Bump version: 5.35 → 5.36

.bumpversion.cfg

@@ -1,7 +1,7 @@
 [bumpversion]
 commit = True
 tag = True
-current_version = 5.31
+current_version = 5.36
 parse = (?P<major>\d+)\.(?P<minor>\d+)
 serialize = 
 	{major}.{minor}

.github/workflows/docker-alpine-image.yaml

@@ -35,7 +35,7 @@ jobs:
         uses: docker/metadata-action@v4
         with:
           images: |
-            ${{ secrets.DOCKERHUB_USERNAME }}/ustreamer,enable=${{ startsWith(github.ref, 'refs/tags/v') }}
+            ${{ secrets.DOCKERHUB_REPO }}/ustreamer,enable=${{ startsWith(github.ref, 'refs/tags/v') }}
             ghcr.io/${{ github.repository }},enable=${{ github.event_name != 'pull_request' }}
           tags: |
             type=ref,event=tag

README.md

@@ -34,16 +34,31 @@ Footnotes:
 If you're going to live-stream from your backyard webcam and need to control it, use mjpg-streamer. If you need a high-quality image with high FPS - µStreamer for the win.
 
 -----
-# Building
+# Installation
+
+## Building
+You need to download the µStreamer onto your system and build it from the sources.
+
+AUR has a package for Arch Linux: https://aur.archlinux.org/packages/ustreamer.
+
+FreeBSD port: https://www.freshports.org/multimedia/ustreamer.
+
+### Preconditions
 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` and `libasound2-dev libspeex-dev libspeexdsp-dev libopus-dev` for `WITH_JANUS=1`.
+* Raspbian: `sudo apt install libevent-dev libjpeg9-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`.
 
 To enable GPIO support install [libgpiod](https://git.kernel.org/pub/scm/libs/libgpiod/libgpiod.git/about) and pass option ```WITH_GPIO=1```. If the compiler reports about a missing function ```pthread_get_name_np()``` (or similar), add option ```WITH_PTHREAD_NP=0``` (it's enabled by default). For the similar error with ```setproctitle()``` add option ```WITH_SETPROCTITLE=0```.
 
+> **Note**
+> Raspian: In case your version of Raspian is too told for there to be a libjpeg9 package, use `libjpeg8-dev` instead: `E: Package 'libjpeg9-dev' has no installation candidate`.
+
+### Make
+The most convenient process is to clone the µStreamer Git repository onto your system. If you don't have Git installed and don't want to install it either, you can download and unzip the sources from GitHub using `wget https://github.com/pikvm/ustreamer/archive/refs/heads/master.zip`.
+
 ```
 $ git clone --depth=1 https://github.com/pikvm/ustreamer
 $ cd ustreamer
@@ -51,9 +66,15 @@ $ make
 $ ./ustreamer --help
 ```
 
-AUR has a package for Arch Linux: https://aur.archlinux.org/packages/ustreamer.
+## Update
+Assuming you have a µStreamer clone as discussed above you can update µStreamer as follows.
 
-FreeBSD port: https://www.freshports.org/multimedia/ustreamer.
+```
+$ cd ustreamer
+$ git pull
+$ make clean
+$ make
+```
 
 -----
 # Usage

docs/h264.md

@@ -189,7 +189,7 @@ The client-side JavaScript application uses the following control flow:
         success: function (pluginHandle) {
           uStreamerPluginHandle = pluginHandle;
           // Instruct the µStreamer Janus plugin to initiate streaming.
-          uStreamerPluginHandle.send({ message: { request: "watch" } });
+          uStreamerPluginHandle.send({ message: { request: "watch", params: {audio: true} } });
         },
 
         // Callback function if the server fails to attach the plugin.
@@ -197,13 +197,6 @@ The client-side JavaScript application uses the following control flow:
 
         // Callback function for processing messages from the Janus server.
         onmessage: function (msg, jsepOffer) {
-          // 503 indicates that the plugin is not ready to stream yet. Retry the
-          // watch request until the video stream is available.
-          if (msg.error_code === 503) {
-            uStreamerPluginHandle.send({ message: { request: "watch" } });
-            return;
-          }
-
           // If there is a JSEP offer, respond to it. This starts the WebRTC
           // connection.
           if (jsepOffer) {

janus/src/audio.c

@@ -55,6 +55,19 @@ static void *_pcm_thread(void *v_audio);
 static void *_encoder_thread(void *v_audio);
 
