Bump version: 6.59 → 6.60

.bumpversion.cfg

@@ -1,7 +1,7 @@
 [bumpversion]
 commit = True
 tag = True
-current_version = 6.58
+current_version = 6.60
 parse = (?P<major>\d+)\.(?P<minor>\d+)
 serialize = 
 	{major}.{minor}

.gitignore

@@ -6,6 +6,7 @@
 /python/dist/
 /python/ustreamer.egg-info/
 /python/root/
+/python/build/
 /janus/build/
 /ustreamer
 /ustreamer-*

janus/src/client.c

@@ -55,6 +55,9 @@ us_janus_client_s *us_janus_client_init(janus_callbacks *gw, janus_plugin_sessio
 	US_CALLOC(client, 1);
 	client->gw = gw;
 	client->session = session;
+	client->video_ssrc = us_get_ssrc();
+	client->audio_ssrc = us_get_ssrc();
+
 	atomic_init(&client->transmit, false);
 	atomic_init(&client->transmit_acap, false);
 	atomic_init(&client->transmit_aplay, false);
@@ -181,6 +184,8 @@ static void *_video_or_acap_thread(void *v_client, bool video) {
 			atomic_load(&client->transmit)
 			&& (video || atomic_load(&client->transmit_acap))
 		) {
+			us_rtp_write_ssrc(&rtp, (rtp.video ? client->video_ssrc : client->audio_ssrc));
+
 			janus_plugin_rtp packet = {
 				.video = rtp.video,
 				.buffer = (char*)rtp.datagram,

janus/src/client.h

@@ -37,6 +37,9 @@
 typedef struct {
 	janus_callbacks			*gw;
 	janus_plugin_session	*session;
+	u32						video_ssrc;
+	u32						audio_ssrc;
+
 	atomic_bool				transmit;
 	atomic_bool				transmit_acap;
 	atomic_bool				transmit_aplay;

janus/src/config.c

@@ -68,8 +68,9 @@ us_config_s *us_config_init(const char *config_dir_path) {
 			US_JLOG_ERROR("config", "Either acap.sampling_rate or acap.tc358743 required");
 			goto error;
 		}
-		config->aplay_dev_name = _get_value(jcfg, "aplay", "device");
 	}
+	config->aplay_dev_name = _get_value(jcfg, "aplay", "device");
+	config->vplay_sink_name = _get_value(jcfg, "vplay", "sink");
 
 	goto ok;
 
@@ -87,6 +88,7 @@ void us_config_destroy(us_config_s *config) {
 	US_DELETE(config->acap_dev_name, free);
 	US_DELETE(config->tc358743_dev_path, free);
 	US_DELETE(config->aplay_dev_name, free);
+	US_DELETE(config->vplay_sink_name, free);
 	free(config);
 }
 

janus/src/config.h

@@ -34,6 +34,7 @@ typedef struct {
 	char	*tc358743_dev_path;
 
 	char	*aplay_dev_name;
+	char	*vplay_sink_name;
 } us_config_s;
 
 

janus/src/plugin.c

@@ -345,7 +345,7 @@ static void *_aplay_thread(void *arg) {
 				}
 			}
 
-			if (!_HAS_WATCHERS || !_HAS_LISTENERS || !_HAS_SPEAKERS) {
+			if (!_HAS_WATCHERS || !_HAS_SPEAKERS) {
 				goto close_aplay;
 			}
 
