Merge branch 'master-v4.3' of https://github.com/sle118/squeezelite-esp32 into master-v4.3

CHANGELOG

@@ -1,3 +1,24 @@
+2024-01-27
+ - complete libflac fix and add chaining enablement
+ - fixed stream Ogg demux issue with unknown granule
+ 
+2024-01-19
+ - fixed libflac with OggFlac
+ - AirPlay missed frame logging
+ 
+2024-01-16
+ - catch-up with cspot latest
+ - refactor airplay flush/first packet
+ - new libFLAC that supports multi-stream OggFlac
+ - fix output threshold
+ - log missed frames
+ 
+2024-01-10
+ - add OggFlac to stream metadata
+ - fix OggFlac deadlock in flac callback when not enough data in streambuf
+ - fix no displayer due to threadshold too high (use 500ms instead)
+ - reset outputbuf when cspot starts
+ 
 2024-01-01
  - ogg stream are parsed to foward metadata to LMS
  - fix some ogg parsing on multi-stream containers
@@ -14,7 +35,7 @@
  - force gpio_pad_select_gpio in dac_controlset in case somebody uses UART gpio's (or other pre-programmed)
  
 2023-11-08
- - execute dac_controlset even whne there is no i2s (for gpio)
+ - execute dac_controlset even when there is no i2s (for gpio)
  
 2023-11-07
  - led-vu gain + misc fixes

README.md

@@ -637,7 +637,11 @@ You can install IDF manually on Linux or Windows (using the Subsystem for Linux)
 **Use the esp-idf 4.3.5 https://github.com/espressif/esp-idf/tree/release/v4.3.5 ** or the 4.4.5 (and above version) if you want to build for esp32-s3
 
 ## Building SqueezeESP32
-When initially cloning the repo, make sure you do it recursively. For example: `git clone --recursive https://github.com/sle118/squeezelite-esp32.git`
+When initially cloning the repo, make sure you do it recursively. For example: `git clone --recursive https://github.com/sle118/squeezelite-esp32.git`. You also should install cspot additional components for protobuf use.
+```
+$ sudo pip3 install protobuf grpcio-tools
+```
+NB: I need to check on a fresh installation, but you might also require "protoc". You should do that within the esp32 local Python environment.
 	
 Don't forget to choose one of the config files in build_scripts/ and rename it sdkconfig.defaults or sdkconfig as many important WiFi/BT options are set there. **The codecs libraries will not be rebuilt by these scripts (it's a tedious process - see below)**
 

components/codecs/inc/FLAC/stream_decoder.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2022  Xiph.Org Foundation
+ * Copyright (C) 2011-2023  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -782,6 +782,25 @@ FLAC_API void FLAC__stream_decoder_delete(FLAC__StreamDecoder *decoder);
  */
 FLAC_API FLAC__bool FLAC__stream_decoder_set_ogg_serial_number(FLAC__StreamDecoder *decoder, long serial_number);
 
+/** Set the "allow Ogg chaining" flag. If set, the Ogg decoder will
+ *  prepare to receive a new stream once the last Ogg page arrives for
+ *  the stream encapsulating the FLAC audio data. This can be used to
+ *  support chained Ogg FLAC streams; a new \c STREAMINFO signals the
+ *  beginning of a new stream.
+ *
+ * \note
+ * This function has no effect with native FLAC decoding.
+ *
+ * \default \c false
+ * \param  decoder          A decoder instance to set.
+ * \param  allow            Whether to allow chained streams.
+ * \assert
+ *    \code decoder != NULL \endcode
+ * \retval FLAC__bool
+ *    \c false if the decoder is already initialized, else \c true.
+ */
+FLAC_API FLAC__bool FLAC__stream_decoder_set_ogg_chaining(FLAC__StreamDecoder* decoder, FLAC__bool value);
+
 /** Set the "MD5 signature checking" flag.  If \c true, the decoder will
  *  compute the MD5 signature of the unencoded audio data while decoding
  *  and compare it to the signature from the STREAMINFO block, if it
@@ -906,6 +925,17 @@ FLAC_API FLAC__StreamDecoderState FLAC__stream_decoder_get_state(const FLAC__Str
  */
 FLAC_API const char *FLAC__stream_decoder_get_resolved_state_string(const FLAC__StreamDecoder *decoder);
 
+/** Get the "allow Ogg chaining" flag as described in
+ *  \code FLAC__stream_decoder_set_ogg_chaining \endcode.
+ *
+ * \param  decoder  A decoder instance to query.
+ * \assert
+ *    \code decoder != NULL \endcode
+ * \retval FLAC__bool
+ *    See above.
+ */
+FLAC_API FLAC__bool FLAC__stream_decoder_get_ogg_chaining(const FLAC__StreamDecoder* decoder);
+
 /** Get the "MD5 signature checking" flag.
  *  This is the value of the setting, not whether or not the decoder is
  *  currently checking the MD5 (remember, it can be turned off automatically

components/raop/rtp.c

@@ -96,6 +96,7 @@ typedef struct __attribute__((__packed__)) audio_buffer_entry {   // decoded aud
     u16_t len;    
     u8_t ready;
     u8_t allocated;
+    u8_t missed;
 } abuf_t;
 
 typedef struct rtp_s {
@@ -126,11 +127,6 @@ typedef struct rtp_s {
 		u32_t 	rtp, time;
 		u8_t  	status;
 	} synchro;
-	struct {
-		u32_t time;
-		seq_t seqno;
-		u32_t rtptime;
-	} record;
 	int latency;			// rtp hold depth in samples
 	u32_t resent_req, resent_rec;	// total resent + recovered frames
 	u32_t silent_frames;	// total silence frames
@@ -147,8 +143,8 @@ typedef struct rtp_s {
 #endif
 
 	struct alac_codec_s *alac_codec;
-	int flush_seqno;
-	bool playing;
+	int first_seqno;
+	enum { RTP_WAIT, RTP_STREAM, RTP_PLAY } state;
     int stalled;
 	raop_data_cb_t data_cb;
 	raop_cmd_cb_t cmd_cb;
@@ -231,7 +227,7 @@ rtp_resp_t rtp_init(struct in_addr host, int latency, char *aeskey, char *aesiv,
 	ctx->rtp_host.sin_family = AF_INET;
 	ctx->rtp_host.sin_addr.s_addr = INADDR_ANY;
 	pthread_mutex_init(&ctx->ab_mutex, 0);
-	ctx->flush_seqno = -1;
+	ctx->first_seqno = -1;
 	ctx->latency = latency;
 	ctx->ab_read = ctx->ab_write;
 
@@ -339,24 +335,23 @@ void rtp_end(rtp_t *ctx)
 
 /*---------------------------------------------------------------------------*/
 bool rtp_flush(rtp_t *ctx, unsigned short seqno, unsigned int rtptime, bool exit_locked)
-{
-	bool rc = true;
-	u32_t now = gettime_ms();
+{  
+    pthread_mutex_lock(&ctx->ab_mutex);
+    
+    // always store flush seqno as we only want stricly above it, even when equal to RECORD
+    ctx->first_seqno = seqno;
+    bool flushed = false;
 
