(), pkt->payload->length(), NULL);
ps_rtp_video_ts = pkt->timestamp;
}else {
jitter_buffer_audio->InsertPacket(*pkt, pkt->payload->bytes(), pkt->payload->length(), NULL);
}
//srs_cond_signal(wait_ps_queue);
}

9
trunk/src/app/srs_app_gb28181.hpp
View file

@ -260,7 +260,6 @@ private:
public:
int64_t parse_ps_timestamp(const uint8_t* p);
std::string get_ps_map_type_str(uint8_t);
bool is_aac();
virtual srs_error_t on_ps_stream(char* ps_data, int ps_size, uint32_t timestamp, uint32_t ssrc);
};
@ -303,8 +302,16 @@ private:
SrsServer* server;
SrsPsJitterBuffer *jitter_buffer;
SrsPsJitterBuffer *jitter_buffer_audio;
char *ps_buffer;
char *ps_buffer_audio;
int ps_buflen;
int ps_buflen_auido;
uint32_t ps_rtp_video_ts;
uint32_t ps_rtp_audio_ts;
bool source_publish;

10
trunk/src/app/srs_app_gb28181_jitbuffer.cpp
View file

@ -306,11 +306,11 @@ PsFrameBufferEnum SrsPsFrameBuffer::InsertPacket(const VCMPacket& packet, const
}
//TODO: not check marker, check a complete frame with timestamp
// if (packet.markerBit &&
// (last_packet_seq_num_ == -1 ||
// IsNewerSequenceNumber(packet.seqNum, last_packet_seq_num_))) {
// last_packet_seq_num_ = packet.seqNum;
// }
if (packet.markerBit &&
(last_packet_seq_num_ == -1 ||
IsNewerSequenceNumber(packet.seqNum, last_packet_seq_num_))) {
last_packet_seq_num_ = packet.seqNum;
}
// The insert operation invalidates the iterator |rit|.
PacketIterator packet_list_it = packets_.insert(rit.base(), packet);

16
trunk/src/app/srs_app_gb28181_sip.cpp
View file

@ -163,7 +163,8 @@ srs_error_t SrsGb28181SipSession::do_cycle()
if (_register_status == SrsGb28181SipSessionRegisterOk &&
_alive_status == SrsGb28181SipSessionAliveOk)
{
std::list<std::string> auto_play_list;
std::map<std::string, SrsGb28181Device*>::iterator it;
for (it = _device_list.begin(); it != _device_list.end(); it++) {
SrsGb28181Device *device = it->second;
@ -191,8 +192,15 @@ srs_error_t SrsGb28181SipSession::do_cycle()
//offline or already invite device does not need to send invite
if (device->device_status != "ON" ||
device->invite_status != SrsGb28181SipSessionUnkonw) continue;
auto_play_list.push_back(chid);
}//end for (it)
//auto send sip invite and create stream chennal
while(auto_play_list.size() > 0){
std::string chid = auto_play_list.front();
auto_play_list.pop_front();
SrsGb28181StreamChannel ch;
ch.set_channel_id(_session_id + "@" + chid);
@ -226,7 +234,7 @@ srs_error_t SrsGb28181SipSession::do_cycle()
srs_trace("gb28181: %s clients device=%s send invite code=%d",
_session_id.c_str(), chid.c_str(), code);
}//end for (it)
}//end while (auto_play_list.size())
}//end if (config)
if (_register_status == SrsGb28181SipSessionRegisterOk &&
@ -911,7 +919,7 @@ srs_error_t SrsGb28181SipService::fetch_or_create_sip_session(SrsSipRequest *req
if ((sess = fetch(req->sip_auth_id)) != NULL) {
*sip_session = sess;
return err;
}
}
sess = new SrsGb28181SipSession(this, req);;
if ((err = sess->serve()) != srs_success) {