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Add debug_id for GSO
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parent
56995db6a5
commit
b7dab00f36
2 changed files with 20 additions and 7 deletions
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@ -462,6 +462,10 @@ srs_error_t SrsDtlsSession::unprotect_rtcp(char* out_buf, const char* in_buf, in
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SrsRtcPackets::SrsRtcPackets()
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{
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#if defined(SRS_DEBUG)
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debug_id = 0;
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#endif
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use_gso = false;
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should_merge_nalus = false;
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@ -490,6 +494,10 @@ void SrsRtcPackets::reset(bool gso, bool merge_nalus)
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packet->reset();
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}
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#if defined(SRS_DEBUG)
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debug_id++;
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#endif
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use_gso = gso;
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should_merge_nalus = merge_nalus;
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@ -951,7 +959,8 @@ srs_error_t SrsRtcSenderThread::send_packets_gso(SrsUdpMuxSocket* skt, SrsRtcPac
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if (padding > 0 && padding < max_padding) {
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#if defined(SRS_DEBUG)
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srs_trace("Padding %d bytes %d=>%d, packets %d, max_padding %d", padding, nn_packet, nn_packet + padding, nn_packets, max_padding);
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srs_trace("#%d, Padding %d bytes %d=>%d, packets %d, max_padding %d", packets.debug_id,
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padding, nn_packet, nn_packet + padding, nn_packets, max_padding);
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#endif
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packet->set_padding(padding);
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nn_packet += padding;
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@ -1063,8 +1072,8 @@ srs_error_t SrsRtcSenderThread::send_packets_gso(SrsUdpMuxSocket* skt, SrsRtcPac
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#if defined(SRS_DEBUG)
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bool is_video = packet->rtp_header.get_payload_type() == video_payload_type;
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srs_trace("Packet %s SSRC=%d, SN=%d, %d bytes", is_video? "Video":"Audio", packet->rtp_header.get_ssrc(),
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packet->rtp_header.get_sequence(), nn_packet);
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srs_trace("#%d, Packet %s SSRC=%d, SN=%d, %d bytes", packets.debug_id, is_video? "Video":"Audio",
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packet->rtp_header.get_ssrc(), packet->rtp_header.get_sequence(), nn_packet);
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if (do_send) {
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for (int j = 0; j < (int)mhdr->msg_hdr.msg_iovlen; j++) {
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iovec* iov = mhdr->msg_hdr.msg_iov + j;
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@ -1072,8 +1081,8 @@ srs_error_t SrsRtcSenderThread::send_packets_gso(SrsUdpMuxSocket* skt, SrsRtcPac
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if (iov->iov_len <= 0) {
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break;
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}
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srs_trace("%s #%d/%d/%d, %d bytes, size %d/%d", (use_gso? "GSO":"RAW"), j, gso_cursor + 1,
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mhdr->msg_hdr.msg_iovlen, iov->iov_len, gso_size, gso_encrypt);
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srs_trace("#%d, %s #%d/%d/%d, %d bytes, size %d/%d", packets.debug_id, (use_gso? "GSO":"RAW"), j,
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gso_cursor + 1, mhdr->msg_hdr.msg_iovlen, iov->iov_len, gso_size, gso_encrypt);
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}
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}
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#endif
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@ -1119,8 +1128,8 @@ srs_error_t SrsRtcSenderThread::send_packets_gso(SrsUdpMuxSocket* skt, SrsRtcPac
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}
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#if defined(SRS_DEBUG)
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srs_trace("RTC PLAY summary, rtp %d/%d, videos %d/%d, audios %d/%d, pad %d", packets.size(), packets.nn_rtp_pkts,
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packets.nn_videos, packets.nn_samples, packets.nn_audios, packets.nn_extras, packets.nn_paddings);
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srs_trace("#%d, RTC PLAY summary, rtp %d/%d, videos %d/%d, audios %d/%d, pad %d", packets.debug_id, packets.size(),
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packets.nn_rtp_pkts, packets.nn_videos, packets.nn_samples, packets.nn_audios, packets.nn_extras, packets.nn_paddings);
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#endif
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return err;
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@ -124,6 +124,10 @@ public:
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bool use_gso;
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bool should_merge_nalus;
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public:
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#if defined(SRS_DEBUG)
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// Debug id.
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uint32_t debug_id;
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#endif
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// The total bytes of RTP packets.
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int nn_bytes;
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// The RTP packets send out by sendmmsg or sendmsg. Note that if many packets group to
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