-	if (now < ctx->record.time + 250 || (ctx->record.seqno == seqno && ctx->record.rtptime == rtptime)) {
-		rc = false;
-		LOG_ERROR("[%p]: FLUSH ignored as same as RECORD (%hu - %u)", ctx, seqno, rtptime);
-	} else {
-		pthread_mutex_lock(&ctx->ab_mutex);
-		buffer_reset(ctx->audio_buffer);
-		ctx->playing = false;
-		ctx->flush_seqno = seqno;
-		if (!exit_locked) pthread_mutex_unlock(&ctx->ab_mutex);
+    // no need to stop playing if recent or equal to record - but first_seqno is needed
+    if (ctx->state == RTP_PLAY) {
+        buffer_reset(ctx->audio_buffer);
+        ctx->state = RTP_WAIT;
+        flushed = true;
+        LOG_INFO("[%p]: FLUSH packets below %hu - %u", ctx, seqno, rtptime);
 	}
-
-	LOG_INFO("[%p]: flush %hu %u", ctx, seqno, rtptime);
-
-	return rc;
+    
+	if (!exit_locked || !flushed) pthread_mutex_unlock(&ctx->ab_mutex);
+	return flushed;
 }
 
 /*---------------------------------------------------------------------------*/
@@ -367,11 +362,9 @@ void rtp_flush_release(rtp_t *ctx) {
 
 /*---------------------------------------------------------------------------*/
 void rtp_record(rtp_t *ctx, unsigned short seqno, unsigned rtptime) {
-	ctx->record.seqno = seqno;
-	ctx->record.rtptime = rtptime;
-	ctx->record.time = gettime_ms();
-
-	LOG_INFO("[%p]: record %hu %u", ctx, seqno, rtptime);
+    ctx->first_seqno = (seqno || rtptime) ? seqno : -1;
+	ctx->state = RTP_WAIT;
+	LOG_INFO("[%p]: record %hu - %u", ctx, seqno, rtptime);	
 }
 
 /*---------------------------------------------------------------------------*/
@@ -442,29 +435,54 @@ static void alac_decode(rtp_t *ctx, s16_t *dest, char *buf, int len, u16_t *outs
 /*---------------------------------------------------------------------------*/
 static void buffer_put_packet(rtp_t *ctx, seq_t seqno, unsigned rtptime, bool first, char *data, int len) {
 	abuf_t *abuf = NULL;
-	u32_t playtime;
 
 	pthread_mutex_lock(&ctx->ab_mutex);
-
-	if (!ctx->playing) {
-		if ((ctx->flush_seqno == -1 || seq_order(ctx->flush_seqno, seqno)) &&
-		   (ctx->synchro.status & RTP_SYNC) && (ctx->synchro.status & NTP_SYNC)) {
-			ctx->ab_write = seqno-1;
-			ctx->ab_read = seqno;
-			ctx->flush_seqno = -1;
-			ctx->playing = true;
-			ctx->resent_req = ctx->resent_rec = ctx->silent_frames = ctx->discarded = 0;
-			playtime = ctx->synchro.time + ((rtptime - ctx->synchro.rtp) * 10) / (RAOP_SAMPLE_RATE / 100);
-			ctx->cmd_cb(RAOP_PLAY, playtime);
-		} else {
-			pthread_mutex_unlock(&ctx->ab_mutex);
-			return;
-		}
-	}
     
+    /* if we have received a RECORD with a seqno, then this is the first allowed rtp sequence number 
+	 * and we are in RTP_WAIT state. If seqno was 0, then we are waiting for a flush that will tell 
+	 * us what should be our first allowed packet but we must accept everything, wait and clean when 
+	 * we the it arrives. This means that first packet moves us to RTP_STREAM state where we accept
+	 * frames but wait for the FLUSH. If this was a FLUSH while playing, then we are also in RTP_WAIT 
+	 * state but we do have an allowed seqno and we should not accept any frame before we have it */
+
+	// if we have a pending first seqno and we are below, always ignore it
+	if (ctx->first_seqno != -1 && seq_order(seqno, ctx->first_seqno)) {
+		pthread_mutex_unlock(&ctx->ab_mutex);
+		return;
+	}
+
+	if (ctx->state == RTP_WAIT) {
+		ctx->ab_write = seqno - 1;
+		ctx->ab_read = ctx->ab_write + 1;
+        ctx->resent_req = ctx->resent_rec = ctx->silent_frames = ctx->discarded = 0;        
+		if (ctx->first_seqno != -1) {
+        	LOG_INFO("[%p]: 1st accepted packet:%d, now playing", ctx, seqno);                                    
+			ctx->state = RTP_PLAY;
+			ctx->first_seqno = -1;
+            u32_t playtime = ctx->synchro.time + ((rtptime - ctx->synchro.rtp) * 10) / (RAOP_SAMPLE_RATE / 100);            
+            ctx->cmd_cb(RAOP_PLAY, playtime);         
+		} else {
+            ctx->state = RTP_STREAM;
+			LOG_INFO("[%p]: 1st accepted packet:%hu, waiting for FLUSH", ctx, seqno);
+		}
+	} else if (ctx->state == RTP_STREAM && ctx->first_seqno != -1 && seq_order(ctx->first_seqno, seqno + 1)) {
+		// now we're talking, but first discard all packets with a seqno below first_seqno AND not ready
+		while (seq_order(ctx->ab_read, ctx->first_seqno) ||
+			!ctx->audio_buffer[BUFIDX(ctx->ab_read)].ready) {
+			ctx->audio_buffer[BUFIDX(ctx->ab_read)].ready = false;
+			ctx->ab_read++;
+		}
+        LOG_INFO("[%p]: done waiting for FLUSH with packet:%d, now playing starting:%hu", ctx, seqno, ctx->ab_read);
+		ctx->state = RTP_PLAY;
+		ctx->first_seqno = -1;
+        u32_t playtime = ctx->synchro.time + ((rtptime - ctx->synchro.rtp) * 10) / (RAOP_SAMPLE_RATE / 100);            
+		ctx->cmd_cb(RAOP_PLAY, playtime);         
+	}   
+    
+    abuf = ctx->audio_buffer + BUFIDX(seqno);
+
 	if (seqno == (u16_t) (ctx->ab_write+1)) {
 		// expected packet
-		abuf = ctx->audio_buffer + BUFIDX(seqno);
 		ctx->ab_write = seqno;
 		LOG_SDEBUG("packet expected seqno:%hu rtptime:%u (W:%hu R:%hu)", seqno, rtptime, ctx->ab_write, ctx->ab_read);
 	} else if (seq_order(ctx->ab_write, seqno)) {
@@ -475,7 +493,7 @@ static void buffer_put_packet(rtp_t *ctx, seq_t seqno, unsigned rtptime, bool fi
             ctx->ab_read = seqno;            
 		} else {
             // request re-send missed frames and evaluate resent date as a whole *after*
-            rtp_request_resend(ctx, ctx->ab_write + 1, seqno-1);
+            if (ctx->state == RTP_PLAY) rtp_request_resend(ctx, ctx->ab_write + 1, seqno-1);
             
