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srs/trunk/src/srt/srt_to_rtmp.cpp
2020-02-13 16:52:26 +08:00

556 lines
17 KiB
C++

#include "srt_to_rtmp.hpp"
#include "stringex.hpp"
#include "time_help.h"
#include <srs_kernel_log.hpp>
#include <srs_kernel_error.hpp>
#include <srs_kernel_buffer.hpp>
#include <srs_kernel_utility.hpp>
#include <srs_app_rtmp_conn.hpp>
#include <srs_app_config.hpp>
#include <srs_kernel_stream.hpp>
#include <list>
std::shared_ptr<srt2rtmp> srt2rtmp::s_srt2rtmp_ptr;
std::shared_ptr<srt2rtmp> srt2rtmp::get_instance() {
if (!s_srt2rtmp_ptr) {
s_srt2rtmp_ptr = std::make_shared<srt2rtmp>();
}
return s_srt2rtmp_ptr;
}
srt2rtmp::srt2rtmp():_lastcheck_ts(0) {
}
srt2rtmp::~srt2rtmp() {
release();
}
srs_error_t srt2rtmp::init() {
srs_error_t err = srs_success;
if (_trd_ptr.get() != nullptr) {
return srs_error_wrap(err, "don't start thread again");
}
_trd_ptr = std::make_shared<SrsSTCoroutine>("srt2rtmp", this);
if ((err = _trd_ptr->start()) != srs_success) {
return srs_error_wrap(err, "start thread");
}
srs_trace("srt2rtmp start coroutine...");
return err;
}
void srt2rtmp::release() {
if (!_trd_ptr) {
return;
}
_trd_ptr->stop();
_trd_ptr = nullptr;
}
void srt2rtmp::insert_data_message(unsigned char* data_p, unsigned int len, const std::string& key_path) {
std::unique_lock<std::mutex> locker(_mutex);
SRT_DATA_MSG_PTR msg_ptr = std::make_shared<SRT_DATA_MSG>(data_p, len, key_path);
_msg_queue.push(msg_ptr);
//_notify_cond.notify_one();
return;
}
void srt2rtmp::insert_ctrl_message(unsigned int msg_type, const std::string& key_path) {
std::unique_lock<std::mutex> locker(_mutex);
SRT_DATA_MSG_PTR msg_ptr = std::make_shared<SRT_DATA_MSG>(key_path, msg_type);
_msg_queue.push(msg_ptr);
//_notify_cond.notify_one();
return;
}
SRT_DATA_MSG_PTR srt2rtmp::get_data_message() {
std::unique_lock<std::mutex> locker(_mutex);
SRT_DATA_MSG_PTR msg_ptr;
if (_msg_queue.empty())
{
return msg_ptr;
}
//while (_msg_queue.empty()) {
// _notify_cond.wait(locker);
//}
msg_ptr = _msg_queue.front();
_msg_queue.pop();
return msg_ptr;
}
void srt2rtmp::check_rtmp_alive() {
const int64_t CHECK_INTERVAL = 5*1000;
const int64_t ALIVE_TIMEOUT_MAX = 5*1000;
if (_lastcheck_ts == 0) {
_lastcheck_ts = now_ms();
return;
}
int64_t timenow_ms = now_ms();
if ((timenow_ms - _lastcheck_ts) > CHECK_INTERVAL) {
_lastcheck_ts = timenow_ms;
for (auto iter = _rtmp_client_map.begin();
iter != _rtmp_client_map.end();) {
RTMP_CLIENT_PTR rtmp_ptr = iter->second;
if ((timenow_ms - rtmp_ptr->get_last_live_ts()) >= ALIVE_TIMEOUT_MAX) {
srs_warn("srt2rtmp client is timeout, url:%s",
rtmp_ptr->get_url().c_str());
_rtmp_client_map.erase(iter++);
rtmp_ptr->close();
} else {
iter++;
}
}
}
return;
}
void srt2rtmp::handle_close_rtmpsession(const std::string& key_path) {
RTMP_CLIENT_PTR rtmp_ptr;
auto iter = _rtmp_client_map.find(key_path);
if (iter == _rtmp_client_map.end()) {
srs_error("fail to close rtmp session fail, can't find session by key_path:%s",
key_path.c_str());
return;
}
rtmp_ptr = iter->second;
_rtmp_client_map.erase(iter);
srs_trace("close rtmp session which key_path is %s", key_path.c_str());
rtmp_ptr->close();
return;
}
//the cycle is running in srs coroutine
srs_error_t srt2rtmp::cycle() {
srs_error_t err = srs_success;
_lastcheck_ts = 0;
while(true) {
SRT_DATA_MSG_PTR msg_ptr = get_data_message();
if (!msg_ptr) {
srs_usleep((30 * SRS_UTIME_MILLISECONDS));
} else {
switch (msg_ptr->msg_type()) {
case SRT_MSG_DATA_TYPE:
{
handle_ts_data(msg_ptr);
break;
}
