Merged dev2 into master.

.gitignore

@@ -0,0 +1 @@
+*.pyc

CONTRIBUTORS

@@ -0,0 +1,5 @@
+This is a list of the great people who contributed code to the OpenWebRX repository. (Names are sorted alphabetically.)
+
+Gnoxter <gnoxter@linuxlounge.net>
+John Seamons, ZL/KF6VO <jks@jks.com>
+

README.md

@@ -14,14 +14,28 @@ It has the following features:
 - it works in Google Chrome, Chromium (above version 37) and Mozilla Firefox (above version 28),
 - currently supports RTL-SDR and HackRF; other SDR hardware may be easily added.
 
-**News:**
+**News (2015-08-18)**
 - My BSc. thesis written on OpenWebRX is <a href="http://openwebrx.org/bsc-thesis.pdf">available here.</a>
 - Several bugs were fixed to improve reliability and stability.
 - OpenWebRX now supports compression of audio and waterfall stream, so the required network uplink bandwidth has been decreased from 2 Mbit/s to about 200 kbit/s per client! (Measured with the default settings. It is also dependent on `fft_size`.)
 - OpenWebRX now uses <a href="https://github.com/simonyiszk/csdr#sdrjs">sdr.js</a> (*libcsdr* compiled to JavaScript) for some client-side DSP tasks. 
-- Receivers can now be listed on <a href="http://sdr.hu/">sdr.hu</a>.
+- Receivers can now be listed on <a href="http://sdr.hu/">SDR.hu</a>.
 - License for OpenWebRX is now Affero GPL v3. 
 
+**News (2015-02-14)**
+- The DDC in *csdr* has been [manually optimized](https://github.com/simonyiszk/csdr/blob/2b54054a9f5de9a908ee075b488a5ee74f41ba18/libcsdr.c#L300) for ARM NEON, so it runs around 3 times faster on the Raspberry Pi 2 than before. 
+- Also we use *ncat* instead of *rtl_mus*, and it is 3 times faster.
+- OpenWebRX now supports URLs like: `http://localhost:8073/#freq=145555000,mod=usb`
+- UI improvements were made, thanks to John Seamons and Gnoxter.
+
+> When upgrading OpenWebRX, please make sure that you upgrade *csdr*, and install the new (optional) dependency *ncat*!
+
+## OpenWebRX servers on SDR.hu
+
+[SDR.hu](http://sdr.hu) is a site which lists the active, public OpenWebRX servers. Your receiver [can also be part of it](http://sdr.hu/openwebrx), if you want.
+
+![sdr.hu](/screenshot-sdrhu.png?raw=true)
+
 ## Setup
 
 OpenWebRX currently requires Linux and python 2.7 to run. 
@@ -30,6 +44,11 @@ First you will need to install the dependencies:
 
 - <a href="https://github.com/simonyiszk/csdr">libcsdr</a>
 - <a href="http://sdr.osmocom.org/trac/wiki/rtl-sdr">rtl-sdr</a>
+- ncat (On Debian/Ubuntu, it is in the *nmap* package). 
+
+> By the way, *nmap* is a tool commonly used for auditing network security, and it is not used by OpenWebRX in any way. We need to install it, because the *ncat* command is packaged with it.
+>
+> *ncat* is a better *netcat* alternative, which is used by OpenWebRX for internally distributing the I/Q data stream. It also solves the problem of having different versions of *netcat* on different Linux distributions, which are not compatible by their command-line arguments.
 
 After cloning this repository and connecting an RTL-SDR dongle to your computer, you can run the server:
 
@@ -39,7 +58,6 @@ You can now open the GUI at <a href="http://localhost:8073">http://localhost:807
 
 Please note that the server is also listening on the following ports (on localhost only):
 
-- port 8888 for the I/Q source,
 - port 4951 for the multi-user I/Q server.
 