 
+bool us_audio_probe(const char *name) {
+	snd_pcm_t *pcm;
+	int err;
+	US_JLOG_INFO("audio", "Probing PCM capture ...");
+	if ((err = snd_pcm_open(&pcm, name, SND_PCM_STREAM_CAPTURE, 0)) < 0) {
+		_JLOG_PERROR_ALSA(err, "audio", "Can't probe PCM capture");
+		return false;
+	}
+	snd_pcm_close(pcm);
+	US_JLOG_INFO("audio", "PCM capture is available");
+	return true;
+}
+
 us_audio_s *us_audio_init(const char *name, unsigned pcm_hz) {
 	us_audio_s *audio;
 	US_CALLOC(audio, 1);

janus/src/audio.h

@@ -52,8 +52,8 @@ typedef struct {
 	SpeexResamplerState	*res;
 	OpusEncoder			*enc;
 
-	us_queue_s				*pcm_queue;
-	us_queue_s				*enc_queue;
+	us_queue_s			*pcm_queue;
+	us_queue_s			*enc_queue;
 	uint32_t			pts;
 
 	pthread_t			pcm_tid;
@@ -63,6 +63,8 @@ typedef struct {
 } us_audio_s;
 
 
+bool us_audio_probe(const char *name);
+
 us_audio_s *us_audio_init(const char *name, unsigned pcm_hz);
 void us_audio_destroy(us_audio_s *audio);
 

janus/src/client.c

@@ -28,53 +28,50 @@ static void *_audio_thread(void *v_client);
 static void *_common_thread(void *v_client, bool video);
 
 
-us_janus_client_s *us_janus_client_init(janus_callbacks *gw, janus_plugin_session *session, bool has_audio) {
+us_janus_client_s *us_janus_client_init(janus_callbacks *gw, janus_plugin_session *session) {
 	us_janus_client_s *client;
 	US_CALLOC(client, 1);
 	client->gw = gw;
 	client->session = session;
-	atomic_init(&client->transmit, true);
+	atomic_init(&client->transmit, false);
+	atomic_init(&client->transmit_audio, false);
 
 	atomic_init(&client->stop, false);
 
 	client->video_queue = us_queue_init(1024);
 	US_THREAD_CREATE(client->video_tid, _video_thread, client);
 
-	if (has_audio) {
-		client->audio_queue = us_queue_init(64);
-		US_THREAD_CREATE(client->audio_tid, _audio_thread, client);
-	}
+	client->audio_queue = us_queue_init(64);
+	US_THREAD_CREATE(client->audio_tid, _audio_thread, client);
+
 	return client;
 }
 
 void us_janus_client_destroy(us_janus_client_s *client) {
 	atomic_store(&client->stop, true);
 	us_queue_put(client->video_queue, NULL, 0);
-	if (client->audio_queue != NULL) {
-		us_queue_put(client->audio_queue, NULL, 0);
-	}
+	us_queue_put(client->audio_queue, NULL, 0);
 
 	US_THREAD_JOIN(client->video_tid);
 	US_QUEUE_DELETE_WITH_ITEMS(client->video_queue, us_rtp_destroy);
-	if (client->audio_queue != NULL) {
-		US_THREAD_JOIN(client->audio_tid);
-		US_QUEUE_DELETE_WITH_ITEMS(client->audio_queue, us_rtp_destroy);
-	}
+
+	US_THREAD_JOIN(client->audio_tid);
+	US_QUEUE_DELETE_WITH_ITEMS(client->audio_queue, us_rtp_destroy);
+
 	free(client);
 }
 
 void us_janus_client_send(us_janus_client_s *client, const us_rtp_s *rtp) {
 	if (
-		!atomic_load(&client->transmit)
-		|| (!rtp->video && client->audio_queue == NULL)
+		atomic_load(&client->transmit)
+		&& (rtp->video || atomic_load(&client->transmit_audio))
 	) {
-		return;
-	}
-	us_rtp_s *const new = us_rtp_dup(rtp);
-	if (us_queue_put((new->video ? client->video_queue : client->audio_queue), new, 0) != 0) {
-		US_JLOG_ERROR("client", "Session %p %s queue is full",
-			client->session, (new->video ? "video" : "audio"));
-		us_rtp_destroy(new);
+		us_rtp_s *const new = us_rtp_dup(rtp);
+		if (us_queue_put((new->video ? client->video_queue : client->audio_queue), new, 0) != 0) {
+			US_JLOG_ERROR("client", "Session %p %s queue is full",
+				client->session, (new->video ? "video" : "audio"));
+			us_rtp_destroy(new);
+		}
 	}
 }
 