             // resend date is after all requests have been sent
             u32_t now = gettime_ms();
@@ -488,16 +506,15 @@ static void buffer_put_packet(rtp_t *ctx, seq_t seqno, unsigned rtptime, bool fi
             LOG_DEBUG("[%p]: packet newer seqno:%hu rtptime:%u (W:%hu R:%hu)", ctx, seqno, rtptime, ctx->ab_write, ctx->ab_read);            
         }        
 
-		abuf = ctx->audio_buffer + BUFIDX(seqno);
 		ctx->ab_write = seqno;
 	} else if (seq_order(ctx->ab_read, seqno + 1)) {
 		// recovered packet, not yet sent
-		abuf = ctx->audio_buffer + BUFIDX(seqno);
 		ctx->resent_rec++;
 		LOG_DEBUG("[%p]: packet recovered seqno:%hu rtptime:%u (W:%hu R:%hu)", ctx, seqno, rtptime, ctx->ab_write, ctx->ab_read);
 	} else {
-		// too late
-		LOG_DEBUG("[%p]: packet too late seqno:%hu rtptime:%u (W:%hu R:%hu)", ctx, seqno, rtptime, ctx->ab_write, ctx->ab_read);
+        // too late
+		if (abuf->missed) LOG_INFO("[%p]: packet too late seqno:%hu rtptime:%u (W:%hu R:%hu)", ctx, seqno, rtptime, ctx->ab_write, ctx->ab_read);
+        abuf = NULL;
 	}
 
 	if (ctx->in_frames++ > 1000) {
@@ -508,6 +525,7 @@ static void buffer_put_packet(rtp_t *ctx, seq_t seqno, unsigned rtptime, bool fi
 	if (abuf) {
 		alac_decode(ctx, abuf->data, data, len, &abuf->len);
 		abuf->ready = 1;
+        abuf->missed = 0;
 		// this is the local rtptime when this frame is expected to play
 		abuf->rtptime = rtptime;
 		buffer_push_packet(ctx);
@@ -528,7 +546,7 @@ static void buffer_push_packet(rtp_t *ctx) {
 	u32_t now, playtime, hold = max((ctx->latency * 1000) / (8 * RAOP_SAMPLE_RATE), 100);
 
 	// not ready to play yet
-	if (!ctx->playing || ctx->synchro.status != (RTP_SYNC | NTP_SYNC)) return;
+	if (ctx->state != RTP_PLAY || ctx->synchro.status != (RTP_SYNC | NTP_SYNC)) return;
 
 	// there is always at least one frame in the buffer
 	do {
@@ -551,6 +569,7 @@ static void buffer_push_packet(rtp_t *ctx) {
 				LOG_DEBUG("[%p]: created zero frame (W:%hu R:%hu)", ctx, ctx->ab_write, ctx->ab_read);
 				ctx->data_cb(silence_frame, ctx->frame_size * 4, playtime);
 				ctx->silent_frames++;
+                curframe->missed = 1;
 			}
 		} else if (curframe->ready) {
 			ctx->data_cb((const u8_t*) curframe->data, curframe->len, playtime);

components/spotify/cspot/bell/external/civetweb/civetweb.c

@@ -22482,9 +22482,13 @@ mg_init_library(unsigned features)
 			file_mutex_init =
 			    pthread_mutex_init(&global_log_file_lock, &pthread_mutex_attr);
 			if (file_mutex_init == 0) {
+#ifdef WINSOCK_START
 				/* Start WinSock */
 				WSADATA data;
 				failed = wsa = WSAStartup(MAKEWORD(2, 2), &data);
+#else
+				failed = wsa = 0;
+#endif
 			}
 #else
 			mutexattr_init = pthread_mutexattr_init(&pthread_mutex_attr);
@@ -22498,7 +22502,9 @@ mg_init_library(unsigned features)
 		if (failed) {
 #if defined(_WIN32)
 			if (wsa == 0) {
+#ifdef WINSOCK_START
 				(void)WSACleanup();
+#endif
 			}
 			if (file_mutex_init == 0) {
 				(void)pthread_mutex_destroy(&global_log_file_lock);
@@ -22598,7 +22604,9 @@ mg_exit_library(void)
 #endif
 
 #if defined(_WIN32)
+#ifdef WINSOCK_START
 		(void)WSACleanup();
+#endif
 		(void)pthread_mutex_destroy(&global_log_file_lock);
 #else
 		(void)pthread_mutexattr_destroy(&pthread_mutex_attr);

components/spotify/cspot/bell/main/io/BellHTTPServer.cpp

@@ -12,6 +12,8 @@
 
 using namespace bell;
 
+std::mutex BellHTTPServer::initMutex;
+
 class WebSocketHandler : public CivetWebSocketHandler {
  public:
   BellHTTPServer::WSDataHandler dataHandler;
@@ -187,6 +189,7 @@ bool BellHTTPServer::handlePost(CivetServer* server,
 }
 
 BellHTTPServer::BellHTTPServer(int serverPort) {
+  std::lock_guard lock(initMutex);
   mg_init_library(0);
   BELL_LOG(info, "HttpServer", "Server listening on port %d", serverPort);
   this->serverPort = serverPort;
@@ -197,6 +200,11 @@ BellHTTPServer::BellHTTPServer(int serverPort) {
   server = std::make_unique<CivetServer>(civetWebOptions);
 }
 
+BellHTTPServer::~BellHTTPServer() {
+  std::lock_guard lock(initMutex);
+  mg_exit_library();
+}
+
 std::unique_ptr<BellHTTPServer::HTTPResponse> BellHTTPServer::makeJsonResponse(
     const std::string& json, int status) {
   auto response = std::make_unique<BellHTTPServer::HTTPResponse>();

components/spotify/cspot/bell/main/io/include/BellHTTPServer.h

@@ -19,6 +19,7 @@ namespace bell {
 class BellHTTPServer : public CivetHandler {
  public:
   BellHTTPServer(int serverPort);
+  ~BellHTTPServer();
 
   enum class WSState { CONNECTED, READY, CLOSED };
 
@@ -100,6 +101,8 @@ class BellHTTPServer : public CivetHandler {
   std::mutex responseMutex;
   HTTPHandler notFoundHandler;
 
+  static std::mutex initMutex;
+
   bool handleGet(CivetServer* server, struct mg_connection* conn);
   bool handlePost(CivetServer* server, struct mg_connection* conn);
 };

components/spotify/cspot/bell/main/platform/linux/MDNSService.cpp

@@ -6,6 +6,7 @@
 #include <cstring>
 #include <vector>
 #include <mutex>
+#include <atomic>
 