 Now the next step is to customize the parameters of your server in `config_webrx.py`.
@@ -61,7 +79,7 @@ If you have any problems installing OpenWebRX, you should check out the <a href=
 
 Sometimes the actual error message is not at the end of the terminal output, you may have to look at the whole output to find it.
 
-If you want to run OpenWebRX on a remote server instead of localhost, do not forget to set *server_hostname* in `config_webrx.py`.
+If you want to run OpenWebRX on a remote server instead of *localhost*, do not forget to set *server_hostname* in `config_webrx.py`.
 
 ## Licensing
 

config_rtl.py

@@ -1,103 +0,0 @@
-'''
-	This file is part of RTL Multi-User Server, 
-	that makes multi-user access to your DVB-T dongle used as an SDR.
-	Copyright (c) 2013-2015 by Andras Retzler <randras@sdr.hu>
-
-    This program is free software: you can redistribute it and/or modify
-    it under the terms of the GNU Affero General Public License as
-    published by the Free Software Foundation, either version 3 of the
-    License, or (at your option) any later version.
-
-    This program is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU Affero General Public License for more details.
-
-    You should have received a copy of the GNU Affero General Public License
-    along with this program.  If not, see <http://www.gnu.org/licenses/>.
-
-	++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
-
-	In addition, as a special exception, the copyright holders
-	state that config_rtl.py and config_webrx.py are not part of the 
-	Corresponding Source defined in GNU AGPL version 3 section 1. 
-	
-	(It means that you do not have to redistribute config_rtl.py and 
-	config_webrx.py if you make any changes to these two configuration files,
-	and use them for running your own web service with OpenWebRX.)
-'''
-
-my_ip='127.0.0.1' # leave blank for listening on all interfaces
-my_listening_port = 4951 
-
-rtl_tcp_host,rtl_tcp_port='localhost',8888
-
-send_first=""
-#send_first=chr(9)+chr(0)+chr(0)+chr(0)+chr(1) # set direct sampling
-
-setuid_on_start = 0					# we normally start with root privileges and setuid() to another user
-uid = 999 							# determine by issuing: $ id -u username
-ignore_clients_without_commands = 1 # we won't serve data to telnet sessions and things like that
-									# we'll start to serve data after getting the first valid command 
-
-freq_allowed_ranges = [[0,2200000000]]
-
-client_cant_set_until=0		
-first_client_can_set=True	# openwebrx - spectrum thread will set things on start # no good, clients set parameters and things
-buffer_size=25000000		# per client
-log_file_path = "/dev/null" # Might be set to /dev/null to turn off logging
-
-'''
-Allow any host to connect:
-	use_ip_access_control=0
-
-Allow from specific ranges:
-	use_ip_access_control=1
-	order_allow_deny=0 # deny and then allow
-	denied_ip_ranges=() # deny from all
-	allowed_ip_ranges=('192.168.','44.','127.0.0.1') # allow only from ...
-
-Deny from specific ranges:
-	use_ip_access_control=1
-	order_allow_deny=0 # allow and then deny
-	allowed_ip_ranges=() # allow from all
-	denied_ip_ranges=('192.168.') # deny any hosts from ...
-'''
-use_ip_access_control=1 #You may want to open up the I/Q server to the public, then set this to zero.
-order_allow_deny=0
-denied_ip_ranges=() # deny from all
-allowed_ip_ranges=('127.0.0.1') # allow only local connections (from openwebrx). 
-allow_gain_set=1
-
-use_dsp_command=False # you can process raw I/Q data with a custom command that starts a process that we can pipe the data into, and also pipe out of.
-debug_dsp_command=False # show sample rate before and after the dsp command
-dsp_command=""
-
-'''
-Example DSP commands:
-  * Compress I/Q data with FLAC:
-    flac --force-raw-format --channels 2 --sample-rate=250000 --sign=unsigned --bps=8 --endian=little -o - -
-  * Decompress FLAC-coded I/Q data:
-    flac --force-raw-format --decode --endian=little --sign=unsigned - -
-'''
-watchdog_interval=0
-reconnect_interval=10 
-'''
-If there's no input I/Q data after N seconds, input will be filled with zero samples, 
-so that GNU Radio won't fail in OpenWebRX. It may reconnect rtl_tcp_thread. 
-If watchdog_interval is 0, then watchdog thread is not started. 
-
-'''
-cache_full_behaviour=2
-'''
-	0 = drop samples
-	1 = close client
-	2 = openwebrx: don't care about that client until it wants samples again (gr-osmosdr bug workaround)
-'''
-
-rtl_tcp_password=None 
-'''
-This one applies to a special version of rtl_tcp that has authentication.
-# You can find more info here: https://github.com/ha7ilm/rtl-sdr
-# If it is set to a string (e.g. rtl_tcp_password="changeme"), rtl_mus will try to authenticate against the rtl_tcp server.
-'''

