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RTC: Remove dead code

This commit is contained in:
winlin 2021-02-27 21:46:50 +08:00
parent 00b0e22402
commit 81dddcbd93
3 changed files with 0 additions and 145 deletions

View file

@ -366,19 +366,6 @@ srs_error_t SrsRtcPLIWorker::cycle()
return err;
}
SrsRtcPlayStreamStatistic::SrsRtcPlayStreamStatistic()
{
nn_rtp_pkts = 0;
nn_audios = nn_extras = 0;
nn_videos = nn_samples = 0;
nn_bytes = nn_rtp_bytes = 0;
nn_padding_bytes = nn_paddings = 0;
}
SrsRtcPlayStreamStatistic::~SrsRtcPlayStreamStatistic()
{
}
SrsRtcPlayStream::SrsRtcPlayStream(SrsRtcConnection* s, const SrsContextId& cid)
{
cid_ = cid;
@ -593,47 +580,6 @@ srs_error_t SrsRtcPlayStream::cycle()
}
}
srs_error_t SrsRtcPlayStream::send_packets(SrsRtcStream* source, const vector<SrsRtpPacket2*>& pkts, SrsRtcPlayStreamStatistic& info)
{
srs_error_t err = srs_success;
// Covert kernel messages to RTP packets.
for (int i = 0; i < (int)pkts.size(); i++) {
SrsRtpPacket2* pkt = pkts[i];
// TODO: FIXME: Maybe refine for performance issue.
if (!audio_tracks_.count(pkt->header.get_ssrc()) && !video_tracks_.count(pkt->header.get_ssrc())) {
srs_warn("ssrc %u not found", pkt->header.get_ssrc());
continue;
}
// For audio, we transcoded AAC to opus in extra payloads.
if (pkt->is_audio()) {
// TODO: FIXME: Any simple solution?
SrsRtcAudioSendTrack* audio_track = audio_tracks_[pkt->header.get_ssrc()];
if ((err = audio_track->on_rtp(pkt)) != srs_success) {
return srs_error_wrap(err, "audio track, SSRC=%u, SEQ=%u", pkt->header.get_ssrc(), pkt->header.get_sequence());
}
// TODO: FIXME: Padding audio to the max payload in RTP packets.
} else {
// TODO: FIXME: Any simple solution?
SrsRtcVideoSendTrack* video_track = video_tracks_[pkt->header.get_ssrc()];
if ((err = video_track->on_rtp(pkt)) != srs_success) {
return srs_error_wrap(err, "video track, SSRC=%u, SEQ=%u", pkt->header.get_ssrc(), pkt->header.get_sequence());
}
}
// Detail log, should disable it in release version.
srs_info("RTC: Update PT=%u, SSRC=%#x, Time=%u, %u bytes", pkt->header.get_payload_type(), pkt->header.get_ssrc(),
pkt->header.get_timestamp(), pkt->nb_bytes());
}
return err;
}
srs_error_t SrsRtcPlayStream::send_packet(SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
@ -2508,60 +2454,6 @@ void SrsRtcConnection::simulate_player_drop_packet(SrsRtpHeader* h, int nn_bytes
nn_simulate_player_nack_drop--;
}
srs_error_t SrsRtcConnection::do_send_packets(const std::vector<SrsRtpPacket2*>& pkts, SrsRtcPlayStreamStatistic& info)
{
srs_error_t err = srs_success;
for (int i = 0; i < (int)pkts.size(); i++) {
SrsRtpPacket2* pkt = pkts.at(i);
// For this message, select the first iovec.
iovec* iov = cache_iov_;
iov->iov_len = kRtpPacketSize;
cache_buffer_->skip(-1 * cache_buffer_->pos());
// Marshal packet to bytes in iovec.
if (true) {
if ((err = pkt->encode(cache_buffer_)) != srs_success) {
return srs_error_wrap(err, "encode packet");
}
iov->iov_len = cache_buffer_->pos();
}
// Cipher RTP to SRTP packet.
if (true) {
int nn_encrypt = (int)iov->iov_len;
if ((err = transport_->protect_rtp(iov->iov_base, &nn_encrypt)) != srs_success) {
return srs_error_wrap(err, "srtp protect");
}
iov->iov_len = (size_t)nn_encrypt;
}
info.nn_rtp_bytes += (int)iov->iov_len;
// When we send out a packet, increase the stat counter.
info.nn_rtp_pkts++;
// For NACK simulator, drop packet.
if (nn_simulate_player_nack_drop) {
simulate_player_drop_packet(&pkt->header, (int)iov->iov_len);
iov->iov_len = 0;
continue;
}
++_srs_pps_srtps->sugar;
// TODO: FIXME: Handle error.
sendonly_skt->sendto(iov->iov_base, iov->iov_len, 0);
// Detail log, should disable it in release version.
srs_info("RTC: SEND PT=%u, SSRC=%#x, SEQ=%u, Time=%u, %u/%u bytes", pkt->header.get_payload_type(), pkt->header.get_ssrc(),
pkt->header.get_sequence(), pkt->header.get_timestamp(), pkt->nb_bytes(), iov->iov_len);
}
return err;
}
srs_error_t SrsRtcConnection::do_send_packet(SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;