Merge branch 'master' into libconfig

Author: lminiero

conf/janus.plugin.videoroom.jcfg.sample

@@ -51,6 +51,7 @@ room-1234: {
 	fir_freq = 10
 	#audiocodec = "opus"
 	#videocodec = "vp8"
+	#transport_wide_cc_ext = true
 	record = false
 	#rec_dir = "/path/to/recordings-folder"
 }
@@ -68,4 +69,5 @@ room-5678: {
 	fir_freq = 10
 	videocodec = "vp9"
 	video_svc = true
+	#transport_wide_cc_ext = true
 }

html/videoroomtest.js

@@ -109,7 +109,7 @@ $(document).ready(function() {
 									Janus.debug("Consent dialog should be " + (on ? "on" : "off") + " now");
 									if(on) {
 										// Darken screen and show hint
-										$.blockUI({ 
+										$.blockUI({
 											message: '<div><img src="up_arrow.png"/></div>',
 											css: {
 												border: 'none',
@@ -457,20 +457,20 @@ function newRemoteFeed(id, display, audio, video) {
 				Janus.log("Plugin attached! (" + remoteFeed.getPlugin() + ", id=" + remoteFeed.getId() + ")");
 				Janus.log("  -- This is a subscriber");
 				// We wait for the plugin to send us an offer
-				var listen = { "request": "join", "room": myroom, "ptype": "subscriber", "feed": id, "private_id": mypvtid };
+				var subscribe = { "request": "join", "room": myroom, "ptype": "subscriber", "feed": id, "private_id": mypvtid };
 				// In case you don't want to receive audio, video or data, even if the
 				// publisher is sending them, set the 'offer_audio', 'offer_video' or
 				// 'offer_data' properties to false (they're true by default), e.g.:
-				// 		listen["offer_video"] = false;
+				// 		subscribe["offer_video"] = false;
 				// For example, if the publisher is VP8 and this is Safari, let's avoid video
 				if(video !== "h264" && Janus.webRTCAdapter.browserDetails.browser === "safari") {
 					if(video)
 						video = video.toUpperCase()
 					toastr.warning("Publisher is using " + video + ", but Safari doesn't support it: disabling video");
-					listen["offer_video"] = false;
+					subscribe["offer_video"] = false;
 				}
 				remoteFeed.videoCodec = video;
-				remoteFeed.send({"message": listen});
+				remoteFeed.send({"message": subscribe});
 			},
 			error: function(error) {
 				Janus.error("  -- Error attaching plugin...", error);
@@ -624,7 +624,7 @@ function newRemoteFeed(id, display, audio, video) {
 				$('#novideo'+remoteFeed.rfindex).remove();
 				$('#curbitrate'+remoteFeed.rfindex).remove();
 				$('#curres'+remoteFeed.rfindex).remove();
-				if(bitrateTimer[remoteFeed.rfindex] !== null && bitrateTimer[remoteFeed.rfindex] !== null) 
+				if(bitrateTimer[remoteFeed.rfindex] !== null && bitrateTimer[remoteFeed.rfindex] !== null)
 					clearInterval(bitrateTimer[remoteFeed.rfindex]);
 				bitrateTimer[remoteFeed.rfindex] = null;
 				remoteFeed.simulcastStarted = false;

ice.c

@@ -2286,10 +2286,11 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 						janus_mutex_unlock(&stream->mutex);
 					}
 				}
-				if(video && stream->rtx_nacked[vindex] != NULL) {
+				if(video) {
 					/* Check if this packet is a duplicate: can happen with RFC4588 */
 					guint16 seqno = ntohs(header->seq_number);
-					int nstate = GPOINTER_TO_INT(g_hash_table_lookup(stream->rtx_nacked[vindex], GUINT_TO_POINTER(seqno)));
+					int nstate = stream->rtx_nacked[vindex] ?
+						GPOINTER_TO_INT(g_hash_table_lookup(stream->rtx_nacked[vindex], GUINT_TO_POINTER(seqno))) : 0;
 					if(nstate == 1) {
 						/* Packet was NACKed and this is the first time we receive it: change state to received */
 						JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Received NACKed packet %"SCNu16" (SSRC %"SCNu32", vindex %d)...\n",
@@ -2300,6 +2301,16 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 						JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Detected duplicate packet %"SCNu16" (SSRC %"SCNu32", vindex %d)...\n",
 							handle->handle_id, seqno, packet_ssrc, vindex);
 						return;
+					} else if(rtx && nstate == 0) {
+						/* We received a retransmission for a packet we didn't NACK: drop it
+						 * FIXME This seems to happen with Chrome when RFC4588 is enabled: in that case,
+						 * Chrome sends the first packet ~8 times as a retransmission, probably to ensure
+						 * we receive it, since the first packet cannot be NACKed (NACKs are triggered
+						 * when there's a gap in between two packets, and the first doesn't have a reference)
+						 * Rather than dropping, we should add a better check in the future */
+						JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Got a retransmission for non-NACKed packet %"SCNu16" (SSRC %"SCNu32", vindex %d)...\n",
+							handle->handle_id, seqno, packet_ssrc, vindex);
+						return;
 					}
 				}
 				/* Backup the RTP header before passing it to the proper RTP switching context */
@@ -2399,7 +2410,8 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 
 				/* Update the RTCP context as well */
 				rtcp_context *rtcp_ctx = video ? stream->video_rtcp_ctx[vindex] : stream->audio_rtcp_ctx;
-				janus_rtcp_process_incoming_rtp(rtcp_ctx, buf, buflen);
+				gboolean count_lost = ((!video && !component->do_audio_nacks) || (video && !component->do_video_nacks)) ? TRUE : FALSE;
+				janus_rtcp_process_incoming_rtp(rtcp_ctx, buf, buflen, count_lost);
 