case SRT_MSG_CLOSE_TYPE:
{
handle_close_rtmpsession(msg_ptr->get_path());
break;
}
default:
{
srs_error("srt to rtmp get wrong message type(%u), path:%s",
msg_ptr->msg_type(), msg_ptr->get_path().c_str());
assert(0);
}
}
}
check_rtmp_alive();
if ((err = _trd_ptr->pull()) != srs_success) {
return srs_error_wrap(err, "forwarder");
}
}
}
void srt2rtmp::handle_ts_data(SRT_DATA_MSG_PTR data_ptr) {
RTMP_CLIENT_PTR rtmp_ptr;
auto iter = _rtmp_client_map.find(data_ptr->get_path());
if (iter == _rtmp_client_map.end()) {
srs_trace("new rtmp client for srt upstream, key_path:%s", data_ptr->get_path().c_str());
rtmp_ptr = std::make_shared<rtmp_client>(data_ptr->get_path());
_rtmp_client_map.insert(std::make_pair(data_ptr->get_path(), rtmp_ptr));
} else {
rtmp_ptr = iter->second;
}
rtmp_ptr->receive_ts_data(data_ptr);
return;
}
rtmp_client::rtmp_client(std::string key_path):_key_path(key_path)
, _connect_flag(false) {
const std::string DEF_VHOST = "DEFAULT_VHOST";
_ts_demux_ptr = std::make_shared<ts_demux>();
_avc_ptr = std::make_shared<SrsRawH264Stream>();
_aac_ptr = std::make_shared<SrsRawAacStream>();
std::vector<std::string> ret_vec;
string_split(key_path, "/", ret_vec);
if (ret_vec.size() >= 3) {
_vhost = ret_vec[0];
_appname = ret_vec[1];
_streamname = ret_vec[2];
} else {
_vhost = DEF_VHOST;
_appname = ret_vec[0];
_streamname = ret_vec[1];
}
char url_sz[128];
std::vector<std::string> ip_ports = _srs_config->get_listens();
int port = 0;
std::string ip;
for (auto item : ip_ports) {
srs_parse_endpoint(item, ip, port);
if (port != 0) {
break;
}
}
if (_vhost == DEF_VHOST) {
sprintf(url_sz, "rtmp://127.0.0.1:%d/%s/%s", port,
_appname.c_str(), _streamname.c_str());
} else {
sprintf(url_sz, "rtmp://127.0.0.1:%d/%s?vhost=%s/%s", port,
_appname.c_str(), _vhost.c_str(), _streamname.c_str());
}
_url = url_sz;
_h264_sps_changed = false;
_h264_pps_changed = false;
_h264_sps_pps_sent = false;
_last_live_ts = now_ms();
srs_trace("rtmp client construct url:%s", url_sz);
}
rtmp_client::~rtmp_client() {
}
void rtmp_client::close() {
_connect_flag = false;
if (!_rtmp_conn_ptr) {
return;
}
srs_trace("rtmp client close url:%s", _url.c_str());
_rtmp_conn_ptr->close();
_rtmp_conn_ptr = nullptr;
}
int64_t rtmp_client::get_last_live_ts() {
return _last_live_ts;
}
std::string rtmp_client::get_url() {
return _url;
}
srs_error_t rtmp_client::connect() {
srs_error_t err = srs_success;
srs_utime_t cto = SRS_CONSTS_RTMP_TIMEOUT;
srs_utime_t sto = SRS_CONSTS_RTMP_PULSE;
_last_live_ts = now_ms();
if (_connect_flag) {
return srs_success;
}
if (_rtmp_conn_ptr.get() != nullptr) {
return srs_error_wrap(err, "repeated connect %s failed, cto=%dms, sto=%dms.",
_url.c_str(), srsu2msi(cto), srsu2msi(sto));
}
_rtmp_conn_ptr = std::make_shared<SrsSimpleRtmpClient>(_url, cto, sto);
if ((err = _rtmp_conn_ptr->connect()) != srs_success) {
close();
return srs_error_wrap(err, "connect %s failed, cto=%dms, sto=%dms.",
_url.c_str(), srsu2msi(cto), srsu2msi(sto));
}
if ((err = _rtmp_conn_ptr->publish(SRS_CONSTS_RTMP_PROTOCOL_CHUNK_SIZE)) != srs_success) {
close();
return srs_error_wrap(err, "publish error, url:%s", _url.c_str());
}
_connect_flag = true;
return err;
}
void rtmp_client::receive_ts_data(SRT_DATA_MSG_PTR data_ptr) {
_ts_demux_ptr->decode(data_ptr, shared_from_this());//on_data_callback is the decode callback
return;
}
srs_error_t rtmp_client::write_h264_sps_pps(uint32_t dts, uint32_t pts) {
srs_error_t err = srs_success;
// TODO: FIMXE: there exists bug, see following comments.