 #if __has_include("avahi-client/client.h")
 #include <avahi-client/client.h>
@@ -41,8 +42,9 @@ class implMDNSService : public MDNSService {
 #endif
   static struct mdnsd* mdnsServer;
   static in_addr_t host;
+  static std::atomic<size_t> instances;
 
-  implMDNSService(struct mdns_service* service) : service(service){};
+  implMDNSService(struct mdns_service* service) : service(service){ instances++; };
 #ifndef BELL_DISABLE_AVAHI
   implMDNSService(AvahiEntryGroup* avahiGroup) : avahiGroup(avahiGroup){};
 #endif
@@ -51,6 +53,7 @@ class implMDNSService : public MDNSService {
 
 struct mdnsd* implMDNSService::mdnsServer = NULL;
 in_addr_t implMDNSService::host = INADDR_ANY;
+std::atomic<size_t> implMDNSService::instances = 0;
 static std::mutex registerMutex;
 #ifndef BELL_DISABLE_AVAHI
 AvahiClient* implMDNSService::avahiClient = NULL;
@@ -66,11 +69,21 @@ void implMDNSService::unregisterService() {
 #ifndef BELL_DISABLE_AVAHI
   if (avahiGroup) {
     avahi_entry_group_free(avahiGroup);
+    if (!--instances && implMDNSService::avahiClient) {
+      avahi_client_free(implMDNSService::avahiClient);
+      avahi_simple_poll_free(implMDNSService::avahiPoll);
+      implMDNSService::avahiClient = nullptr;
+      implMDNSService::avahiPoll = nullptr;
+    }
   } else
 #endif
   {
     mdns_service_remove(implMDNSService::mdnsServer, service);
-  }
+    if (!--instances && implMDNSService::mdnsServer) {
+     mdnsd_stop(implMDNSService::mdnsServer);
+     implMDNSService::mdnsServer = nullptr;
+    }
+  }  
 }
 
 std::unique_ptr<MDNSService> MDNSService::registerService(
@@ -180,19 +193,14 @@ std::unique_ptr<MDNSService> MDNSService::registerService(
     std::string type(serviceType + "." + serviceProto + ".local");
 
     BELL_LOG(info, "MDNS", "using built-in mDNS for %s", serviceName.c_str());
-    struct mdns_service* mdnsService =
+    auto service =
         mdnsd_register_svc(implMDNSService::mdnsServer, serviceName.c_str(),
-                           type.c_str(), servicePort, NULL, txt.data());
-    if (mdnsService) {
-      auto service =
-          mdnsd_register_svc(implMDNSService::mdnsServer, serviceName.c_str(),
                              type.c_str(), servicePort, NULL, txt.data());
 
-      return std::make_unique<implMDNSService>(service);
-    }
+    if (service) return std::make_unique<implMDNSService>(service);
   }
 
   BELL_LOG(error, "MDNS", "cannot start any mDNS listener for %s",
            serviceName.c_str());
-  return NULL;
+  return nullptr;
 }

components/spotify/cspot/bell/main/platform/win32/MDNSService.cpp

@@ -19,13 +19,12 @@ using namespace bell;
 class implMDNSService : public MDNSService {
  private:
   struct mdns_service* service;
-  void unregisterService(void) {
-    mdns_service_remove(implMDNSService::mdnsServer, service);
-  };
+  void unregisterService(void);
 
  public:
   static struct mdnsd* mdnsServer;
-  implMDNSService(struct mdns_service* service) : service(service){};
+  static std::atomic<size_t> instances;
+  implMDNSService(struct mdns_service* service) : service(service) { instances++; };
 };
 
 /**
@@ -33,8 +32,18 @@ class implMDNSService : public MDNSService {
  **/
 
 struct mdnsd* implMDNSService::mdnsServer = NULL;
+std::atomic<size_t> implMDNSService::instances = 0;
+
 static std::mutex registerMutex;
 
+void implMDNSService::unregisterService() {
+  mdns_service_remove(implMDNSService::mdnsServer, service);
+  if (!--instances && implMDNSService::mdnsServer) {
+    mdnsd_stop(implMDNSService::mdnsServer);
+    implMDNSService::mdnsServer = nullptr;
+  }   
+}
+
 std::unique_ptr<MDNSService> MDNSService::registerService(
     const std::string& serviceName, const std::string& serviceType,
     const std::string& serviceProto, const std::string& serviceHost,
@@ -94,5 +103,5 @@ std::unique_ptr<MDNSService> MDNSService::registerService(
       mdnsd_register_svc(implMDNSService::mdnsServer, serviceName.c_str(),
                          type.c_str(), servicePort, NULL, txt.data());
 
-  return std::make_unique<implMDNSService>(service);
+  return service ? std::make_unique<implMDNSService>(service) : nullptr;
 }

components/spotify/cspot/bell/main/utilities/include/BellTask.h

@@ -72,7 +72,11 @@ class Task {
                           (LPTHREAD_START_ROUTINE)taskEntryFunc, this, 0, NULL);
     return thread != NULL;
 #else
-    return (pthread_create(&thread, NULL, taskEntryFunc, this) == 0);
+    if (!pthread_create(&thread, NULL, taskEntryFunc, this)) {
+        pthread_detach(thread);
+        return true;
+    }
+    return false;
 #endif
   }
 
@@ -108,13 +112,7 @@ class Task {
 #endif
 
   static void* taskEntryFunc(void* This) {
-    Task* self = (Task*)This;
-    self->runTask();
-#if _WIN32
-    WaitForSingleObject(self->thread, INFINITE);
-#else
-    pthread_join(self->thread, NULL);
-#endif
+    ((Task*)This)->runTask();
     return NULL;
   }
 };

components/spotify/cspot/include/TrackQueue.h

@@ -24,7 +24,7 @@ class CDNAudioFile;
 // Used in got track info event
 struct TrackInfo {
   std::string name, album, artist, imageUrl, trackId;
-  uint32_t duration;
+  uint32_t duration, number, discNumber;
 
   void loadPbTrack(Track* pbTrack, const std::vector<uint8_t>& gid);
   void loadPbEpisode(Episode* pbEpisode, const std::vector<uint8_t>& gid);

components/spotify/cspot/protobuf/metadata.proto

@@ -32,6 +32,8 @@ message Artist {
 message Track {
     optional bytes gid = 1;
     optional string name = 2;
+    optional sint32 number = 5;
+    optional sint32 disc_number = 6;
     optional sint32 duration = 0x7;
     optional Album album = 0x3;
     repeated Artist artist = 0x4;
@@ -44,6 +46,7 @@ message Episode {
     optional bytes gid = 1;
     optional string name = 2;
     optional sint32 duration = 7;
+    optional sint32 number = 65;
     repeated AudioFile file = 12;
     repeated Restriction restriction = 0x4B;
     optional ImageGroup covers = 0x44;

components/spotify/cspot/src/TrackQueue.cpp

@@ -97,6 +97,8 @@ void TrackInfo::loadPbTrack(Track* pbTrack, const std::vector<uint8_t>& gid) {
     }
   }
 
