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Code Issues Releases Wiki Activity

For #307, support padding for GSO

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This commit is contained in:
winlin 2020-04-15 15:58:17 +08:00
parent 6869ccca50
commit bbdd2d7eed
5 changed files with 68 additions and 17 deletions
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55
trunk/src/app/srs_app_rtc_conn.cpp
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@ -468,6 +468,7 @@ SrsRtcPackets::SrsRtcPackets(bool gso, bool merge_nalus)
nn_rtp_pkts = 0;
nn_audios = nn_extras = 0;
nn_videos = nn_samples = 0;
nn_paddings = 0;
}
SrsRtcPackets::~SrsRtcPackets()
@ -490,6 +491,7 @@ SrsRtcSenderThread::SrsRtcSenderThread(SrsRtcSession* s, SrsUdpMuxSocket* u, int
sendonly_ukt = u->copy_sendonly();
gso = false;
merge_nalus = false;
max_padding = 0;
audio_timestamp = 0;
audio_sequence = 0;
@ -519,8 +521,9 @@ srs_error_t SrsRtcSenderThread::initialize(const uint32_t& vssrc, const uint32_t
gso = _srs_config->get_rtc_server_gso();
merge_nalus = _srs_config->get_rtc_server_merge_nalus();
srs_trace("RTC sender video(ssrc=%d, pt=%d), audio(ssrc=%d, pt=%d), package(gso=%d, merge_nalus=%d)",
video_ssrc, video_payload_type, audio_ssrc, audio_payload_type, gso, merge_nalus);
max_padding = _srs_config->get_rtc_server_padding();
srs_trace("RTC sender video(ssrc=%d, pt=%d), audio(ssrc=%d, pt=%d), package(gso=%d, merge_nalus=%d), padding=%d",
video_ssrc, video_payload_type, audio_ssrc, audio_payload_type, gso, merge_nalus, max_padding);
return err;
}
@ -543,6 +546,14 @@ srs_error_t SrsRtcSenderThread::on_reload_rtc_server()
}
}
if (true) {
bool v = _srs_config->get_rtc_server_padding();
if (max_padding != v) {
srs_trace("Reload padding %d=>%d", max_padding, v);
max_padding = v;
}
}
return srs_success;
}
@ -665,7 +676,7 @@ srs_error_t SrsRtcSenderThread::cycle()
// Stat the RTP packets going into kernel.
stat->perf_on_gso_packets(pkts.nn_rtp_pkts);
#if defined(SRS_DEBUG)
srs_trace("RTC PLAY packets, msgs %d/%d, rtp %d, gso %d, %d audios, %d extras, %d videos, %d samples, %d bytes",
srs_trace("RTC PLAY done, msgs %d/%d, rtp %d, gso %d, %d audios, %d extras, %d videos, %d samples, %d bytes",
msg_count, nn_rtc_packets, pkts.packets.size(), pkts.nn_rtp_pkts, pkts.nn_audios, pkts.nn_extras, pkts.nn_videos,
pkts.nn_samples, pkts.nn_bytes);
#endif
@ -673,9 +684,9 @@ srs_error_t SrsRtcSenderThread::cycle()
pprint->elapse();
if (pprint->can_print()) {
// TODO: FIXME: Print stat like frame/s, packet/s, loss_packets.
srs_trace("-> RTC PLAY %d msgs, %d/%d packets, %d audios, %d extras, %d videos, %d samples, %d bytes",
srs_trace("-> RTC PLAY %d msgs, %d/%d packets, %d audios, %d extras, %d videos, %d samples, %d bytes, %d pad",
msg_count, pkts.packets.size(), pkts.nn_rtp_pkts, pkts.nn_audios, pkts.nn_extras, pkts.nn_videos,
pkts.nn_samples, pkts.nn_bytes);
pkts.nn_samples, pkts.nn_bytes, pkts.nn_paddings);
}
}
}
@ -862,13 +873,28 @@ srs_error_t SrsRtcSenderThread::send_packets_gso(SrsUdpMuxSocket* skt, SrsRtcPac
int nn_packets = (int)packets.packets.size();
for (int i = 0; i < nn_packets; i++) {
SrsRtpPacket2* packet = packets.packets[i];
// The handler to send message.
mmsghdr* mhdr = NULL;
// Check whether we can use GSO to send it.
int nn_packet = packet->nb_bytes();
SrsRtpPacket2* next_packet = NULL;
int nn_next_packet = 0;
if (i < nn_packets - 1) {
next_packet = (i < nn_packets - 1)? packets.packets[i + 1]:NULL;
nn_next_packet = next_packet? next_packet->nb_bytes() : 0;
}
// Padding the first packet if size is similar to the next one.
if (i == 0 && max_padding > 0 && next_packet && nn_packet < nn_next_packet && nn_next_packet - nn_packet < max_padding) {
#if defined(SRS_DEBUG)
srs_trace("Padding %d bytes %d=>%d, packets %d, max_padding %d", nn_next_packet - nn_packet,
nn_packet, nn_next_packet, nn_packets, max_padding);
#endif
packet->set_padding(nn_next_packet - nn_packet);
nn_packet = nn_next_packet;
packets.nn_paddings++;
}
// Check whether we can use GSO to send it.
mmsghdr* mhdr = NULL;
if ((gso_size && gso_size == nn_packet) || (use_gso && !gso_final)) {
use_gso = true;
gso_final = (gso_size && gso_size != nn_packet);
@ -893,10 +919,7 @@ srs_error_t SrsRtcSenderThread::send_packets_gso(SrsUdpMuxSocket* skt, SrsRtcPac
}
// Change the state according to the next packet.
if (i < nn_packets - 1) {
SrsRtpPacket2* next_packet = (i < nn_packets - 1)? packets.packets[i + 1]:NULL;
int nn_next_packet = next_packet? next_packet->nb_bytes() : 0;
if (next_packet) {
// If GSO, but next is bigger than this one, we must enter the final state.
if (use_gso && !gso_final) {
gso_final = (nn_packet < nn_next_packet);
@ -1025,8 +1048,8 @@ srs_error_t SrsRtcSenderThread::send_packets_gso(SrsUdpMuxSocket* skt, SrsRtcPac
}
#if defined(SRS_DEBUG)
srs_trace("RTC PLAY summary, rtp %d/%d, videos %d/%d, audios %d/%d", packets.packets.size(),
packets.nn_rtp_pkts, packets.nn_videos, packets.nn_samples, packets.nn_audios, packets.nn_extras);
srs_trace("RTC PLAY summary, rtp %d/%d, videos %d/%d, audios %d/%d, pad %d", packets.packets.size(), packets.nn_rtp_pkts,
packets.nn_videos, packets.nn_samples, packets.nn_audios, packets.nn_extras, packets.nn_paddings);
#endif
return err;