Bump version: 5.28 → 5.29

* Assign stream index on outgoing RTP packets (#5)

* Correctly assign mindex on outgoing rtp packets

Previously mindex was not set and defaulted to zero. This lead to most packets
getting dropped because of sequence number reuse when streaming both audio and
video. This reorders the SDP entries for video and audio so that video is first
in both a video-only and a audio+video configuration. This means that the mindex
for video packets should always be zero, and for audio (if present) should
always be one. This assumes there will never be an audio-only configuration.

* Adjust comments

Co-authored-by: Michael Lynch <git@mtlynch.io>

* Add preprocessor conditional to guard packet.mindex setting

The mindex field wasn't added to the janus_plugin_rtp_packet until Janus 1.0, so this change adds a precompiler check to ensure JANUS_PLUGIN_API_VERSION is >= 100 before assigning a value to the mindex field.

* Preserve audio-then-video ordering for Janus 0.x

Co-authored-by: Louis Goessling <louis@goessling.com>

.bumpversion.cfg

@@ -1,7 +1,7 @@
 [bumpversion]
 commit = True
 tag = True
-current_version = 5.27
+current_version = 5.29
 parse = (?P<major>\d+)\.(?P<minor>\d+)
 serialize = 
 	{major}.{minor}

README.md

@@ -38,7 +38,7 @@ If you're going to live-stream from your backyard webcam and need to control it,
 You'll need  ```make```, ```gcc```, ```libevent``` with ```pthreads``` support, ```libjpeg9```/```libjpeg-turbo``` and ```libbsd``` (only for Linux).
 
 * Arch: `sudo pacman -S libevent libjpeg-turbo libutil-linux libbsd`.
-* Raspbian: `sudo apt install libevent-dev libjpeg8-dev libbsd-dev`. Add `libgpiod-dev` for `WITH_GPIO=1` and `libsystemd-dev` for `WITH_SYSTEMD=1`.
+* Raspbian: `sudo apt install libevent-dev libjpeg8-dev libbsd-dev`. Add `libgpiod-dev` for `WITH_GPIO=1` and `libsystemd-dev` for `WITH_SYSTEMD=1` and `libasound2-dev libspeex-dev libspeexdsp-dev libopus-dev` for `WITH_JANUS=1`.
 * Debian/Ubuntu: `sudo apt install build-essential libevent-dev libjpeg-dev libbsd-dev`.
 * Alpine: `sudo apk add libevent-dev libbsd-dev libjpeg-turbo-dev musl-dev`. Build with `WITH_PTHREAD_NP=0`.
 

janus/src/client.c

@@ -102,6 +102,11 @@ static void *_common_thread(void *v_client, bool video) {
 				packet.video = rtp->video;
 				packet.buffer = (char *)rtp->datagram;
 				packet.length = rtp->used;
+#				if JANUS_PLUGIN_API_VERSION >= 100
+				// The uStreamer Janus plugin places video in stream index 0 and audio
+				// (if available) in stream index 1.
+				packet.mindex = (rtp->video ? 0 : 1);
+#				endif
 				janus_plugin_rtp_extensions_reset(&packet.extensions);
 				// FIXME: Это очень эффективный способ уменьшить задержку, но WebRTC стек в хроме и фоксе
 				// слишком корявый, чтобы обработать это, из-за чего на кейфреймах начинаются заикания.

