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4 Commits
| Author | SHA1 | Date | |
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88460b72e1 | ||
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8c69c77481 | ||
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5331ae14aa | ||
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0127dcf018 |
@ -1,7 +1,7 @@
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[bumpversion]
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commit = True
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tag = True
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current_version = 6.53
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current_version = 6.55
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parse = (?P<major>\d+)\.(?P<minor>\d+)
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serialize =
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{major}.{minor}
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@ -27,8 +27,13 @@
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// Max RTP size for WebRTC is 1200 bytes:
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// - https://stackoverflow.com/questions/47635545/why-webrtc-chose-rtp-max-packet-size-to-1200-bytes
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// We take this and substract 50 bytes for possible RTP extensions, see sdp.c
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#define US_RTP_TOTAL_SIZE (1200 - 50)
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// But(!) Tailscale has 1200 MTU. So to fit it required to substract:
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// 1. possible RTP extensions (see sdp.c)
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// 2. additional SRTP fields (>= 10 bytes)
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// 3. UDP header (8 bytes)
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// 4. IPv6 header (40 bytes)
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// Finally it looks like 100 bytes for all above should be enough
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#define US_RTP_TOTAL_SIZE (1200 - 100)
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#define US_RTP_HEADER_SIZE 12
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#define US_RTP_PAYLOAD_SIZE (US_RTP_TOTAL_SIZE - US_RTP_HEADER_SIZE)
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@ -106,6 +106,7 @@ void _rtpv_process_nalu(us_rtpv_s *rtpv, const u8 *data, uz size, u32 pts, bool
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u8 *dg = rtpv->rtp->datagram;
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// Set *_of_frame flags only for non-SPS/PPS packages
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/*
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# define CALL_FOR_SERVICE { \
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const bool m_fof = rtpv->rtp->first_of_frame; \
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const bool m_lof = rtpv->rtp->last_of_frame; \
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@ -115,18 +116,19 @@ void _rtpv_process_nalu(us_rtpv_s *rtpv, const u8 *data, uz size, u32 pts, bool
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rtpv->rtp->first_of_frame = m_fof; \
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rtpv->rtp->last_of_frame = m_lof; \
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}
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*/
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if (size + US_RTP_HEADER_SIZE <= US_RTP_TOTAL_SIZE) {
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us_rtp_write_header(rtpv->rtp, pts, marked);
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memcpy(dg + US_RTP_HEADER_SIZE, data, size);
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rtpv->rtp->used = size + US_RTP_HEADER_SIZE;
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if (type == 7 || type == 8) {
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CALL_FOR_SERVICE;
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} else {
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// if (type == 7 || type == 8) {
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// CALL_FOR_SERVICE;
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// } else {*/
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rtpv->rtp->last_of_frame = true;
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rtpv->callback(rtpv->rtp);
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rtpv->rtp->first_of_frame = false;
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}
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// }
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return;
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}
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@ -158,13 +160,13 @@ void _rtpv_process_nalu(us_rtpv_s *rtpv, const u8 *data, uz size, u32 pts, bool
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memcpy(dg + fu_overhead, src, frag_size);
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rtpv->rtp->used = fu_overhead + frag_size;
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if (type == 7 || type == 8) {
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CALL_FOR_SERVICE;
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} else {
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// if (type == 7 || type == 8) {
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// CALL_FOR_SERVICE;
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// } else {
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rtpv->rtp->last_of_frame = last;
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rtpv->callback(rtpv->rtp);
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rtpv->rtp->first_of_frame = false;
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}
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// }
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src += frag_size;
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remaining -= frag_size;
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@ -1,6 +1,6 @@
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.\" Manpage for ustreamer-dump.
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.\" Open an issue or pull request to https://github.com/pikvm/ustreamer to correct errors or typos
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.TH USTREAMER-DUMP 1 "version 6.53" "January 2021"
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.TH USTREAMER-DUMP 1 "version 6.55" "January 2021"
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.SH NAME
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ustreamer-dump \- Dump uStreamer's memory sink to file
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@ -1,6 +1,6 @@
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.\" Manpage for ustreamer.
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.\" Open an issue or pull request to https://github.com/pikvm/ustreamer to correct errors or typos
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.TH USTREAMER 1 "version 6.53" "November 2020"
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.TH USTREAMER 1 "version 6.55" "November 2020"
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.SH NAME
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ustreamer \- stream MJPEG video from any V4L2 device to the network
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@ -3,7 +3,7 @@
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pkgname=ustreamer
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pkgver=6.53
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pkgver=6.55
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pkgrel=1
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pkgdesc="Lightweight and fast MJPEG-HTTP streamer"
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url="https://github.com/pikvm/ustreamer"
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@ -9,7 +9,7 @@
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include $(TOPDIR)/rules.mk
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PKG_NAME:=ustreamer
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PKG_VERSION:=6.53
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PKG_VERSION:=6.55
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PKG_RELEASE:=1
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PKG_MAINTAINER:=Maxim Devaev <mdevaev@gmail.com>
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@ -34,7 +34,7 @@ def main() -> None:
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flags = _find_flags()
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setup(
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name="ustreamer",
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version="6.53",
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version="6.55",
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description="uStreamer tools",
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author="Maxim Devaev",
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author_email="mdevaev@gmail.com",
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@ -26,7 +26,7 @@
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#define US_VERSION_MAJOR 6
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#define US_VERSION_MINOR 53
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#define US_VERSION_MINOR 55
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#define US_MAKE_VERSION2(_major, _minor) #_major "." #_minor
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#define US_MAKE_VERSION1(_major, _minor) US_MAKE_VERSION2(_major, _minor)
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