@@ -97,7 +94,11 @@ static void *_common_thread(void *v_client, bool video) {
 			if (rtp == NULL) {
 				break;
 			}
-			if (atomic_load(&client->transmit)) {
+
+			if (
+				atomic_load(&client->transmit)
+				&& (video || atomic_load(&client->transmit_audio))
+			) {
 				janus_plugin_rtp packet = {0};
 				packet.video = rtp->video;
 				packet.buffer = (char *)rtp->datagram;
@@ -107,14 +108,17 @@ static void *_common_thread(void *v_client, bool video) {
 				// (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.
-				//   - https://github.com/Glimesh/janus-ftl-plugin/issues/101
-				if (rtp->zero_playout_delay) {
+				/*if (rtp->zero_playout_delay) {
+					// https://github.com/pikvm/pikvm/issues/784
 					packet.extensions.min_delay = 0;
 					packet.extensions.max_delay = 0;
-				}
+				} else {
+					packet.extensions.min_delay = 0;
+					packet.extensions.max_delay = 1000;
+				}*/
+
 				client->gw->relay_rtp(client->session, &packet);
 			}
 			us_rtp_destroy(rtp);

janus/src/client.h

@@ -41,21 +41,22 @@
 
 typedef struct us_janus_client_sx {
 	janus_callbacks			*gw;
-    janus_plugin_session	*session;
-    atomic_bool				transmit;
+	janus_plugin_session	*session;
+	atomic_bool				transmit;
+	atomic_bool				transmit_audio;
 
-	pthread_t	video_tid;
-	pthread_t	audio_tid;
-	atomic_bool	stop;
+	pthread_t				video_tid;
+	pthread_t				audio_tid;
+	atomic_bool				stop;
 
-	us_queue_s		*video_queue;
-	us_queue_s		*audio_queue;
+	us_queue_s				*video_queue;
+	us_queue_s				*audio_queue;
 
     US_LIST_STRUCT(struct us_janus_client_sx);
 } us_janus_client_s;
 
 
-us_janus_client_s *us_janus_client_init(janus_callbacks *gw, janus_plugin_session *session, bool has_audio);
+us_janus_client_s *us_janus_client_init(janus_callbacks *gw, janus_plugin_session *session);
 void us_janus_client_destroy(us_janus_client_s *client);
 
 void us_janus_client_send(us_janus_client_s *client, const us_rtp_s *rtp);

janus/src/config.c

@@ -24,7 +24,7 @@
 
 
 static char *_get_value(janus_config *jcfg, const char *section, const char *option);
-static bool _get_bool(janus_config *jcfg, const char *section, const char *option, bool def);
+// static bool _get_bool(janus_config *jcfg, const char *section, const char *option, bool def);
 
 
 us_config_s *us_config_init(const char *config_dir_path) {
@@ -51,11 +51,7 @@ us_config_s *us_config_init(const char *config_dir_path) {
 		US_JLOG_ERROR("config", "Missing config value: video.sink (ex. memsink.object)");
 		goto error;
 	}
-	if ((config->video_zero_playout_delay = _get_bool(jcfg, "video", "zero_playout_delay", false)) == true) {
-		US_JLOG_INFO("config", "Enabled the experimental Playout-Delay=0 RTP extension for VIDEO");
-	}
 	if ((config->audio_dev_name = _get_value(jcfg, "audio", "device")) != NULL) {
-		US_JLOG_INFO("config", "Enabled the experimental AUDIO feature");
 		if ((config->tc358743_dev_path = _get_value(jcfg, "audio", "tc358743")) == NULL) {
 			US_JLOG_INFO("config", "Missing config value: audio.tc358743");
 			goto error;
@@ -88,7 +84,7 @@ static char *_get_value(janus_config *jcfg, const char *section, const char *opt
 	return us_strdup(option_obj->value);
 }
 
-static bool _get_bool(janus_config *jcfg, const char *section, const char *option, bool def) {
+/*static bool _get_bool(janus_config *jcfg, const char *section, const char *option, bool def) {
 	char *const tmp = _get_value(jcfg, section, option);
 	bool value = def;
 	if (tmp != NULL) {
@@ -96,4 +92,4 @@ static bool _get_bool(janus_config *jcfg, const char *section, const char *optio
 		free(tmp);
 	}
 	return value;
-}
+}*/

janus/src/config.h

@@ -36,7 +36,6 @@
 
 typedef struct {
 	char	*video_sink_name;
-	bool	video_zero_playout_delay;
 
 	char	*audio_dev_name;
 	char	*tc358743_dev_path;

janus/src/logging.h

@@ -36,3 +36,11 @@
 		JANUS_LOG(LOG_ERR, "[%s/%-9s] " x_msg ": %s\n", US_PLUGIN_NAME, x_prefix, ##__VA_ARGS__, m_perror_str); \
 		free(m_perror_str); \
 	}
+
+#define US_ONCE(...) { \
+		const int m_reported = __LINE__; \
+		if (m_reported != once) { \
+			__VA_ARGS__; \
+			once = m_reported; \
+		} \
+	}

janus/src/plugin.c

@@ -77,6 +77,7 @@ static pthread_mutex_t	_g_audio_lock = PTHREAD_MUTEX_INITIALIZER;
 static atomic_bool		_g_ready = false;
 static atomic_bool		_g_stop = false;
 static atomic_bool		_g_has_watchers = false;
+static atomic_bool		_g_has_listeners = false;
 static atomic_bool		_g_key_required = false;
 