@@ -443,9 +443,9 @@ static int _plugin_init(janus_callbacks *gw, const char *config_dir_path) {
 	if (_g_config->acap_dev_name != NULL) {
 		_g_rtpa = us_rtpa_init(_relay_rtp_clients);
 		US_THREAD_CREATE(_g_acap_tid, _acap_thread, NULL);
-		if (_g_config->aplay_dev_name != NULL) {
+	}
-			US_THREAD_CREATE(_g_aplay_tid, _aplay_thread, NULL);
+	if (_g_config->aplay_dev_name != NULL) {
-		}
+		US_THREAD_CREATE(_g_aplay_tid, _aplay_thread, NULL);
 	}
 	US_THREAD_CREATE(_g_video_rtp_tid, _video_rtp_thread, NULL);
 	US_THREAD_CREATE(_g_video_sink_tid, _video_sink_thread, NULL);
@@ -612,21 +612,20 @@ static struct janus_plugin_result *_plugin_handle_message(
 		uint video_orient = 0;
 		bool with_acap = false;
 		bool with_aplay = false;
+		bool with_vplay = false;
 		{
 			json_t *const params = json_object_get(msg, "params");
 			if (params != NULL) {
-				{
+#				define READ_BOOL(x_target, x_key, x_cond) { \
-					json_t *const obj = json_object_get(params, "audio");
+						json_t *const m_obj = json_object_get(params, x_key); \
-					if (obj != NULL && json_is_boolean(obj)) {
+						if (m_obj != NULL && json_is_boolean(m_obj)) { \
-						with_acap = (us_au_probe(_g_config->acap_dev_name) && json_boolean_value(obj));
+							x_target = (json_boolean_value(m_obj) && (x_cond)); \
+						} \
 					}
-				}
+				READ_BOOL(with_acap, "audio", us_au_probe(_g_config->acap_dev_name));
-				{
+				READ_BOOL(with_aplay, "mic", us_au_probe(_g_config->aplay_dev_name));
-					json_t *const obj = json_object_get(params, "mic");
+				READ_BOOL(with_vplay, "camera", (_g_config->vplay_sink_name != NULL));
-					if (obj != NULL && json_is_boolean(obj)) {
+#				undef READ_BOOL
-						with_aplay = (us_au_probe(_g_config->aplay_dev_name) && json_boolean_value(obj));
-					}
-				}
 				{
 					json_t *const obj = json_object_get(params, "orientation");
 					if (obj != NULL && json_is_integer(obj)) {
@@ -640,23 +639,23 @@ static struct janus_plugin_result *_plugin_handle_message(
 			}
 		}
 
-		{
-			char *const sdp = us_sdp_create(
-				_g_rtpv,
-				(with_acap ? _g_rtpa : NULL),
-				(with_acap && with_aplay));
-			json_t *const offer_jsep = json_pack("{ssss}", "type", "offer", "sdp", sdp);
-			PUSH_STATUS("started", NULL, offer_jsep);
-			json_decref(offer_jsep);
-			free(sdp);
-		}
-
 		{
 			_LOCK_ALL;
 			bool has_listeners = false;
 			bool has_speakers = false;
 			US_LIST_ITERATE(_g_clients, client, {
 				if (client->session == session) {
+					char *const sdp = us_sdp_create(
+						client->video_ssrc,
+						client->audio_ssrc,
+						with_acap,
+						with_aplay,
+						with_vplay);
+					json_t *const offer_jsep = json_pack("{ssss}", "type", "offer", "sdp", sdp);
+					PUSH_STATUS("started", NULL, offer_jsep);
+					json_decref(offer_jsep);
+					free(sdp);
+
 					atomic_store(&client->transmit_acap, with_acap);
 					atomic_store(&client->transmit_aplay, with_aplay);
 					atomic_store(&client->video_orient, video_orient);
@@ -671,11 +670,10 @@ static struct janus_plugin_result *_plugin_handle_message(
 