+  number = pbTrack->has_number ? pbTrack->number : 0;
+  discNumber = pbTrack->has_disc_number ? pbTrack->disc_number : 0;
   duration = pbTrack->duration;
 }
 
@@ -113,6 +115,8 @@ void TrackInfo::loadPbEpisode(Episode* pbEpisode,
     imageUrl = "https://i.scdn.co/image/" + bytesToHexString(imageId);
   }
 
+  number = pbEpisode->has_number ? pbEpisode->number : 0;
+  discNumber = 0;
   duration = pbEpisode->duration;
 }
 

components/squeezelite/decode_external.c

@@ -288,6 +288,7 @@ static bool raop_sink_cmd_handler(raop_event_t event, va_list args)
 			output.frames_played = 0;
 			output.external = DECODE_RAOP;
 			output.state = OUTPUT_STOPPED;
+
 			if (decode.state != DECODE_STOPPED) decode.state = DECODE_ERROR;
 			LOG_INFO("resizing buffer %u", outputbuf->size);
 			break;
@@ -377,6 +378,7 @@ static bool cspot_cmd_handler(cspot_event_t cmd, va_list args)
 		output.state = OUTPUT_STOPPED;
         sink_state = SINK_ABORT;
 		_buf_flush(outputbuf);
+        _buf_limit(outputbuf, 0);
 		if (decode.state != DECODE_STOPPED) decode.state = DECODE_ERROR;
 		LOG_INFO("CSpot start track");
 		break;

components/squeezelite/flac.c

@@ -70,6 +70,7 @@ struct flac {
 	);
 	FLAC__bool (* FLAC__stream_decoder_process_single)(FLAC__StreamDecoder *decoder);
 	FLAC__StreamDecoderState (* FLAC__stream_decoder_get_state)(const FLAC__StreamDecoder *decoder);
+    FLAC__bool (*FLAC__stream_decoder_set_ogg_chaining)(FLAC__StreamDecoder* decoder, FLAC__bool allow);
 #endif
 };
 
@@ -121,6 +122,9 @@ static FLAC__StreamDecoderReadStatus read_cb(const FLAC__StreamDecoder *decoder,
 	_buf_inc_readp(streambuf, bytes);
 	UNLOCK_S;
 
+    // give some time for stream to acquire data otherwise flac will hammer us
+    if (!end && !bytes) usleep(100 * 1000);
+
 	*want = bytes;
 
 	return end ? FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM : FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
@@ -137,8 +141,8 @@ static FLAC__StreamDecoderWriteStatus write_cb(const FLAC__StreamDecoder *decode
 	FLAC__int32 *rptr = (FLAC__int32 *)buffer[channels > 1 ? 1 : 0];
 	
 	if (decode.new_stream) {
+		LOG_INFO("setting track_start");        
 		LOCK_O;
-		LOG_INFO("setting track_start");
 		output.track_start = outputbuf->writep;
 		decode.new_stream = false;
 
@@ -253,6 +257,7 @@ static void flac_open(u8_t sample_size, u8_t sample_rate, u8_t channels, u8_t en
 	
 	if ( f->container == 'o' ) {
 		LOG_INFO("ogg/flac container - using init_ogg_stream");
+        FLAC(f, stream_decoder_set_ogg_chaining, f->decoder, true);
 		FLAC(f, stream_decoder_init_ogg_stream, f->decoder, &read_cb, NULL, NULL, NULL, NULL, &write_cb, NULL, &error_cb, NULL);
 	} else {
 		FLAC(f, stream_decoder_init_stream, f->decoder, &read_cb, NULL, NULL, NULL, NULL, &write_cb, NULL, &error_cb, NULL);
@@ -295,6 +300,7 @@ static bool load_flac() {
 	f->FLAC__stream_decoder_init_ogg_stream = dlsym(handle, "FLAC__stream_decoder_init_ogg_stream");
 	f->FLAC__stream_decoder_process_single = dlsym(handle, "FLAC__stream_decoder_process_single");
 	f->FLAC__stream_decoder_get_state = dlsym(handle, "FLAC__stream_decoder_get_state");
+    f->FLAC__stream_decoder_set_ogg_chaining = dlsym(handle, "FLAC__stream_decoder_set_ogg_chaining");
 
 	if ((err = dlerror()) != NULL) {
 		LOG_INFO("dlerror: %s", err);		

components/squeezelite/opus.c

@@ -194,7 +194,7 @@ static int read_opus_header(void) {
 		// nothing has been found and we have no more bytes, come back later
 		if (status <= 0) break;
 
-		// always set stream serialno if we have a new one
+		// always set stream serialno if we have a new one (no multiplexed streams)
 		if (OG(&go, page_bos, &u->page)) OG(&go, stream_reset_serialno, &u->state, OG(&go, page_serialno, &u->page));
 
 		// bring new page in if we want it (otherwise we're just skipping)

components/squeezelite/output.c

@@ -60,7 +60,7 @@ frames_t _output_frames(frames_t avail) {
 	silence = false;
 
 	// start when threshold met
-	if (output.state == OUTPUT_BUFFER && (frames * BYTES_PER_FRAME) > output.threshold * output.next_sample_rate / 10 && frames > output.start_frames) {
+	if (output.state == OUTPUT_BUFFER && frames > output.threshold * output.next_sample_rate / 10 && frames > output.start_frames) {
 		output.state = OUTPUT_RUNNING;
 		LOG_INFO("start buffer frames: %u", frames);
 		wake_controller();

components/squeezelite/output_bt.c

@@ -139,8 +139,11 @@ static int _write_frames(frames_t out_frames, bool silence, s32_t gainL, s32_t g
 		u8_t *buf = silencebuf;
 		memcpy(btout + oframes * BYTES_PER_FRAME, buf, out_frames * BYTES_PER_FRAME);
 	}
-	
-	output_visu_export(btout + oframes * BYTES_PER_FRAME, out_frames, output.current_sample_rate, silence, (gainL  + gainR) / 2);
+
+    // don't update visu if we don't have enough data in buffer (500 ms)
+    if (silence || _buf_used(outputbuf) >  BYTES_PER_FRAME * output.current_sample_rate / 2) {
+    	output_visu_export(btout + oframes * BYTES_PER_FRAME, out_frames, output.current_sample_rate, silence, (gainL  + gainR) / 2);
+    }
 	
 	oframes += out_frames;
 

components/squeezelite/output_i2s.c

@@ -439,7 +439,7 @@ void output_init_i2s(log_level level, char *device, unsigned output_buf_size, ch
 		static DRAM_ATTR StaticTask_t xTaskBuffer __attribute__ ((aligned (4)));
 		static EXT_RAM_ATTR StackType_t xStack[OUTPUT_THREAD_STACK_SIZE] __attribute__ ((aligned (4)));
 		output_i2s_task = xTaskCreateStaticPinnedToCore( (TaskFunction_t) output_thread_i2s, "output_i2s", OUTPUT_THREAD_STACK_SIZE, 
-											  NULL, CONFIG_ESP32_PTHREAD_TASK_PRIO_DEFAULT + 1, xStack, &xTaskBuffer, 0 );
+											  NULL, CONFIG_ESP32_PTHREAD_TASK_PRIO_DEFAULT + 10, xStack, &xTaskBuffer, 0 );
 	}
 }
 
