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RTC: Fix GSO build failed in linux

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This commit is contained in:
winlin 2020-05-14 13:14:52 +08:00
parent 3cb797dccd
commit 4e1935f678
4 changed files with 20 additions and 19 deletions
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4
trunk/src/app/srs_app_config.cpp
View file

@ -4879,7 +4879,9 @@ bool SrsConfig::get_rtc_server_gso()
}
#elif LINUX_VERSION_CODE < KERNEL_VERSION(4,18,0)
if (v) {
utsname un = {0};
utsname un;
memset((void*)&un, 0, sizeof(utsname));
int r0 = uname(&un);
if (r0 || strcmp(un.release, "4.18.0") < 0) {
gso_disabled = true;

32
trunk/src/app/srs_app_rtc_conn.cpp
View file

@ -788,8 +788,8 @@ srs_error_t SrsRtcPlayer::send_messages(SrsRtcSource* source, vector<SrsRtpPacke
// If enabled GSO, send out some packets in a msghdr.
// @remark When NACK simulator is on, we don't use GSO.
// TODO: FIXME: Support GSO.
if (packets.use_gso && !nn_simulate_nack_drop) {
if ((err = send_packets_gso(info)) != srs_success) {
if (info.use_gso && !nn_simulate_nack_drop) {
if ((err = send_packets_gso(pkts, info)) != srs_success) {
return srs_error_wrap(err, "gso send");
}
return err;
@ -937,7 +937,7 @@ srs_error_t SrsRtcPlayer::send_packets(std::vector<SrsRtpPacket2*>& pkts, SrsRtc
}
// TODO: FIXME: We can gather and pad audios, because they have similar size.
srs_error_t SrsRtcPlayer::send_packets_gso(SrsRtcOutgoingPackets& packets)
srs_error_t SrsRtcPlayer::send_packets_gso(vector<SrsRtpPacket2*>& pkts, SrsRtcOutgoingPackets& info)
{
srs_error_t err = srs_success;
@ -952,9 +952,9 @@ srs_error_t SrsRtcPlayer::send_packets_gso(SrsRtcOutgoingPackets& packets)
// The message will marshal in iovec.
iovec* iov = NULL;
int nn_packets = packets.size();
int nn_packets = pkts.size();
for (int i = 0; i < nn_packets; i++) {
SrsRtpPacket2* packet = packets.at(i);
SrsRtpPacket2* packet = pkts.at(i);
int nn_packet = packet->nb_bytes();
int padding = 0;
@ -962,7 +962,7 @@ srs_error_t SrsRtcPlayer::send_packets_gso(SrsRtcOutgoingPackets& packets)
int nn_next_packet = 0;
if (max_padding > 0) {
if (i < nn_packets - 1) {
next_packet = (i < nn_packets - 1)? packets.at(i + 1):NULL;
next_packet = (i < nn_packets - 1)? pkts.at(i + 1):NULL;
nn_next_packet = next_packet? next_packet->nb_bytes() : 0;
}
@ -987,13 +987,13 @@ srs_error_t SrsRtcPlayer::send_packets_gso(SrsRtcOutgoingPackets& packets)
if (padding > 0) {
#if defined(SRS_DEBUG)
srs_trace("#%d, Padding %d bytes %d=>%d, packets %d, max_padding %d", packets.debug_id,
srs_trace("#%d, Padding %d bytes %d=>%d, packets %d, max_padding %d", info.debug_id,
padding, nn_packet, nn_packet + padding, nn_packets, max_padding);
#endif
packet->add_padding(padding);
nn_packet += padding;
packets.nn_paddings++;
packets.nn_padding_bytes += padding;
info.nn_paddings++;
info.nn_padding_bytes += padding;
}
}
}
@ -1081,7 +1081,7 @@ srs_error_t SrsRtcPlayer::send_packets_gso(SrsRtcOutgoingPackets& packets)
}
}
packets.nn_rtp_bytes += (int)iov->iov_len;
info.nn_rtp_bytes += (int)iov->iov_len;
// If GSO, they must has same size, except the final one.
if (using_gso && !gso_final && gso_encrypt && gso_encrypt != (int)iov->iov_len) {
@ -1102,12 +1102,12 @@ srs_error_t SrsRtcPlayer::send_packets_gso(SrsRtcOutgoingPackets& packets)
#if defined(SRS_DEBUG)
bool is_video = packet->rtp_header.get_payload_type() == video_payload_type;
srs_trace("#%d, Packet %s SSRC=%d, SN=%d, %d/%d bytes", packets.debug_id, is_video? "Video":"Audio",
srs_trace("#%d, Packet %s SSRC=%d, SN=%d, %d/%d bytes", info.debug_id, is_video? "Video":"Audio",
packet->rtp_header.get_ssrc(), packet->rtp_header.get_sequence(), nn_packet - padding, padding);
if (do_send) {
for (int j = 0; j < (int)mhdr->msg_hdr.msg_iovlen; j++) {
iovec* iov = mhdr->msg_hdr.msg_iov + j;
srs_trace("#%d, %s #%d/%d/%d, %d/%d bytes, size %d/%d", packets.debug_id, (using_gso? "GSO":"RAW"), j,
srs_trace("#%d, %s #%d/%d/%d, %d/%d bytes, size %d/%d", info.debug_id, (using_gso? "GSO":"RAW"), j,
gso_cursor + 1, mhdr->msg_hdr.msg_iovlen, iov->iov_len, padding, gso_size, gso_encrypt);
}
}
@ -1138,7 +1138,7 @@ srs_error_t SrsRtcPlayer::send_packets_gso(SrsRtcOutgoingPackets& packets)
#endif
// When we send out a packet, we commit a RTP packet.
packets.nn_rtp_pkts++;
info.nn_rtp_pkts++;
if ((err = sender->sendmmsg(mhdr)) != srs_success) {
return srs_error_wrap(err, "send msghdr");
@ -1151,9 +1151,9 @@ srs_error_t SrsRtcPlayer::send_packets_gso(SrsRtcOutgoingPackets& packets)
}
#if defined(SRS_DEBUG)
srs_trace("#%d, RTC PLAY summary, rtp %d/%d, videos %d/%d, audios %d/%d, pad %d/%d/%d", packets.debug_id, packets.size(),
packets.nn_rtp_pkts, packets.nn_videos, packets.nn_samples, packets.nn_audios, packets.nn_extras, packets.nn_paddings,
packets.nn_padding_bytes, packets.nn_rtp_bytes);
srs_trace("#%d, RTC PLAY summary, rtp %d/%d, videos %d/%d, audios %d/%d, pad %d/%d/%d", info.debug_id, pkts.size(),
info.nn_rtp_pkts, info.nn_videos, info.nn_samples, info.nn_audios, info.nn_extras, info.nn_paddings,
info.nn_padding_bytes, info.nn_rtp_bytes);
#endif
return err;