 	} else if (!strcmp(request_str, "features")) {
 		const char *const ice_url = getenv("JANUS_USTREAMER_WEB_ICE_URL");
-		const bool acap_avail = us_au_probe(_g_config->acap_dev_name);
 		json_t *const features = json_pack(
 			"{s:b, s:b, s:{s:s?}}",
-			"audio", acap_avail,
+			"audio", us_au_probe(_g_config->acap_dev_name),
-			"mic", (acap_avail && us_au_probe(_g_config->aplay_dev_name)),
+			"mic", us_au_probe(_g_config->aplay_dev_name),
 			"ice", "url", (ice_url != NULL ? ice_url : default_ice_url)
 		);
 		PUSH_STATUS("features", features, NULL);

janus/src/rtp.c

@@ -44,9 +44,11 @@ void us_rtp_destroy(us_rtp_s *rtp) {
 void us_rtp_assign(us_rtp_s *rtp, uint payload, bool video) {
 	rtp->payload = payload;
 	rtp->video = video;
-	rtp->ssrc = us_triple_u32(us_get_now_monotonic_u64());
 }
 
+#define WRITE_BE_U32(x_offset, x_value) \
+	*((u32*)(rtp->datagram + x_offset)) = __builtin_bswap32(x_value)
+
 void us_rtp_write_header(us_rtp_s *rtp, u32 pts, bool last_header) {
 	u32 word0 = 0x80000000;
 	if (last_header) {
@@ -56,10 +58,13 @@ void us_rtp_write_header(us_rtp_s *rtp, u32 pts, bool last_header) {
 	word0 |= rtp->seq;
 	++rtp->seq;
 
-#	define WRITE_BE_U32(x_offset, x_value) \
-		*((u32*)(rtp->datagram + x_offset)) = __builtin_bswap32(x_value)
 	WRITE_BE_U32(0, word0);
 	WRITE_BE_U32(4, pts);
-	WRITE_BE_U32(8, rtp->ssrc);
+	WRITE_BE_U32(8, 0); // SSRC should be written separately for each client
-#	undef WRITE_BE_U32
 }
+
+void us_rtp_write_ssrc(us_rtp_s *rtp, u32 ssrc) {
+	WRITE_BE_U32(8, ssrc);
+}
+
+#undef WRITE_BE_U32

janus/src/rtp.h

@@ -23,6 +23,7 @@
 #pragma once
 
 #include "uslibs/types.h"
+#include "uslibs/tools.h"
 
 
 // Max RTP size for WebRTC is 1200 bytes:
@@ -47,7 +48,6 @@
 typedef struct {
 	uint	payload;
 	bool	video;
-	u32		ssrc;
 
 	u16		seq;
 	u8		datagram[US_RTP_TOTAL_SIZE];
@@ -62,8 +62,13 @@ typedef struct {
 typedef void (*us_rtp_callback_f)(const us_rtp_s *rtp);
 
 
+static inline u32 us_get_ssrc(void) {
+	return us_triple_u32(us_get_now_monotonic_u64());
+}
+
 us_rtp_s *us_rtp_init(void);
 void us_rtp_destroy(us_rtp_s *rtp);
 
 void us_rtp_assign(us_rtp_s *rtp, uint payload, bool video);
 void us_rtp_write_header(us_rtp_s *rtp, u32 pts, bool last_header);
+void us_rtp_write_ssrc(us_rtp_s *rtp, u32 ssrc);

janus/src/sdp.c

@@ -34,12 +34,12 @@
 #include "rtpa.h"
 
 
-char *us_sdp_create(us_rtpv_s *rtpv, us_rtpa_s *rtpa, bool mic) {
+char *us_sdp_create(u32 video_ssrc, u32 audio_ssrc, bool acap, bool aplay, bool vplay) {
 	char *video_sdp;
 	{
 		// https://tools.ietf.org/html/rfc6184
 		// https://github.com/meetecho/janus-gateway/issues/2443
-		const uint pl = rtpv->rtp->payload;
+		const uint pl = US_RTP_H264_PAYLOAD;
 		US_ASPRINTF(
 			video_sdp,
 			"m=video 1 RTP/SAVPF %u" RN
@@ -56,16 +56,25 @@ char *us_sdp_create(us_rtpv_s *rtpv, us_rtpa_s *rtpa, bool mic) {
 			"a=extmap:1/sendonly urn:3gpp:video-orientation" RN
 			"a=extmap:2/sendonly http://www.webrtc.org/experiments/rtp-hdrext/playout-delay" RN
 			"a=extmap:3/sendonly http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time" RN
-			"a=sendonly" RN,
+			"a=%s" RN,
 			pl, pl, pl, pl, pl, pl, pl,
-			rtpv->rtp->ssrc);
+			video_ssrc,
+			(vplay ? "sendrecv" : "sendonly"));
 	}
 