@@ -485,8 +485,8 @@ static int _i2s_write_frames(frames_t out_frames, bool silence, s32_t gainL, s32
 		memcpy(obuf + oframes * BYTES_PER_FRAME, silencebuf, out_frames * BYTES_PER_FRAME);
 	}
 
-	// don't update visu if we don't have enough data in buffer
-	if (silence || output.external || _buf_used(outputbuf) > outputbuf->size >> 2 ) {
+	// don't update visu if we don't have enough data in buffer (500 ms)
+	if (silence || _buf_used(outputbuf) >  BYTES_PER_FRAME * output.current_sample_rate / 2) {
 		output_visu_export(obuf + oframes * BYTES_PER_FRAME, out_frames, output.current_sample_rate, silence, (gainL + gainR) / 2);
 	}
 		

components/squeezelite/slimproto.c

@@ -393,7 +393,9 @@ static void process_strm(u8_t *pkt, int len) {
 				stream_file(header, header_len, strm->threshold * 1024);
 				autostart -= 2;
 			} else {
-				stream_sock(ip, port, strm->format, header, header_len, strm->threshold * 1024, autostart >= 2);
+				stream_sock(ip, port, strm->flags & 0x20, 
+                            strm->format == 'o' || strm->format == 'u' || (strm->format == 'f' && strm->pcm_sample_size == 'o'),
+                            header, header_len, strm->threshold * 1024, autostart >= 2);
 			}
 			sendSTAT("STMc", 0);
 			sentSTMu = sentSTMo = sentSTMl = false;
@@ -412,8 +414,8 @@ static void process_strm(u8_t *pkt, int len) {
 			output.fade_secs = strm->transition_period;
 			output.invert = (strm->flags & 0x03) == 0x03;
 			output.channels = (strm->flags & 0x0c) >> 2;
+			UNLOCK_O;            
 			LOG_DEBUG("set fade: %u, channels: %u, invert: %u", output.fade_mode, output.channels, output.invert);
-			UNLOCK_O;
 		}
 		break;
 	default:

components/squeezelite/squeezelite.h

@@ -580,26 +580,12 @@ struct streamstate {
 	u32_t meta_next;
 	u32_t meta_left;
 	bool  meta_send;
-    struct {
-		enum { STREAM_OGG_OFF, STREAM_OGG_SYNC, STREAM_OGG_HEADER, STREAM_OGG_SEGMENTS, STREAM_OGG_PAGE } state;
-		u32_t want, miss, match;
-		u8_t* data, segments[255];
-#pragma pack(push, 1)
-		struct {
-			char pattern[4];
-			u8_t version, type;
-			u64_t granule;
-			u32_t serial, page, checksum;
-			u8_t count;
-		} header;
-    } ogg;
-#pragma pack(pop)
 };
 
 void stream_init(log_level level, unsigned stream_buf_size);
 void stream_close(void);
 void stream_file(const char *header, size_t header_len, unsigned threshold);
-void stream_sock(u32_t ip, u16_t port, char codec, const char *header, size_t header_len, unsigned threshold, bool cont_wait);
+void stream_sock(u32_t ip, u16_t port, bool use_ssl, bool use_ogg, const char *header, size_t header_len, unsigned threshold, bool cont_wait);
 bool stream_disconnect(void);
 
 // decode.c

components/squeezelite/stream.c

@@ -61,6 +61,23 @@ static sockfd fd;
 
 struct EXT_RAM_ATTR streamstate stream;
 
+static EXT_RAM_ATTR struct {
+    bool flac;
+    u64_t serial;
+	enum { OGG_OFF, OGG_SYNC, OGG_HEADER, OGG_SEGMENTS, OGG_PAGE } state;
+	size_t want, miss, match;
+	u8_t* data, segments[255];
+#pragma pack(push, 1)    
+	struct {
+		char pattern[4];
+		u8_t version, type;
+		u64_t granule;
+		u32_t serial, page, checksum;
+		u8_t count;
+	} header;
+#pragma pack(pop)    
+} ogg;
+
 #if USE_SSL
 static SSL_CTX *SSLctx;
 SSL *ssl;
@@ -113,7 +130,6 @@ static int _poll(SSL *ssl, struct pollfd *pollinfo, int timeout) {
 }
 #endif
 
-
 static bool send_header(void) {
 	char *ptr = stream.header;
 	int len = stream.header_len;
@@ -148,8 +164,8 @@ static bool running = true;
 static void _disconnect(stream_state state, disconnect_code disconnect) {
 	stream.state = state;
 	stream.disconnect = disconnect;
-    if (stream.ogg.state == STREAM_OGG_PAGE && stream.ogg.data) free(stream.ogg.data);
-    stream.ogg.data = NULL;
+    if (ogg.state == OGG_PAGE && ogg.data) free(ogg.data);
+    ogg.data = NULL;
 #if USE_SSL
 	if (ssl) {
 		SSL_shutdown(ssl);
@@ -162,81 +178,96 @@ static void _disconnect(stream_state state, disconnect_code disconnect) {
 	wake_controller();
 }
 
-static u32_t memfind(const u8_t* haystack, u32_t n, const char* needle, u32_t len, u32_t *offset) {
-	int i;
+static size_t memfind(const u8_t* haystack, size_t n, const char* needle, size_t len, size_t* offset) {
+	size_t i;
 	for (i = 0; i < n && *offset != len; i++) *offset = (haystack[i] == needle[*offset]) ? *offset + 1 : 0;
 	return i;
 }
 
+/* https://xiph.org/ogg/doc/framing.html 
+ * https://xiph.org/flac/ogg_mapping.html
+ * https://xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-610004.2 */
+ 
 static void stream_ogg(size_t n) {
-	if (stream.ogg.state == STREAM_OGG_OFF) return;
+	if (ogg.state == OGG_OFF) return;
 	u8_t* p = streambuf->writep;
 
 	while (n) {
-		size_t consumed = min(stream.ogg.miss, n);
+		size_t consumed = min(ogg.miss, n);
 
 		// copy as many bytes as possible and come back later if we do'nt have enough
-		if (stream.ogg.data) {
-			memcpy(stream.ogg.data + stream.ogg.want - stream.ogg.miss, p, consumed);
-			stream.ogg.miss -= consumed;
-			if (stream.ogg.miss) return;
+		if (ogg.data) {
+			memcpy(ogg.data + ogg.want - ogg.miss, p, consumed);
+			ogg.miss -= consumed;
+			if (ogg.miss) return;
 		}
 