 
@@ -92,16 +93,12 @@ static atomic_bool		_g_key_required = false;
 #define _READY			atomic_load(&_g_ready)
 #define _STOP			atomic_load(&_g_stop)
 #define _HAS_WATCHERS	atomic_load(&_g_has_watchers)
+#define _HAS_LISTENERS	atomic_load(&_g_has_listeners)
 
 
 janus_plugin *create(void);
 
 
-#define _IF_NOT_REPORTED(...) { \
-		const unsigned _error_code = __LINE__; \
-		if (error_reported != _error_code) { __VA_ARGS__; error_reported = _error_code; } \
-	}
-
 static void *_video_rtp_thread(UNUSED void *arg) {
 	US_THREAD_RENAME("us_video_rtp");
 	atomic_store(&_g_video_rtp_tid_created, true);
@@ -123,11 +120,11 @@ static void *_video_sink_thread(UNUSED void *arg) {
 	atomic_store(&_g_video_sink_tid_created, true);
 
 	uint64_t frame_id = 0;
-	unsigned error_reported = 0;
+	int once = 0;
 
 	while (!_STOP) {
 		if (!_HAS_WATCHERS) {
-			_IF_NOT_REPORTED({ US_JLOG_INFO("video", "No active watchers, memsink disconnected"); });
+			US_ONCE({ US_JLOG_INFO("video", "No active watchers, memsink disconnected"); });
 			usleep(_g_watchers_polling);
 			continue;
 		}
@@ -136,16 +133,16 @@ static void *_video_sink_thread(UNUSED void *arg) {
 		us_memsink_shared_s *mem = NULL;
 
 		if ((fd = shm_open(_g_config->video_sink_name, O_RDWR, 0)) <= 0) {
-			_IF_NOT_REPORTED({ US_JLOG_PERROR("video", "Can't open memsink"); });
+			US_ONCE({ US_JLOG_PERROR("video", "Can't open memsink"); });
 			goto close_memsink;
 		}
 
 		if ((mem = us_memsink_shared_map(fd)) == NULL) {
-			_IF_NOT_REPORTED({ US_JLOG_PERROR("video", "Can't map memsink"); });
+			US_ONCE({ US_JLOG_PERROR("video", "Can't map memsink"); });
 			goto close_memsink;
 		}
 
-		error_reported = 0;
+		once = 0;
 
 		US_JLOG_INFO("video", "Memsink opened; reading frames ...");
 		while (!_STOP && _HAS_WATCHERS) {
@@ -159,7 +156,7 @@ static void *_video_sink_thread(UNUSED void *arg) {
 					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_ONCE({ US_JLOG_PERROR("video", "Video queue is full"); });
 					us_frame_destroy(frame);
 				}
 			} else if (result == -1) {
@@ -186,10 +183,10 @@ static void *_audio_thread(UNUSED void *arg) {
 	assert(_g_config->audio_dev_name != NULL);
 	assert(_g_config->tc358743_dev_path != NULL);
 
-	unsigned error_reported = 0;
+	int once = 0;
 
 	while (!_STOP) {
-		if (!_HAS_WATCHERS) {
+		if (!_HAS_WATCHERS || !_HAS_LISTENERS) {
 			usleep(_g_watchers_polling);
 			continue;
 		}
@@ -201,17 +198,17 @@ static void *_audio_thread(UNUSED void *arg) {
 			goto close_audio;
 		}
 		if (!info.has_audio) {
-			_IF_NOT_REPORTED({ US_JLOG_INFO("audio", "No audio presented from the host"); });
+			US_ONCE({ US_JLOG_INFO("audio", "No audio presented from the host"); });
 			goto close_audio;
 		}
-		_IF_NOT_REPORTED({ US_JLOG_INFO("audio", "Detected host audio"); });
+		US_ONCE({ US_JLOG_INFO("audio", "Detected host audio"); });
 		if ((audio = us_audio_init(_g_config->audio_dev_name, info.audio_hz)) == NULL) {
 			goto close_audio;
 		}
 
-		error_reported = 0;
+		once = 0;
 
-		while (!_STOP && _HAS_WATCHERS) {
+		while (!_STOP && _HAS_WATCHERS && _HAS_LISTENERS) {
 			if (
 				us_tc358743_read_info(_g_config->tc358743_dev_path, &info) < 0
 				|| !info.has_audio
@@ -240,8 +237,6 @@ static void *_audio_thread(UNUSED void *arg) {
 	return NULL;
 }
 