janus/src/plugin.c

@@ -154,9 +154,9 @@ static void *_video_sink_thread(UNUSED void *arg) {
 				if (frame == NULL) {
 					goto close_memsink;
 				}
-//				if (frame->key) {
-//					atomic_store(&_g_key_required, false);
-//				}
+				if (frame->key) {
+					atomic_store(&_g_key_required, false);
+				}
 				if (us_queue_put(_g_video_queue, frame, 0) != 0) {
 					_IF_NOT_REPORTED({ US_JLOG_PERROR("video", "Video queue is full"); });
 					us_frame_destroy(frame);
@@ -427,12 +427,7 @@ static struct janus_plugin_result *_plugin_handle_message(
 	} else if (!strcmp(request_str, "watch")) {
 		char *sdp;
 		{
-//			atomic_store(&_g_key_required, true);
 			char *const video_sdp = us_rtpv_make_sdp(_g_rtpv);
-			if (video_sdp == NULL) {
-				PUSH_ERROR(503, "Haven't received SPS/PPS from memsink yet");
-				goto ok_wait;
-			}
 			char *const audio_sdp = (_g_rtpa ? us_rtpa_make_sdp(_g_rtpa) : us_strdup(""));
 			US_ASPRINTF(sdp,
 				"v=0" RN
@@ -440,7 +435,16 @@ static struct janus_plugin_result *_plugin_handle_message(
 				"s=PiKVM uStreamer" RN
 				"t=0 0" RN
 				"%s%s",
-				us_get_now_id() >> 1, audio_sdp, video_sdp
+				us_get_now_id() >> 1,
+#				if JANUS_PLUGIN_API_VERSION >= 100
+				// Place video SDP before audio SDP so that the video and audio streams
+				// have predictable indices, even if audio is not available.
+				// See also client.c.
+				video_sdp, audio_sdp
+#				else
+				// For versions of Janus prior to 1.x, place the audio SDP first.
+				audio_sdp, video_sdp
+#				endif
 			);
 			free(audio_sdp);
 			free(video_sdp);
@@ -450,6 +454,10 @@ static struct janus_plugin_result *_plugin_handle_message(
 		PUSH_STATUS("started", offer_jsep);
 		json_decref(offer_jsep);
 
+	} else if (!strcmp(request_str, "key_required")) {
+		US_JLOG_INFO("main", "Got keyframe request from a client");
+		atomic_store(&_g_key_required, true);
+
 	} else {
 		PUSH_ERROR(405, "Not implemented");
 	}

janus/src/rtpv.c

@@ -36,35 +36,15 @@ us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback, bool zero_playout_delay) {
 	US_CALLOC(rtpv, 1);
 	rtpv->rtp = us_rtp_init(96, true, zero_playout_delay);
 	rtpv->callback = callback;
-	rtpv->sps = us_frame_init();
-	rtpv->pps = us_frame_init();
-	US_MUTEX_INIT(rtpv->mutex);
 	return rtpv;
 }
 
 void us_rtpv_destroy(us_rtpv_s *rtpv) {
-	US_MUTEX_DESTROY(rtpv->mutex);
-	us_frame_destroy(rtpv->pps);
-	us_frame_destroy(rtpv->sps);
 	us_rtp_destroy(rtpv->rtp);
 	free(rtpv);
 }
 
 char *us_rtpv_make_sdp(us_rtpv_s *rtpv) {
-	US_MUTEX_LOCK(rtpv->mutex);
-
-	if (rtpv->sps->used == 0 || rtpv->pps->used == 0) {
-		US_MUTEX_UNLOCK(rtpv->mutex);
-		return NULL;
-	}
-
-	char *sps = NULL;
-	char *pps = NULL;
-	us_base64_encode(rtpv->sps->data, rtpv->sps->used, &sps, NULL);
-	us_base64_encode(rtpv->pps->data, rtpv->pps->used, &pps, NULL);
-
-	US_MUTEX_UNLOCK(rtpv->mutex);
-
 #	define PAYLOAD rtpv->rtp->payload
 	// https://tools.ietf.org/html/rfc6184
 	// https://github.com/meetecho/janus-gateway/issues/2443
@@ -75,8 +55,6 @@ char *us_rtpv_make_sdp(us_rtpv_s *rtpv) {
 		"a=rtpmap:%u H264/90000" RN
 		"a=fmtp:%u profile-level-id=42E01F" RN
 		"a=fmtp:%u packetization-mode=1" RN
-		"a=fmtp:%u sprop-sps=%s" RN
-		"a=fmtp:%u sprop-pps=%s" RN
 		"a=rtcp-fb:%u nack" RN
 		"a=rtcp-fb:%u nack pli" RN
 		"a=rtcp-fb:%u goog-remb" RN
@@ -84,17 +62,12 @@ char *us_rtpv_make_sdp(us_rtpv_s *rtpv) {
 		"%s" // playout-delay
 		"a=sendonly" RN,
 		PAYLOAD, PAYLOAD, PAYLOAD, PAYLOAD,
-		PAYLOAD, sps,
-		PAYLOAD, pps,
 		PAYLOAD, PAYLOAD, PAYLOAD,
 		rtpv->rtp->ssrc,
 		(rtpv->rtp->zero_playout_delay ? "a=extmap:1 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay" RN : "")
 	);
-#	undef PAYLOAD
-
-	free(sps);
-	free(pps);
 	return sdp;
+#	undef PAYLOAD
 }
 