 		// we have what we want, let's parse
-		switch (stream.ogg.state) {
-		case STREAM_OGG_SYNC: {
-			stream.ogg.miss -= consumed;
+		switch (ogg.state) {
+		case OGG_SYNC: {
+			ogg.miss -= consumed;
 			if (consumed) break;
 
 			// we have to memorize position in case any of last 3 bytes match...
-			int pos = memfind(p, n, "OggS", 4, &stream.ogg.match);
-			if (stream.ogg.match == 4) {
-				consumed = pos - stream.ogg.match;
-				stream.ogg.state = STREAM_OGG_HEADER;
-				stream.ogg.miss = stream.ogg.want = sizeof(stream.ogg.header);
-				stream.ogg.data = (u8_t*) &stream.ogg.header;
-				stream.ogg.match = 0;
+			size_t pos = memfind(p, n, "OggS", 4, &ogg.match);
+			if (ogg.match == 4) {
+				consumed = pos - ogg.match;
+				ogg.state = OGG_HEADER;
+				ogg.miss = ogg.want = sizeof(ogg.header);
+				ogg.data = (u8_t*) &ogg.header;
+				ogg.match = 0;
 			} else {
-				LOG_INFO("OggS not at expected position");
+                if (!ogg.match) LOG_INFO("OggS not at expected position %zu/%zu", pos, n);
+                LOG_INFO("OggS not at expected position %zu/%zu", pos, n);
 				return;
 			}
 			break;
 		}
-		case STREAM_OGG_HEADER:
-			if (!memcmp(stream.ogg.header.pattern, "OggS", 4)) {
-				stream.ogg.miss = stream.ogg.want = stream.ogg.header.count;
-				stream.ogg.data = stream.ogg.segments;
-				stream.ogg.state = STREAM_OGG_SEGMENTS;
+		case OGG_HEADER:
+			if (!memcmp(ogg.header.pattern, "OggS", 4)) {
+				ogg.miss = ogg.want = ogg.header.count;
+				ogg.data = ogg.segments;
+				ogg.state = OGG_SEGMENTS;
 			} else {
-				stream.ogg.state = STREAM_OGG_SYNC;
-				stream.ogg.data = NULL;
+				ogg.state = OGG_SYNC;
+				ogg.data = NULL;
 			}
 			break;
-		case STREAM_OGG_SEGMENTS:
+		case OGG_SEGMENTS:
 			// calculate size of page using lacing values
-			for (int i = 0; i < stream.ogg.want; i++) stream.ogg.miss += stream.ogg.data[i];
-			stream.ogg.want = stream.ogg.miss;
+			for (size_t i = 0; i < ogg.want; i++) ogg.miss += ogg.data[i];
+			ogg.want = ogg.miss;
+            
+            // acquire serial number when we are looking for headers and hit a bos
+			if (ogg.serial == ULLONG_MAX && (ogg.header.type & 0x02)) ogg.serial = ogg.header.serial;
 
-			if (stream.ogg.header.granule == 0) {
-				// granule 0 means a new stream, so let's look into it
-				stream.ogg.state = STREAM_OGG_PAGE;
-				stream.ogg.data = malloc(stream.ogg.want);
-			} else {
+			// we have overshot and missed header, reset serial number to restart search (O and -1 are le/be)
+			if (ogg.header.serial == ogg.serial && ogg.header.granule && ogg.header.granule != -1) ogg.serial = ULLONG_MAX;
+
+			// not our serial (the above protected us from granule > 0)
+			if (ogg.header.serial != ogg.serial) {
 				// otherwise, jump over data
-				stream.ogg.state = STREAM_OGG_SYNC;
-				stream.ogg.data = NULL;
+				ogg.state = OGG_SYNC;
+				ogg.data = NULL;
+			} else {
+				ogg.state = OGG_PAGE;
+				ogg.data = malloc(ogg.want);
 			}
 			break;
-		case STREAM_OGG_PAGE: {
-			u32_t offset = 0;
+		case OGG_PAGE: {
+			char** tag = (char* []){ "\x3vorbis", "OpusTags", NULL };
+			size_t ofs = 0;
 
-			// try to find one of valid Ogg pattern (vorbis, opus)
-			for (char** tag = (char*[]) { "\x3vorbis", "OpusTags", NULL }; *tag; tag++, offset = 0) {
-				u32_t pos = memfind(stream.ogg.data, stream.ogg.want, *tag, strlen(*tag), &offset);
-				if (offset != strlen(*tag)) continue;
-				
+			/* with OggFlac, we need the next page (packet) - VorbisComment is wrapped into a FLAC_METADATA
+			 * and except with vorbis, comment packet starts a new page but even in vorbis, it won't span
+			 * accross multiple pages */
+			if (ogg.flac) ofs = 4;
+			else if (!memcmp(ogg.data, "\x7f""FLAC", 5)) ogg.flac = true;
+			else for (size_t n = 0; *tag; tag++, ofs = 0) if ((ofs = memfind(ogg.data, ogg.want, *tag, strlen(*tag), &n)) && n == strlen(*tag)) break;
+	
+			if (ofs) {			
 				// u32:len,char[]:vendorId, u32:N, N x (u32:len,char[]:comment)
-				char* p = (char*) stream.ogg.data + pos;
+				char* p = (char*) ogg.data + ofs;
 				p += *p + 4;
 				u32_t count = *p;
 				p += 4;
@@ -256,16 +287,19 @@ static void stream_ogg(size_t n) {
 						stream.header[stream.header_len++] = len;
 						memcpy(stream.header + stream.header_len, p, len);
 						stream.header_len += len;
+                        LOG_INFO("metadata: %.*s", len, p);
 					}
 				}
 
+                ogg.flac = false;
+                ogg.serial = ULLONG_MAX;
 				stream.meta_send = true;
 				wake_controller();
 				LOG_INFO("Ogg metadata length: %u", stream.header_len - 3);
-				break;
 			}
-			free(stream.ogg.data);
-			stream.ogg.state = STREAM_OGG_SYNC;
+			free(ogg.data);
+            ogg.data = NULL;
+			ogg.state = OGG_SYNC;
 			break;
 		}
         default: 
@@ -603,7 +637,7 @@ void stream_file(const char *header, size_t header_len, unsigned threshold) {
 	UNLOCK;
 }
 
-void stream_sock(u32_t ip, u16_t port, char codec, const char *header, size_t header_len, unsigned threshold, bool cont_wait) {
+void stream_sock(u32_t ip, u16_t port, bool use_ssl, bool use_ogg, const char *header, size_t header_len, unsigned threshold, bool cont_wait) {
 	struct sockaddr_in addr;
 
 #if EMBEDDED
@@ -703,8 +737,10 @@ void stream_sock(u32_t ip, u16_t port, char codec, const char *header, size_t he
 	stream.bytes = 0;
 	stream.threshold = threshold;
     