 	char *audio_sdp;
-	if (rtpa == NULL) {
+	if (acap || aplay) {
-		audio_sdp = us_strdup("");
+		const uint pl = US_RTP_OPUS_PAYLOAD;
-	} else {
+
-		const uint pl = rtpa->rtp->payload;
+		char *dir = "";
+		if (acap && aplay) {
+			dir = "sendrecv";
+		} else if (acap && !aplay) {
+			dir = "sendonly";
+		} else if (!acap && aplay) {
+			dir = "recvonly";
+		}
+
 		US_ASPRINTF(
 			audio_sdp,
 			"m=audio 1 RTP/SAVPF %u" RN
@@ -79,8 +88,10 @@ char *us_sdp_create(us_rtpv_s *rtpv, us_rtpa_s *rtpa, bool mic) {
 			pl, pl,
 			US_RTP_OPUS_HZ, US_RTP_OPUS_CH,
 			pl,
-			rtpa->rtp->ssrc,
+			audio_ssrc,
-			(mic ? "sendrecv" : "sendonly"));
+			dir);
+	} else {
+		audio_sdp = us_strdup("");
 	}
 
 	char *sdp;

janus/src/sdp.h

@@ -24,8 +24,5 @@
 
 #include "uslibs/types.h"
 
-#include "rtpv.h"
-#include "rtpa.h"
 
-
+char *us_sdp_create(u32 video_ssrc, u32 audio_ssrc, bool acap, bool aplay, bool vplay);
-char *us_sdp_create(us_rtpv_s *rtpv, us_rtpa_s *rtpa, bool mic);

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 6.58" "January 2021"
+.TH USTREAMER-DUMP 1 "version 6.60" "January 2021"
 
 .SH NAME
 ustreamer-dump \- Dump uStreamer's memory sink to file

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 6.58" "November 2020"
+.TH USTREAMER 1 "version 6.60" "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=6.58
+pkgver=6.60
 pkgrel=1
 pkgdesc="Lightweight and fast MJPEG-HTTP streamer"
 url="https://github.com/pikvm/ustreamer"

pkg/openwrt/Makefile

@@ -9,7 +9,7 @@
 include $(TOPDIR)/rules.mk
 
 PKG_NAME:=ustreamer
-PKG_VERSION:=6.58
+PKG_VERSION:=6.60
 PKG_RELEASE:=1
 PKG_MAINTAINER:=Maxim Devaev <mdevaev@gmail.com>
 

python/Makefile

@@ -17,4 +17,4 @@ install:
 
 
 clean:
-	rm -rf root dist ustreamer.egg-info
+	rm -rf root dist build ustreamer.egg-info

python/setup.py

@@ -34,7 +34,7 @@ def main() -> None:
     flags = _find_flags()
     setup(
         name="ustreamer",
-        version="6.58",
+        version="6.60",
         description="uStreamer tools",
         author="Maxim Devaev",
         author_email="mdevaev@gmail.com",

src/libs/const.h

@@ -26,7 +26,7 @@
 
 
 #define US_VERSION_MAJOR 6
-#define US_VERSION_MINOR 58
+#define US_VERSION_MINOR 60
 
 #define US_MAKE_VERSION2(_major, _minor) #_major "." #_minor
 #define US_MAKE_VERSION1(_major, _minor) US_MAKE_VERSION2(_major, _minor)