-#undef _IF_NOT_REPORTED
-
 static void _relay_rtp_clients(const us_rtp_s *rtp) {
 	US_LIST_ITERATE(_g_clients, client, {
 		us_janus_client_send(client, rtp);
@@ -262,8 +257,8 @@ static int _plugin_init(janus_callbacks *gw, const char *config_dir_path) {
 	_g_gw = gw;
 
 	_g_video_queue = us_queue_init(1024);
-	_g_rtpv = us_rtpv_init(_relay_rtp_clients, _g_config->video_zero_playout_delay);
-	if (_g_config->audio_dev_name != NULL) {
+	_g_rtpv = us_rtpv_init(_relay_rtp_clients);
+	if (_g_config->audio_dev_name != NULL && us_audio_probe(_g_config->audio_dev_name)) {
 		_g_rtpa = us_rtpa_init(_relay_rtp_clients);
 		US_THREAD_CREATE(_g_audio_tid, _audio_thread, NULL);
 	}
@@ -302,7 +297,7 @@ static void _plugin_create_session(janus_plugin_session *session, int *err) {
 	_IF_DISABLED({ *err = -1; return; });
 	_LOCK_ALL;
 	US_JLOG_INFO("main", "Creating session %p ...", session);
-	us_janus_client_s *const client = us_janus_client_init(_g_gw, session, (_g_config->audio_dev_name != NULL));
+	us_janus_client_s *const client = us_janus_client_init(_g_gw, session);
 	US_LIST_APPEND(_g_clients, client);
 	atomic_store(&_g_has_watchers, true);
 	_UNLOCK_ALL;
@@ -313,6 +308,7 @@ static void _plugin_destroy_session(janus_plugin_session* session, int *err) {
 	_LOCK_ALL;
 	bool found = false;
 	bool has_watchers = false;
+	bool has_listeners = false;
 	US_LIST_ITERATE(_g_clients, client, {
 		if (client->session == session) {
 			US_JLOG_INFO("main", "Removing session %p ...", session);
@@ -321,6 +317,7 @@ static void _plugin_destroy_session(janus_plugin_session* session, int *err) {
 			found = true;
 		} else {
 			has_watchers = (has_watchers || atomic_load(&client->transmit));
+			has_listeners = (has_listeners || atomic_load(&client->transmit_audio));
 		}
 	});
 	if (!found) {
@@ -328,6 +325,7 @@ static void _plugin_destroy_session(janus_plugin_session* session, int *err) {
 		*err = -2;
 	}
 	atomic_store(&_g_has_watchers, has_watchers);
+	atomic_store(&_g_has_listeners, has_listeners);
 	_UNLOCK_ALL;
 }
 
@@ -396,40 +394,54 @@ static struct janus_plugin_result *_plugin_handle_message(
 			json_decref(m_event); \
 		}
 
-	json_t *const request_obj = json_object_get(msg, "request");
-	if (request_obj == NULL) {
+	json_t *const request = json_object_get(msg, "request");
+	if (request == NULL) {
 		PUSH_ERROR(400, "Request missing");
 		goto ok_wait;
 	}
 
-	const char *const request_str = json_string_value(request_obj);
+	const char *const request_str = json_string_value(request);
 	if (request_str == NULL) {
 		PUSH_ERROR(400, "Request not a string");
 		goto ok_wait;
 	}
 	// US_JLOG_INFO("main", "Message: %s", request_str);
 
-#	define PUSH_STATUS(x_status, x_jsep) { \
+#	define PUSH_STATUS(x_status, x_payload, x_jsep) { \
 			json_t *const m_event = json_object(); \
 			json_object_set_new(m_event, "ustreamer", json_string("event")); \
 			json_t *const m_result = json_object(); \
 			json_object_set_new(m_result, "status", json_string(x_status)); \
+			if (x_payload != NULL) { \
+				json_object_set_new(m_result, x_status, x_payload); \
+			} \
 			json_object_set_new(m_event, "result", m_result); \
 			_g_gw->push_event(session, create(), transaction, m_event, x_jsep); \
 			json_decref(m_event); \
 		}
 
 	if (!strcmp(request_str, "start")) {
-		PUSH_STATUS("started", NULL);
+		PUSH_STATUS("started", NULL, NULL);
 
 	} else if (!strcmp(request_str, "stop")) {
-		PUSH_STATUS("stopped", NULL);
+		PUSH_STATUS("stopped", NULL, NULL);
 