config_webrx.py

@@ -81,7 +81,7 @@ start_rtl_thread=True
 
 # >> RTL-SDR via rtl_sdr 
 
-start_rtl_command="rtl_sdr -s {samp_rate} -f {center_freq} -p {ppm} - | nc -vvl 127.0.0.1 8888".format(rf_gain=rf_gain, center_freq=center_freq, samp_rate=samp_rate, ppm=ppm)
+start_rtl_command="rtl_sdr -s {samp_rate} -f {center_freq} -p {ppm} -g {rf_gain} -".format(rf_gain=rf_gain, center_freq=center_freq, samp_rate=samp_rate, ppm=ppm)
 format_conversion="csdr convert_u8_f"
 
 #start_rtl_command="hackrf_transfer -s {samp_rate} -f {center_freq} -g {rf_gain} -l16 -a0 -r hackrf_pipe & cat hackrf_pipe | nc -vvl 127.0.0.1 8888".format(rf_gain=rf_gain, center_freq=center_freq, samp_rate=samp_rate, ppm=ppm)
@@ -94,23 +94,31 @@ format_conversion="csdr convert_u8_f"
 # >> Sound card SDR (needs ALSA)
 #I did not have the chance to properly test it.
 #samp_rate = 96000
-#start_rtl_command="arecord -f S16_LE -r {samp_rate} -c2 - | nc -vvl 127.0.0.1 8888".format(samp_rate=samp_rate)
+#start_rtl_command="arecord -f S16_LE -r {samp_rate} -c2 -".format(samp_rate=samp_rate)
-#format_conversion="csdr convert_i16_f | csdr gain_ff 30"
+#format_conversion="csdr convert_s16_f | csdr gain_ff 30"
-
-# >> RTL_SDR via rtl_tcp
-#start_rtl_command="rtl_tcp -s {samp_rate} -f {center_freq} -g {rf_gain} -P {ppm} -p 8888".format(rf_gain=rf_gain, center_freq=center_freq, samp_rate=samp_rate, ppm=ppm)
-#format_conversion="csdr convert_u8_f"
 
 # >> /dev/urandom test signal source
 #samp_rate = 2400000
-#start_rtl_command="cat /dev/urandom | (pv -qL `python -c 'print int({samp_rate} * 2.2)'` 2>&1) | nc -vvl 127.0.0.1  8888".format(rf_gain=rf_gain, center_freq=center_freq, samp_rate=samp_rate)
+#start_rtl_command="cat /dev/urandom | (pv -qL `python -c 'print int({samp_rate} * 2.2)'` 2>&1)".format(rf_gain=rf_gain, center_freq=center_freq, samp_rate=samp_rate)
 #format_conversion="csdr convert_u8_f"
 
 #You can use other SDR hardware as well, by giving your own command that outputs the I/Q samples...
 