// when sps or pps changed, update the sequence header,
// for the pps maybe not changed while sps changed.
// so, we must check when each video ts message frame parsed.
if (!_h264_sps_changed || !_h264_pps_changed) {
return err;
}
// h264 raw to h264 packet.
std::string sh;
if ((err = _avc_ptr->mux_sequence_header(_h264_sps, _h264_pps, dts, pts, sh)) != srs_success) {
return srs_error_wrap(err, "mux sequence header");
}
// h264 packet to flv packet.
int8_t frame_type = SrsVideoAvcFrameTypeKeyFrame;
int8_t avc_packet_type = SrsVideoAvcFrameTraitSequenceHeader;
char* flv = NULL;
int nb_flv = 0;
if ((err = _avc_ptr->mux_avc2flv(sh, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != srs_success) {
return srs_error_wrap(err, "avc to flv");
}
// the timestamp in rtmp message header is dts.
uint32_t timestamp = dts;
if ((err = rtmp_write_packet(SrsFrameTypeVideo, timestamp, flv, nb_flv)) != srs_success) {
return srs_error_wrap(err, "write packet");
}
// reset sps and pps.
_h264_sps_changed = false;
_h264_pps_changed = false;
_h264_sps_pps_sent = true;
return err;
}
srs_error_t rtmp_client::write_h264_ipb_frame(char* frame, int frame_size, uint32_t dts, uint32_t pts) {
srs_error_t err = srs_success;
// when sps or pps not sent, ignore the packet.
// @see https://github.com/ossrs/srs/issues/203
if (!_h264_sps_pps_sent) {
return srs_error_new(ERROR_H264_DROP_BEFORE_SPS_PPS, "drop sps/pps");
}
// 5bits, 7.3.1 NAL unit syntax,
// ISO_IEC_14496-10-AVC-2003.pdf, page 44.
// 7: SPS, 8: PPS, 5: I Frame, 1: P Frame
SrsAvcNaluType nal_unit_type = (SrsAvcNaluType)(frame[0] & 0x1f);
// for IDR frame, the frame is keyframe.
SrsVideoAvcFrameType frame_type = SrsVideoAvcFrameTypeInterFrame;
if (nal_unit_type == SrsAvcNaluTypeIDR) {
frame_type = SrsVideoAvcFrameTypeKeyFrame;
}
std::string ibp;
if ((err = _avc_ptr->mux_ipb_frame(frame, frame_size, ibp)) != srs_success) {
return srs_error_wrap(err, "mux frame");
}
int8_t avc_packet_type = SrsVideoAvcFrameTraitNALU;
char* flv = NULL;
int nb_flv = 0;
if ((err = _avc_ptr->mux_avc2flv(ibp, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != srs_success) {
return srs_error_wrap(err, "mux avc to flv");
}
// the timestamp in rtmp message header is dts.
uint32_t timestamp = dts;
return rtmp_write_packet(SrsFrameTypeVideo, timestamp, flv, nb_flv);
}
srs_error_t rtmp_client::write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, uint32_t dts) {
srs_error_t err = srs_success;
char* data = NULL;
int size = 0;
if ((err = _aac_ptr->mux_aac2flv(frame, frame_size, codec, dts, &data, &size)) != srs_success) {
return srs_error_wrap(err, "mux aac to flv");
}
return rtmp_write_packet(SrsFrameTypeAudio, dts, data, size);
}
srs_error_t rtmp_client::rtmp_write_packet(char type, uint32_t timestamp, char* data, int size) {
srs_error_t err = srs_success;
SrsSharedPtrMessage* msg = NULL;
if ((err = srs_rtmp_create_msg(type, timestamp, data, size, _rtmp_conn_ptr->sid(), &msg)) != srs_success) {
return srs_error_wrap(err, "create message");
}
srs_assert(msg);
// send out encoded msg.
if ((err = _rtmp_conn_ptr->send_and_free_message(msg)) != srs_success) {
close();
return srs_error_wrap(err, "send messages");
}
return err;
}
srs_error_t rtmp_client::on_ts_video(std::shared_ptr<SrsBuffer> avs_ptr, uint64_t dts, uint64_t pts) {
srs_error_t err = srs_success;
// ensure rtmp connected.
if ((err = connect()) != srs_success) {
return srs_error_wrap(err, "connect");
}
// send each frame.
while (!avs_ptr->empty()) {
char* frame = NULL;
int frame_size = 0;
if ((err = _avc_ptr->annexb_demux(avs_ptr.get(), &frame, &frame_size)) != srs_success) {
return srs_error_wrap(err, "demux annexb");
}
//srs_trace_data(frame, frame_size, "video annexb demux:");
// 5bits, 7.3.1 NAL unit syntax,
// ISO_IEC_14496-10-AVC-2003.pdf, page 44.