 	} else if (!strcmp(request_str, "watch")) {
-		char *sdp;
+		bool with_audio = false;
 		{
+			json_t *const params = json_object_get(msg, "params");
+			if (params != NULL) {
+				json_t *const audio = json_object_get(params, "audio");
+				if (audio != NULL && json_is_boolean(audio)) {
+					with_audio = (_g_rtpa != NULL && json_boolean_value(audio));
+				}
+			}
+		}
+
+		{
+			char *sdp;
 			char *const video_sdp = us_rtpv_make_sdp(_g_rtpv);
-			char *const audio_sdp = (_g_rtpa ? us_rtpa_make_sdp(_g_rtpa) : us_strdup(""));
+			char *const audio_sdp = (with_audio ? us_rtpa_make_sdp(_g_rtpa) : us_strdup(""));
 			US_ASPRINTF(sdp,
 				"v=0" RN
 				"o=- %" PRIu64 " 1 IN IP4 0.0.0.0" RN
@@ -447,13 +459,31 @@ static struct janus_plugin_result *_plugin_handle_message(
 				audio_sdp, video_sdp
 #				endif
 			);
+			json_t *const offer_jsep = json_pack("{ssss}", "type", "offer", "sdp", sdp);
+			PUSH_STATUS("started", NULL, offer_jsep);
+			json_decref(offer_jsep);
 			free(audio_sdp);
 			free(video_sdp);
+			free(sdp);
 		}
-		json_t *const offer_jsep = json_pack("{ssss}", "type", "offer", "sdp", sdp);
-		free(sdp);
-		PUSH_STATUS("started", offer_jsep);
-		json_decref(offer_jsep);
+
+		{
+			_LOCK_ALL;
+			bool has_listeners = false;
+			US_LIST_ITERATE(_g_clients, client, {
+				if (client->session == session) {
+					atomic_store(&client->transmit_audio, with_audio);
+				}
+				has_listeners = (has_listeners || atomic_load(&client->transmit_audio));
+			});
+			atomic_store(&_g_has_listeners, has_listeners);
+			_UNLOCK_ALL;
+		}
+
+	} else if (!strcmp(request_str, "features")) {
+		json_t *const features = json_pack("{sb}", "audio", (_g_rtpa != NULL));
+		PUSH_STATUS("features", features, NULL);
+		json_decref(features);
 
 	} else if (!strcmp(request_str, "key_required")) {
 		// US_JLOG_INFO("main", "Got key_required message");

janus/src/rtp.c

@@ -26,12 +26,11 @@
 #include "rtp.h"
 
 
-us_rtp_s *us_rtp_init(unsigned payload, bool video, bool zero_playout_delay) {
+us_rtp_s *us_rtp_init(unsigned payload, bool video) {
 	us_rtp_s *rtp;
 	US_CALLOC(rtp, 1);
 	rtp->payload = payload;
 	rtp->video = video;
-	rtp->zero_playout_delay = zero_playout_delay; // See client.c
 	rtp->ssrc = us_triple_u32(us_get_now_monotonic_u64());
 	return rtp;
 }

janus/src/rtp.h

@@ -39,18 +39,18 @@
 typedef struct {
 	unsigned	payload;
 	bool		video;
-	bool		zero_playout_delay;
 	uint32_t	ssrc;
 
 	uint16_t	seq;
 	uint8_t		datagram[US_RTP_DATAGRAM_SIZE];
 	size_t		used;
+	bool		zero_playout_delay;
 } us_rtp_s;
 
 typedef void (*us_rtp_callback_f)(const us_rtp_s *rtp);
 
 
-us_rtp_s *us_rtp_init(unsigned payload, bool video, bool zero_playout_delay);
+us_rtp_s *us_rtp_init(unsigned payload, bool video);
 us_rtp_s *us_rtp_dup(const us_rtp_s *rtp);
 void us_rtp_destroy(us_rtp_s *rtp);
 

janus/src/rtpa.c

@@ -26,7 +26,7 @@
 us_rtpa_s *us_rtpa_init(us_rtp_callback_f callback) {
 	us_rtpa_s *rtpa;
 	US_CALLOC(rtpa, 1);
-	rtpa->rtp = us_rtp_init(111, false, false);
+	rtpa->rtp = us_rtp_init(111, false);
 	rtpa->callback = callback;
 	return rtpa;
 }

janus/src/rtpv.c

@@ -31,10 +31,10 @@ void _rtpv_process_nalu(us_rtpv_s *rtpv, const uint8_t *data, size_t size, uint3
 static ssize_t _find_annexb(const uint8_t *data, size_t size);
 