2
trunk/src/app/srs_app_rtc_conn.hpp
View file

@ -258,7 +258,7 @@ private:
srs_error_t send_messages(SrsRtcSource* source, std::vector<SrsRtpPacket2*>& pkts, SrsRtcOutgoingPackets& info);
srs_error_t messages_to_packets(SrsRtcSource* source, std::vector<SrsRtpPacket2*>& pkts, SrsRtcOutgoingPackets& info);
srs_error_t send_packets(std::vector<SrsRtpPacket2*>& pkts, SrsRtcOutgoingPackets& info);
srs_error_t send_packets_gso(SrsRtcOutgoingPackets& packets);
srs_error_t send_packets_gso(std::vector<SrsRtpPacket2*>& pkts, SrsRtcOutgoingPackets& info);
private:
srs_error_t package_opus(SrsRtpPacket2* pkt);
srs_error_t package_video(SrsRtpPacket2* pkt);

1
trunk/src/app/srs_app_rtc_source.cpp
View file

@ -669,7 +669,6 @@ srs_error_t SrsRtcFromRtmpBridger::transcode(char* adts_audio, int nn_adts_audio
}
int nn_max_extra_payload = 0;
SrsSample samples[nn_opus_packets];
for (int i = 0; i < nn_opus_packets; i++) {
char* data = (char*)opus_payloads[i];
int size = (int)opus_sizes[i];