-    stream.ogg.miss = stream.ogg.match = 0;
-	stream.ogg.state = (codec == 'o' || codec == 'p') ? STREAM_OGG_SYNC : STREAM_OGG_OFF;
+    ogg.miss = ogg.match = 0;
+	ogg.state = use_ogg ? OGG_SYNC : OGG_OFF;
+    ogg.flac = false;
+    ogg.serial = ULLONG_MAX;
 
 	UNLOCK;
 }
@@ -725,8 +761,8 @@ bool stream_disconnect(void) {
 		disc = true;
 	}
 	stream.state = STOPPED;
-    if (stream.ogg.state == STREAM_OGG_PAGE && stream.ogg.data) free(stream.ogg.data);
-    stream.ogg.data = NULL;
+    if (ogg.state == OGG_PAGE && ogg.data) free(ogg.data);
+    ogg.data = NULL;
 	UNLOCK;
 	return disc;
 }

components/squeezelite/vorbis.c

@@ -201,7 +201,7 @@ static int read_vorbis_header(void) {
 		// nothing has been found and we have no more bytes, come back later
 		if (status <= 0) break;
 
-		// always set stream serialno if we have a new one
+		// always set stream serialno if we have a new one (no multiplexed streams)
 		if (OG(&go, page_bos, &v->page)) OG(&go, stream_reset_serialno, &v->state, OG(&go, page_serialno, &v->page));
 
 		// bring new page in if we want it (otherwise we're just skipping)

components/wifi-manager/webapp/dist/src/js/test.d.ts

@@ -73,6 +73,12 @@ declare function getStatus(): {};
 declare function getStatus(): {};
 declare function getStatus(): {};
 declare function getStatus(): {};
+declare function getStatus(): {};
+declare function getStatus(): {};
+declare function getStatus(): {};
+declare function getRadioButton(entry: any): string;
+declare function getRadioButton(entry: any): string;
+declare function getRadioButton(entry: any): string;
 declare function getRadioButton(entry: any): string;
 declare function getRadioButton(entry: any): string;
 declare function getRadioButton(entry: any): string;
@@ -223,6 +229,9 @@ declare function pushStatus(): void;
 declare function pushStatus(): void;
 declare function pushStatus(): void;
 declare function pushStatus(): void;
+declare function pushStatus(): void;
+declare function pushStatus(): void;
+declare function pushStatus(): void;
 declare let sd: {};
 declare let rf: boolean;
 declare function refreshStatus(): void;

components/wifi-manager/webapp/webapp.cmake

@@ -1,5 +1,5 @@
 target_add_binary_data( __idf_wifi-manager webapp/dist/css/index.4bbe29a78a667faa2b6f.css.gz BINARY)
 target_add_binary_data( __idf_wifi-manager webapp/dist/favicon-32x32.png BINARY)
 target_add_binary_data( __idf_wifi-manager webapp/dist/index.html.gz BINARY)
-target_add_binary_data( __idf_wifi-manager webapp/dist/js/index.105fd5.bundle.js.gz BINARY)
-target_add_binary_data( __idf_wifi-manager webapp/dist/js/node_vendors.105fd5.bundle.js.gz BINARY)
+target_add_binary_data( __idf_wifi-manager webapp/dist/js/index.4ae048.bundle.js.gz BINARY)
+target_add_binary_data( __idf_wifi-manager webapp/dist/js/node_vendors.4ae048.bundle.js.gz BINARY)

components/wifi-manager/webapp/webpack.c

@@ -6,29 +6,29 @@ extern const uint8_t _favicon_32x32_png_start[] asm("_binary_favicon_32x32_png_s
 extern const uint8_t _favicon_32x32_png_end[] asm("_binary_favicon_32x32_png_end");
 extern const uint8_t _index_html_gz_start[] asm("_binary_index_html_gz_start");
 extern const uint8_t _index_html_gz_end[] asm("_binary_index_html_gz_end");
-extern const uint8_t _index_105fd5_bundle_js_gz_start[] asm("_binary_index_105fd5_bundle_js_gz_start");
-extern const uint8_t _index_105fd5_bundle_js_gz_end[] asm("_binary_index_105fd5_bundle_js_gz_end");
-extern const uint8_t _node_vendors_105fd5_bundle_js_gz_start[] asm("_binary_node_vendors_105fd5_bundle_js_gz_start");
-extern const uint8_t _node_vendors_105fd5_bundle_js_gz_end[] asm("_binary_node_vendors_105fd5_bundle_js_gz_end");
+extern const uint8_t _index_4ae048_bundle_js_gz_start[] asm("_binary_index_4ae048_bundle_js_gz_start");
+extern const uint8_t _index_4ae048_bundle_js_gz_end[] asm("_binary_index_4ae048_bundle_js_gz_end");
+extern const uint8_t _node_vendors_4ae048_bundle_js_gz_start[] asm("_binary_node_vendors_4ae048_bundle_js_gz_start");
+extern const uint8_t _node_vendors_4ae048_bundle_js_gz_end[] asm("_binary_node_vendors_4ae048_bundle_js_gz_end");
 const char * resource_lookups[] = {
 	"/css/index.4bbe29a78a667faa2b6f.css.gz",
 	"/favicon-32x32.png",
 	"/index.html.gz",
-	"/js/index.105fd5.bundle.js.gz",
-	"/js/node_vendors.105fd5.bundle.js.gz",
+	"/js/index.4ae048.bundle.js.gz",
+	"/js/node_vendors.4ae048.bundle.js.gz",
 ""
 };
 const uint8_t * resource_map_start[] = {
 	_index_4bbe29a78a667faa2b6f_css_gz_start,
 	_favicon_32x32_png_start,
 	_index_html_gz_start,
-	_index_105fd5_bundle_js_gz_start,
-	_node_vendors_105fd5_bundle_js_gz_start
+	_index_4ae048_bundle_js_gz_start,
+	_node_vendors_4ae048_bundle_js_gz_start
 };
 const uint8_t * resource_map_end[] = {
 	_index_4bbe29a78a667faa2b6f_css_gz_end,
 	_favicon_32x32_png_end,
 	_index_html_gz_end,
-	_index_105fd5_bundle_js_gz_end,
-	_node_vendors_105fd5_bundle_js_gz_end
+	_index_4ae048_bundle_js_gz_end,
+	_node_vendors_4ae048_bundle_js_gz_end
 };

components/wifi-manager/webapp/webpack.h

@@ -1,6 +1,6 @@
 /***********************************
 webpack_headers
-dist/css/index.4bbe29a78a667faa2b6f.css.gz,dist/favicon-32x32.png,dist/index.html.gz,dist/js/index.105fd5.bundle.js.gz,dist/js/node_vendors.105fd5.bundle.js.gz
+dist/css/index.4bbe29a78a667faa2b6f.css.gz,dist/favicon-32x32.png,dist/index.html.gz,dist/js/index.4ae048.bundle.js.gz,dist/js/node_vendors.4ae048.bundle.js.gz
 ***********************************/
 #pragma once
 #include <inttypes.h>