 
-us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback, bool zero_playout_delay) {
+us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback) {
 	us_rtpv_s *rtpv;
 	US_CALLOC(rtpv, 1);
-	rtpv->rtp = us_rtp_init(96, true, zero_playout_delay);
+	rtpv->rtp = us_rtp_init(96, true);
 	rtpv->callback = callback;
 	return rtpv;
 }
@@ -59,12 +59,11 @@ char *us_rtpv_make_sdp(us_rtpv_s *rtpv) {
 		"a=rtcp-fb:%u nack pli" RN
 		"a=rtcp-fb:%u goog-remb" RN
 		"a=ssrc:%" PRIu32 " cname:ustreamer" RN
-		"%s" // playout-delay
+		"a=extmap:1 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay" RN
 		"a=sendonly" RN,
 		PAYLOAD, PAYLOAD, PAYLOAD, PAYLOAD,
 		PAYLOAD, PAYLOAD, PAYLOAD,
-		rtpv->rtp->ssrc,
-		(rtpv->rtp->zero_playout_delay ? "a=extmap:1 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay" RN : "")
+		rtpv->rtp->ssrc
 	);
 	return sdp;
 #	undef PAYLOAD
@@ -78,6 +77,8 @@ void us_rtpv_wrap(us_rtpv_s *rtpv, const us_frame_s *frame) {
 
 	assert(frame->format == V4L2_PIX_FMT_H264);
 
+	rtpv->rtp->zero_playout_delay = (frame->gop == 0);
+
 	const uint32_t pts = us_get_now_monotonic_u64() * 9 / 100; // PTS units are in 90 kHz
 	ssize_t last_offset = -_PRE;
 

janus/src/rtpv.h

@@ -43,7 +43,7 @@ typedef struct {
 } us_rtpv_s;
 
 
-us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback, bool zero_playout_delay);
+us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback);
 void us_rtpv_destroy(us_rtpv_s *rtpv);
 
 char *us_rtpv_make_sdp(us_rtpv_s *rtpv);

linters/Dockerfile

@@ -1,4 +1,4 @@
-FROM archlinux/archlinux:base-devel
+FROM archlinux/archlinux:base
 
 RUN mkdir -p /etc/pacman.d/hooks \
 	&& ln -s /dev/null /etc/pacman.d/hooks/30-systemd-tmpfiles.hook

linters/tox.ini

@@ -26,7 +26,7 @@ commands = cppcheck \
 whitelist_externals = bash
 commands = bash -c 'flake8 --config=linters/flake8.ini tools/*.py' python/*.py
 deps =
-	flake8
+	flake8==5.0.4
 	flake8-quotes
 
 [testenv:pylint]

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 5.31" "January 2021"
+.TH USTREAMER-DUMP 1 "version 5.36" "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 5.31" "November 2020"
+.TH USTREAMER 1 "version 5.36" "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=5.31
+pkgver=5.36
 pkgrel=1
 pkgdesc="Lightweight and fast MJPEG-HTTP streamer"
 url="https://github.com/pikvm/ustreamer"

pkg/docker/Dockerfile.alpine

@@ -24,7 +24,7 @@ RUN apk add --no-cache \
 WORKDIR /ustreamer
 COPY --from=build /build/ustreamer/src/ustreamer.bin ustreamer
 
-RUN wget https://raw.githubusercontent.com/pikvm/kvmd/master/configs/kvmd/tc358743-edid.hex -O /edid.hex
+RUN wget https://raw.githubusercontent.com/pikvm/kvmd/master/configs/kvmd/edid/v3-hdmi.hex -O /edid.hex
 COPY pkg/docker/entry.sh /
 
 EXPOSE 8080

pkg/openwrt/Makefile

@@ -6,7 +6,7 @@
 include $(TOPDIR)/rules.mk
 
 PKG_NAME:=ustreamer
-PKG_VERSION:=5.31
+PKG_VERSION:=5.36
 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="5.31",
+        version="5.36",
         description="uStreamer tools",
         author="Maxim Devaev",
         author_email="mdevaev@gmail.com",

python/src/ustreamer.c

@@ -236,6 +236,7 @@ static PyObject *_MemsinkObject_wait_frame(_MemsinkObject *self, PyObject *args,
 	SET_NUMBER(stride, Long, Long);
 	SET_NUMBER(online, Long, Bool);
 	SET_NUMBER(key, Long, Bool);
+	SET_NUMBER(gop, Long, Long);
 	SET_NUMBER(grab_ts, Double, Float);
 	SET_NUMBER(encode_begin_ts, Double, Float);
 	SET_NUMBER(encode_end_ts, Double, Float);