 #define _PRE 3 // Annex B prefix length
@@ -136,22 +109,11 @@ void us_rtpv_wrap(us_rtpv_s *rtpv, const us_frame_s *frame) {
 }
 
 void _rtpv_process_nalu(us_rtpv_s *rtpv, const uint8_t *data, size_t size, uint32_t pts, bool marked) {
+#	define DG rtpv->rtp->datagram
+
 	const unsigned ref_idc = (data[0] >> 5) & 3;
 	const unsigned type = data[0] & 0x1F;
 
-	us_frame_s *ps = NULL;
-	switch (type) {
-		case 7: ps = rtpv->sps; break;
-		case 8: ps = rtpv->pps; break;
-	}
-	if (ps != NULL) {
-		US_MUTEX_LOCK(rtpv->mutex);
-		us_frame_set_data(ps, data, size);
-		US_MUTEX_UNLOCK(rtpv->mutex);
-	}
-
-#	define DG rtpv->rtp->datagram
-
 	if (size + US_RTP_HEADER_SIZE <= US_RTP_DATAGRAM_SIZE) {
 		us_rtp_write_header(rtpv->rtp, pts, marked);
 		memcpy(DG + US_RTP_HEADER_SIZE, data, size);

janus/src/rtpv.h

@@ -31,12 +31,8 @@
 #include <sys/types.h>
 #include <linux/videodev2.h>
 
-#include <pthread.h>
-
 #include "uslibs/tools.h"
-#include "uslibs/threading.h"
 #include "uslibs/frame.h"
-#include "uslibs/base64.h"
 
 #include "rtp.h"
 
@@ -44,9 +40,6 @@
 typedef struct {
 	us_rtp_s			*rtp;
 	us_rtp_callback_f	callback;
-	us_frame_s			*sps; // Actually not a frame, just a bytes storage
-	us_frame_s			*pps;
-	pthread_mutex_t		mutex;
 } us_rtpv_s;
 
 

janus/src/uslibs/base64.c

@@ -1 +0,0 @@
-../../../src/libs/base64.c

janus/src/uslibs/base64.h

@@ -1 +0,0 @@
-../../../src/libs/base64.h

man/ustreamer-dump.1

@@ -1,6 +1,6 @@
 .\" Manpage for ustreamer-dump.
 .\" Open an issue or pull request to https://github.com/pikvm/ustreamer to correct errors or typos
-.TH USTREAMER-DUMP 1 "version 5.27" "January 2021"
+.TH USTREAMER-DUMP 1 "version 5.29" "January 2021"
 
 .SH NAME
 ustreamer-dump \- Dump uStreamer's memory sink to file
@@ -45,7 +45,7 @@ Limit the number of frames. Default: 0 (infinite).
 .BR \-i ", "\-\-interval\ \fIsec
 Delay between reading frames (float). Default: 0.
 .TP
-.BR \-k ", " \-\-key
+.BR \-k ", " \-\-key\-required
 Request keyframe from the sink. Default: disabled.
 