// 7: SPS, 8: PPS, 5: I Frame, 1: P Frame
SrsAvcNaluType nal_unit_type = (SrsAvcNaluType)(frame[0] & 0x1f);
// ignore the nalu type sps(7), pps(8), aud(9)
if (nal_unit_type == SrsAvcNaluTypeAccessUnitDelimiter) {
continue;
}
// for sps
if (_avc_ptr->is_sps(frame, frame_size)) {
std::string sps;
if ((err = _avc_ptr->sps_demux(frame, frame_size, sps)) != srs_success) {
return srs_error_wrap(err, "demux sps");
}
if (_h264_sps == sps) {
continue;
}
_h264_sps_changed = true;
_h264_sps = sps;
if ((err = write_h264_sps_pps(dts, pts)) != srs_success) {
return srs_error_wrap(err, "write sps/pps");
}
continue;
}
// for pps
if (_avc_ptr->is_pps(frame, frame_size)) {
std::string pps;
if ((err = _avc_ptr->pps_demux(frame, frame_size, pps)) != srs_success) {
return srs_error_wrap(err, "demux pps");
}
if (_h264_pps == pps) {
continue;
}
_h264_pps_changed = true;
_h264_pps = pps;
if ((err = write_h264_sps_pps(dts, pts)) != srs_success) {
return srs_error_wrap(err, "write sps/pps");
}
continue;
}
// ibp frame.
// TODO: FIXME: we should group all frames to a rtmp/flv message from one ts message.
srs_info("mpegts: demux avc ibp frame size=%d, dts=%d", frame_size, dts);
if ((err = write_h264_ipb_frame(frame, frame_size, dts, pts)) != srs_success) {
return srs_error_wrap(err, "write frame");
}
_last_live_ts = now_ms();
}
return err;
}
srs_error_t rtmp_client::on_ts_audio(std::shared_ptr<SrsBuffer> avs_ptr, uint64_t dts, uint64_t pts) {
srs_error_t err = srs_success;
// ensure rtmp connected.
if ((err = connect()) != srs_success) {
return srs_error_wrap(err, "connect");
}
// send each frame.
while (!avs_ptr->empty()) {
char* frame = NULL;
int frame_size = 0;
SrsRawAacStreamCodec codec;
if ((err = _aac_ptr->adts_demux(avs_ptr.get(), &frame, &frame_size, codec)) != srs_success) {
return srs_error_wrap(err, "demux adts");
}
//srs_trace("audio annexb demux sampling_frequency_index:%d, aac_packet_type:%d, sound_rate:%d, sound_size:%d",
// codec.sampling_frequency_index, codec.aac_packet_type, codec.sound_rate,
// codec.sound_size);
//srs_trace_data(frame, frame_size, "audio annexb demux:");
// ignore invalid frame,
// * atleast 1bytes for aac to decode the data.
if (frame_size <= 0) {
continue;
}
// generate sh.
if (_aac_specific_config.empty()) {
std::string sh;
if ((err = _aac_ptr->mux_sequence_header(&codec, sh)) != srs_success) {
return srs_error_wrap(err, "mux sequence header");
}
_aac_specific_config = sh;
codec.aac_packet_type = 0;
if ((err = write_audio_raw_frame((char*)sh.data(), (int)sh.length(), &codec, dts)) != srs_success) {
return srs_error_wrap(err, "write raw audio frame");
}
}
// audio raw data.
codec.aac_packet_type = 1;
if ((err = write_audio_raw_frame(frame, frame_size, &codec, dts)) != srs_success) {
return srs_error_wrap(err, "write audio raw frame");
}
_last_live_ts = now_ms();
}
return err;
}
void rtmp_client::on_data_callback(SRT_DATA_MSG_PTR data_ptr, unsigned int media_type,
uint64_t dts, uint64_t pts)
{
if (!data_ptr || (data_ptr->get_data() == nullptr) || (data_ptr->data_len() == 0)) {
assert(0);
return;
}
auto avs_ptr = std::make_shared<SrsBuffer>((char*)data_ptr->get_data(), data_ptr->data_len());
dts = dts / 90;
pts = pts / 90;
if (media_type == STREAM_TYPE_VIDEO_H264) {
on_ts_video(avs_ptr, dts, pts);
} else if (media_type == STREAM_TYPE_AUDIO_AAC) {
on_ts_audio(avs_ptr, dts, pts);
} else {
srs_error("mpegts demux unkown stream type:0x%02x", media_type);
assert(0);
}
return;
}