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RTC: Remove dead code

Browse Source
pull/2252/head
winlin 4 years ago
parent 3f36397f98
commit 831a1b146f
3 changed files with 0 additions and 145 deletions
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108
trunk/src/app/srs_app_rtc_conn.cpp
Unescape Escape 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;

36
trunk/src/app/srs_app_rtc_conn.hpp
Unescape Escape View File

@ -209,38 +209,6 @@ public:
virtual srs_error_t cycle();
};
// A group of RTP packets for outgoing(send to players).
class SrsRtcPlayStreamStatistic
{
public:
// The total bytes of AVFrame packets.
int nn_bytes;
// The total bytes of RTP packets.
int nn_rtp_bytes;
// The total padded bytes.
int nn_padding_bytes;
public:
// The RTP packets send out by sendmmsg or sendmsg. Note that if many packets group to
// one msghdr by GSO, it's only one RTP packet, because we only send once.
int nn_rtp_pkts;
// For video, the samples or NALUs.
// TODO: FIXME: Remove it because we may don't know.
int nn_samples;
// For audio, the generated extra audio packets.
// For example, when transcoding AAC to opus, may many extra payloads for a audio.
// TODO: FIXME: Remove it because we may don't know.
int nn_extras;
// The original audio messages.
int nn_audios;
// The original video messages.
int nn_videos;
// The number of padded packet.
int nn_paddings;
public:
SrsRtcPlayStreamStatistic();
virtual ~SrsRtcPlayStreamStatistic();
};
// A RTC play stream, client pull and play stream from SRS.
class SrsRtcPlayStream : virtual public ISrsCoroutineHandler, virtual public ISrsReloadHandler
, virtual public ISrsHourGlass, virtual public ISrsRtcPLIWorkerHandler
@ -268,8 +236,6 @@ private:
private:
// Whether palyer started.
bool is_started;
// The statistic for consumer to send packets to player.
SrsRtcPlayStreamStatistic info;
public:
SrsRtcPlayStream(SrsRtcConnection* s, const SrsContextId& cid);
virtual ~SrsRtcPlayStream();
@ -286,7 +252,6 @@ public:
public:
virtual srs_error_t cycle();
private:
srs_error_t send_packets(SrsRtcStream* source, const std::vector<SrsRtpPacket2*>& pkts, SrsRtcPlayStreamStatistic& info);
srs_error_t send_packet(SrsRtpPacket2* pkt);
public:
// Directly set the status of track, generally for init to set the default value.
@ -551,7 +516,6 @@ public:
// Simulate the NACK to drop nn packets.
void simulate_nack_drop(int nn);
void simulate_player_drop_packet(SrsRtpHeader* h, int nn_bytes);
srs_error_t do_send_packets(const std::vector<SrsRtpPacket2*>& pkts, SrsRtcPlayStreamStatistic& info);
srs_error_t do_send_packet(SrsRtpPacket2* pkt);
// Directly set the status of play track, generally for init to set the default value.
void set_all_tracks_status(std::string stream_uri, bool is_publish, bool status);

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

@ -55,7 +55,6 @@ class SrsRtcConnection;
class SrsRtpRingBuffer;
class SrsRtpNackForReceiver;
class SrsJsonObject;
class SrsRtcPlayStreamStatistic;
class SrsErrorPithyPrint;
class SrsRtcDummyBridger;

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