src/dump/file.c

@@ -52,11 +52,11 @@ void us_output_file_write(void *v_output, const us_frame_s *frame) {
 		us_base64_encode(frame->data, frame->used, &output->base64_data, &output->base64_allocated);
 		fprintf(output->fp,
 			"{\"size\": %zu, \"width\": %u, \"height\": %u,"
-			" \"format\": %u, \"stride\": %u, \"online\": %u, \"key\": %u,"
+			" \"format\": %u, \"stride\": %u, \"online\": %u, \"key\": %u, \"gop\": %u,"
 			" \"grab_ts\": %.3Lf, \"encode_begin_ts\": %.3Lf, \"encode_end_ts\": %.3Lf,"
 			" \"data\": \"%s\"}\n",
 			frame->used, frame->width, frame->height,
-			frame->format, frame->stride, frame->online, frame->key,
+			frame->format, frame->stride, frame->online, frame->key, frame->gop,
 			frame->grab_ts, frame->encode_begin_ts, frame->encode_end_ts,
 			output->base64_data);
 	} else {

src/dump/main.c

@@ -260,16 +260,16 @@ static int _dump_sink(
 			const long long now_second = us_floor_ms(now);
 
 			char fourcc_str[8];
-			US_LOG_VERBOSE("Frame: res=%ux%u, fmt=%s, stride=%u, online=%d, key=%d, kr=%d, latency=%.3Lf, diff=%.3Lf, size=%zu",
-				frame->width, frame->height,
+			US_LOG_VERBOSE("Frame: %s - %ux%u -- online=%d, key=%d, kr=%d, gop=%u, latency=%.3Lf, backlog=%.3Lf, size=%zu",
 				us_fourcc_to_string(frame->format, fourcc_str, 8),
-				frame->stride, frame->online, frame->key, key_requested,
+				frame->width, frame->height,
+				frame->online, frame->key, key_requested, frame->gop,
 				now - frame->grab_ts, (last_ts ? now - last_ts : 0),
 				frame->used);
 			last_ts = now;
 
-			US_LOG_DEBUG("       grab_ts=%.3Lf, encode_begin_ts=%.3Lf, encode_end_ts=%.3Lf",
-				frame->grab_ts, frame->encode_begin_ts, frame->encode_end_ts);
+			US_LOG_DEBUG("       stride=%u, grab_ts=%.3Lf, encode_begin_ts=%.3Lf, encode_end_ts=%.3Lf",
+				frame->stride, frame->grab_ts, frame->encode_begin_ts, frame->encode_end_ts);
 
 			if (now_second != fps_second) {
 				fps = fps_accum;

src/libs/const.h

@@ -23,7 +23,7 @@
 #pragma once
 
 #define US_VERSION_MAJOR 5
-#define US_VERSION_MINOR 31
+#define US_VERSION_MINOR 36
 
 #define US_MAKE_VERSION2(_major, _minor) #_major "." #_minor
 #define US_MAKE_VERSION1(_major, _minor) US_MAKE_VERSION2(_major, _minor)

src/libs/frame.h

@@ -50,6 +50,7 @@ typedef struct {
 
 	bool		online;
 	bool		key;
+	unsigned	gop;
 
 	long double	grab_ts;
 	long double	encode_begin_ts;
@@ -64,6 +65,7 @@ typedef struct {
 		x_dest->stride = x_src->stride; \
 		x_dest->online = x_src->online; \
 		x_dest->key = x_src->key; \
+		x_dest->gop = x_src->gop; \
 		x_dest->grab_ts = x_src->grab_ts; \
 		x_dest->encode_begin_ts = x_src->encode_begin_ts; \
 		x_dest->encode_end_ts = x_src->encode_end_ts; \
@@ -81,6 +83,7 @@ static inline void us_frame_copy_meta(const us_frame_s *src, us_frame_s *dest) {
 		&& x_a->stride == x_b->stride \
 		&& x_a->online == x_b->online \
 		&& x_a->key == x_b->key \
+		&& x_a->gop == x_b->gop \
 	)
 
 

src/libs/memsinksh.h

@@ -30,7 +30,7 @@
 
 
 #define US_MEMSINK_MAGIC	((uint64_t)0xCAFEBABECAFEBABE)
-#define US_MEMSINK_VERSION	((uint32_t)3)
+#define US_MEMSINK_VERSION	((uint32_t)4)
 
 #ifndef US_CFG_MEMSINK_MAX_DATA
 #	define US_CFG_MEMSINK_MAX_DATA 33554432
@@ -51,6 +51,7 @@ typedef struct {
 	unsigned	stride;
 	bool		online;
 	bool		key;
+	unsigned	gop;
 
 	long double	grab_ts;
 	long double	encode_begin_ts;

src/ustreamer/m2m.c

@@ -476,6 +476,7 @@ static int _m2m_encoder_compress_raw(us_m2m_encoder_s *enc, const us_frame_s *sr
 			} else {
 				us_frame_set_data(dest, _RUN(output_bufs[output_buf.index].data), output_plane.bytesused);
 				dest->key = output_buf.flags & V4L2_BUF_FLAG_KEYFRAME;
+				dest->gop = enc->gop;
 				done = true;
 			}