 				/* Keep track of RTP sequence numbers, in case we need to NACK them */
 				/* 	Note: unsigned int overflow/underflow wraps (defined behavior) */
@@ -2466,6 +2478,7 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 						} else if(cur_seq->state == SEQ_MISSING && now - cur_seq->ts > SEQ_MISSING_WAIT) {
 							JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Missed sequence number %"SCNu16" (%s stream #%d), sending 1st NACK\n",
 								handle->handle_id, cur_seq->seq, video ? "video" : "audio", vindex);
+							rtcp_ctx->lost++;
 							nacks = g_slist_prepend(nacks, GUINT_TO_POINTER(cur_seq->seq));
 							cur_seq->state = SEQ_NACKED;
 							if(video && janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX)) {
@@ -3949,7 +3962,8 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 					janus_rtp_header *header = (janus_rtp_header *)pkt->data;
 					guint32 timestamp = ntohl(header->timestamp);
 					guint16 seq = ntohs(header->seq_number);
-					JANUS_LOG(LOG_ERR, "[%"SCNu64"] ... SRTP protect error... %s (len=%d-->%d, ts=%"SCNu32", seq=%"SCNu16")...\n", handle->handle_id, janus_srtp_error_str(res), pkt->length, protected, timestamp, seq);
+					JANUS_LOG(LOG_DBG, "[%"SCNu64"] ... SRTP protect error... %s (len=%d-->%d, ts=%"SCNu32", seq=%"SCNu16")...\n",
+						handle->handle_id, janus_srtp_error_str(res), pkt->length, protected, timestamp, seq);
 					janus_ice_free_rtp_packet(p);
 				} else {
 					/* Shoot! */

ice.h

@@ -354,7 +354,7 @@ struct janus_ice_stream {
 	/*! \brief Codecs used by this stream */
 	char *audio_codec, *video_codec;
 	/*! \brief Pointer to function to check if a packet is a keyframe (depends on negotiated codec) */
-	gboolean (* video_is_keyframe)(char* buffer, int len);
+	gboolean (* video_is_keyframe)(const char* buffer, int len);
 	/*! \brief Media direction */
 	gboolean audio_send, audio_recv, video_send, video_recv;
 	/*! \brief RTCP context for the audio stream */
@@ -378,7 +378,7 @@ struct janus_ice_stream {
 	/*! \brief  Wether we do transport wide cc for video */
 	gboolean do_transport_wide_cc;
 	/*! \brief Transport wide cc rtp ext ID */
-	guint transport_wide_cc_ext_id;
+	gint transport_wide_cc_ext_id;
 	/*! \brief Last received transport wide seq num */
 	guint32 transport_wide_cc_last_seq_num;
 	/*! \brief Last transport wide seq num sent on feedback */

janus.c

@@ -1227,7 +1227,7 @@ int janus_process_incoming_request(janus_request *request) {
 				} else {
 					/* Check if transport wide CC is supported */
 					int transport_wide_cc_ext_id = janus_rtp_header_extension_get_id(jsep_sdp, JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC);
-					handle->stream->do_transport_wide_cc = TRUE;
+					handle->stream->do_transport_wide_cc = transport_wide_cc_ext_id > 0 ? TRUE : FALSE;
 					handle->stream->transport_wide_cc_ext_id = transport_wide_cc_ext_id;
 				}
 			} else {
@@ -2333,6 +2333,11 @@ json_t *janus_admin_stream_summary(janus_ice_stream *stream) {
 			json_object_set_new(sc, "video-codec", json_string(stream->video_codec));
 		json_object_set_new(s, "codecs", sc);
 	}
+	json_t *bwe = json_object();
+	json_object_set_new(bwe, "twcc", stream->do_transport_wide_cc ? json_true() : json_false());
+	if(stream->transport_wide_cc_ext_id >= 0)
+		json_object_set_new(bwe, "twcc-ext-id", json_integer(stream->transport_wide_cc_ext_id));
+	json_object_set_new(s, "bwe", bwe);
 	json_t *components = json_array();
 	if(stream->component) {
 		json_t *c = janus_admin_component_summary(stream->component);
@@ -2922,12 +2927,6 @@ json_t *janus_plugin_handle_sdp(janus_plugin_session *plugin_session, janus_plug
 		janus_mutex_unlock(&ice_handle->mutex);
 		return NULL;
 	}
-	if (!offer) {
-		/* Check if transport wide CC is supported */
-		int transport_wide_cc_ext_id = janus_rtp_header_extension_get_id(sdp, JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC);
-		stream->do_transport_wide_cc = TRUE;
-		stream->transport_wide_cc_ext_id = transport_wide_cc_ext_id;
-	}
 	if(janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX) &&
 			stream->rtx_payload_types == NULL) {
 		/* Make sure we have a list of rtx payload types to generate, if needed */

plugins/janus_audiobridge.c

@@ -1506,13 +1506,14 @@ void janus_audiobridge_destroy(void) {
 	/* FIXME We should destroy the sessions cleanly */
 	janus_mutex_lock(&sessions_mutex);
 	g_hash_table_destroy(sessions);
+	sessions = NULL;
 	janus_mutex_unlock(&sessions_mutex);
 	janus_mutex_lock(&rooms_mutex);
 	g_hash_table_destroy(rooms);
+	rooms = NULL;
 	janus_mutex_unlock(&rooms_mutex);
 	g_async_queue_unref(messages);
 	messages = NULL;
-	sessions = NULL;
 
 	janus_config_destroy(config);
 	g_free(admin_key);