 shown_center_freq = center_freq #you can change this if you use an upconverter
 
-client_audio_buffer_size = 4 
+client_audio_buffer_size = 5
 #increasing client_audio_buffer_size will:
 # - also increase the latency 
 # - decrease the chance of audio underruns
+
+start_freq = center_freq
+start_mod = "nfm" #nfm, am, lsb, usb, cw
+
+iq_server_port = 4951 #TCP port for ncat to listen on. It will send I/Q data over its connections, for internal use in OpenWebRX. It is only accessible from the localhost by default.
+
+#access_log = "~/openwebrx_access.log"
+
+#Warning! The settings below are very experimental.
+csdr_dynamic_bufsize = False # This allows you to change the buffering mode of csdr.
+csdr_print_bufsizes = False  # This prints the buffer sizes used for csdr processes.
+csdr_through = False # Setting this True will print out how much data is going into the DSP chains.

htdocs/inactive.html

@@ -0,0 +1,85 @@
+<html>
+<!--
+
+	This file is part of OpenWebRX, 
+	an open-source SDR receiver software with a web UI.
+	Copyright (c) 2013-2015 by Andras Retzler <randras@sdr.hu>
+
+    This program is free software: you can redistribute it and/or modify
+    it under the terms of the GNU Affero General Public License as
+    published by the Free Software Foundation, either version 3 of the
+    License, or (at your option) any later version.
+
+    This program is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+    GNU Affero General Public License for more details.
+
+    You should have received a copy of the GNU Affero General Public License
+    along with this program.  If not, see <http://www.gnu.org/licenses/>.
+
+-->
+<head><title>OpenWebRX</title><meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
+<style>
+html, body
+{
+	font-family: "DejaVu Sans", Verdana, Geneva, sans-serif;
+	width: 100%;
+	text-align: center;
+	margin: 0;
+	padding: 0;
+}
+img.logo
+{ 
+	margin-top: 120px;
+}
+div.frame
+{
+	text-align: left;
+	margin:0px auto;
+	width: 800px;
+}
+
+div.panel
+{
+	text-align: center;
+	background-color:#777777; 
+	border-radius: 15px;
+	padding: 12px;
+	font-weight: bold;
+	color: White;
+	font-size: 13pt;
+	/*text-shadow: 1px 1px 4px #444;*/
+	font-family: sans;
+}
+
+div.alt
+{
+	font-size: 10pt;
+	padding-top: 10px;
+}
+
+
+body div a
+{
+	color: #5ca8ff;
+	text-shadow: none;
+}
+
+span.browser
+{
+}
+
+</style>
+</head>
+<body>
+
+<div class="frame">
+	<img class="logo" src="gfx/openwebrx-logo-big.png" style="height: 60px;"/>
+	<div class="panel">
+		Sorry, the receiver is inactive due to internal error.
+	</div>
+</div>
+</body>
+</html>
+

htdocs/index.wrx

@@ -23,11 +23,13 @@
 	<head>
 		<title>OpenWebRX | Open Source SDR Web App for Everyone!</title>
 		<script type="text/javascript">
-			//Local variables
+			//Global variables
-			client_id="%[CLIENT_ID]";
+			var client_id="%[CLIENT_ID]";
-			ws_url="%[WS_URL]";
+			var ws_url="%[WS_URL]";
-			rx_photo_height=%[RX_PHOTO_HEIGHT];
+			var rx_photo_height=%[RX_PHOTO_HEIGHT];
 			var audio_buffering_fill_to=%[AUDIO_BUFSIZE];
+			var starting_mod = "%[START_MOD]";
+			var starting_offset_frequency = %[START_OFFSET_FREQ];
 		</script>
 		<script src="sdr.js"></script>
 		<script src="openwebrx.js"></script>

htdocs/openwebrx.js

@@ -988,6 +988,32 @@ function resize_waterfall_container(check_init)
 	canvas_container.style.height=(window.innerHeight-e("webrx-top-container").clientHeight-e("openwebrx-scale-container").clientHeight).toString()+"px";
 }
 