 .SS "Logging options"

man/ustreamer.1

@@ -1,6 +1,6 @@
 .\" Manpage for ustreamer.
 .\" Open an issue or pull request to https://github.com/pikvm/ustreamer to correct errors or typos
-.TH USTREAMER 1 "version 5.27" "November 2020"
+.TH USTREAMER 1 "version 5.29" "November 2020"
 
 .SH NAME
 ustreamer \- stream MJPEG video from any V4L2 device to the network

pkg/arch/PKGBUILD

@@ -3,7 +3,7 @@
 
 
 pkgname=ustreamer
-pkgver=5.27
+pkgver=5.29
 pkgrel=1
 pkgdesc="Lightweight and fast MJPEG-HTTP streamer"
 url="https://github.com/pikvm/ustreamer"

pkg/openwrt/Makefile

@@ -6,7 +6,7 @@
 include $(TOPDIR)/rules.mk
 
 PKG_NAME:=ustreamer
-PKG_VERSION:=5.27
+PKG_VERSION:=5.29
 PKG_RELEASE:=1
 PKG_MAINTAINER:=Maxim Devaev <mdevaev@gmail.com>
 

python/setup.py

@@ -17,7 +17,7 @@ def _find_sources(suffix: str) -> list[str]:
 if __name__ == "__main__":
     setup(
         name="ustreamer",
-        version="5.27",
+        version="5.29",
         description="uStreamer tools",
         author="Maxim Devaev",
         author_email="mdevaev@gmail.com",

src/dump/main.c

@@ -48,7 +48,7 @@ enum _OPT_VALUES {
 	_O_OUTPUT_JSON = 'j',
 	_O_COUNT = 'c',
 	_O_INTERVAL = 'i',
-	_O_KEY = 'k',
+	_O_KEY_REQUIRED = 'k',
 
 	_O_HELP = 'h',
 	_O_VERSION = 'v',
@@ -68,7 +68,7 @@ static const struct option _LONG_OPTS[] = {
 	{"output-json",			no_argument,		NULL,	_O_OUTPUT_JSON},
 	{"count",				required_argument,	NULL,	_O_COUNT},
 	{"interval",			required_argument,	NULL,	_O_INTERVAL},
-	{"key",					no_argument,		NULL,	_O_KEY},
+	{"key-required",		no_argument,		NULL,	_O_KEY_REQUIRED},
 
 	{"log-level",			required_argument,	NULL,	_O_LOG_LEVEL},
 	{"perf",				no_argument,		NULL,	_O_PERF},
@@ -154,7 +154,7 @@ int main(int argc, char *argv[]) {
 			case _O_OUTPUT_JSON:	OPT_SET(output_json, true);
 			case _O_COUNT:			OPT_NUMBER("--count", count, 0, LLONG_MAX, 0);
 			case _O_INTERVAL:		OPT_LDOUBLE("--interval", interval, 0, 60);
-			case _O_KEY:			OPT_SET(key_required, true);
+			case _O_KEY_REQUIRED:	OPT_SET(key_required, true);
 
 			case _O_LOG_LEVEL:			OPT_NUMBER("--log-level", us_g_log_level, US_LOG_LEVEL_INFO, US_LOG_LEVEL_DEBUG, 0);
 			case _O_PERF:				OPT_SET(us_g_log_level, US_LOG_LEVEL_PERF);
@@ -332,7 +332,7 @@ static void _help(FILE *fp) {
 	SAY("    -j|--output-json  ──────── Format output as JSON. Required option --output. Default: disabled.\n");
 	SAY("    -c|--count  <N>  ───────── Limit the number of frames. Default: 0 (infinite).\n");
 	SAY("    -i|--interval <sec>  ───── Delay between reading frames (float). Default: 0.\n");
-	SAY("    -k|--key  ──────────────── Request keyframe from the sink. Default: disabled.\n");
+	SAY("    -k|--key-required  ─────── Request keyframe from the sink. Default: disabled.\n");
 	SAY("Logging options:");
 	SAY("════════════════");
 	SAY("    --log-level <N>  ──── Verbosity level of messages from 0 (info) to 3 (debug).");

src/libs/const.h

@@ -23,7 +23,7 @@
 #pragma once
 
 #define US_VERSION_MAJOR 5
-#define US_VERSION_MINOR 27
+#define US_VERSION_MINOR 29
 
 #define US_MAKE_VERSION2(_major, _minor) #_major "." #_minor
 #define US_MAKE_VERSION1(_major, _minor) US_MAKE_VERSION2(_major, _minor)