+
+audio_server_output_rate=11025;
+audio_client_resampling_factor=4;
+
+
+function audio_calculate_resampling(targetRate)
+{ //both at the server and the client
+	output_range_max = 12000;
+	output_range_min = 8000;
+	i = 1;
+	while(true)
+	{
+		audio_server_output_rate = Math.floor(targetRate / i);
+		if(audio_server_output_rate < output_range_min) 
+		{ 
+			audio_client_resampling_factor = audio_server_output_rate = 0; 
+			divlog("Your audio card sampling rate ("+targetRate.toString()+") is not supported.<br />Please change your operating system default settings in order to fix this.",1);
+		}
+		if(audio_server_output_rate >= output_range_min	&& audio_server_output_rate <= output_range_max) break; //okay, we're done
+		i++;
+	}
+	audio_client_resampling_factor=i;
+	console.log("audio_calculate_resampling() :: "+audio_client_resampling_factor.toString()+", "+audio_server_output_rate.toString());
+}
+
+
 debug_ws_data_received=0;
 max_clients_num=0;
 
@@ -1003,6 +1029,7 @@ function on_ws_recv(evt)
 	{
 		var stringData=arrayBufferToString(evt.data);
 		if(stringData.substring(0,16)=="CLIENT DE SERVER") divlog("Acknowledged WebSocket connection: "+stringData);
+		
 	}
 	if(firstChars=="AUD")
 	{
@@ -1021,6 +1048,7 @@ function on_ws_recv(evt)
 		else if(fft_compression="adpcm")
 		{
 			fft_codec.reset();
+			
 			var waterfall_i16=fft_codec.decode(new Uint8Array(evt.data,4));
 			var waterfall_f32=new Float32Array(waterfall_i16.length-COMPRESS_FFT_PAD_N);
 			for(var i=0;i<waterfall_i16.length;i++) waterfall_f32[i]=waterfall_i16[i+COMPRESS_FFT_PAD_N]/100;
@@ -1039,6 +1067,7 @@ function on_ws_recv(evt)
 				{
 					case "setup":
 						waterfall_init();
+						audio_preinit();
 						break;					
 					case "bandwidth":
 						bandwidth=parseInt(param[1]);
@@ -1154,7 +1183,7 @@ var mute = false;
 
 var audio_received = Array();
 var audio_buffer_index = 0;
-var audio_resampler=new sdrjs.RationalResamplerFF(4,1);
+var audio_resampler;
 var audio_codec=new sdrjs.ImaAdpcm();
 var audio_compression="unknown";
 var audio_node;
@@ -1199,7 +1228,7 @@ function audio_prepare(data)
 		audio_prepared_buffers.push(audio_rebuffer.take());
 		audio_buffer_current_count_debug++;
 	}
-	if(audio_buffering && audio_prepared_buffers.length>audio_buffering_fill_to) audio_buffering=false;
+	if(audio_buffering && audio_prepared_buffers.length>audio_buffering_fill_to) { console.log("buffers now: "+audio_prepared_buffers.length.toString()); audio_buffering=false; }
 }
 
 
@@ -1278,6 +1307,8 @@ var audio_buffer_progressbar_update_disabled=false;
 
 var audio_buffer_total_average_level=0;
 var audio_buffer_total_average_level_length=0;
+var audio_overrun_cnt = 0;
+var audio_underrun_cnt = 0;
 
 function audio_buffer_progressbar_update()
 {
@@ -1287,8 +1318,8 @@ function audio_buffer_progressbar_update()
 	var overrun=audio_buffer_value>audio_buffer_maximal_length_sec;
 	var underrun=audio_prepared_buffers.length==0;
 	var text="buffer";
-	if(overrun) text="overrun";
+	if(overrun) { text="overrun"; console.log("audio overrun, "+(++audio_overrun_cnt).toString()); }
-	if(underrun) text="underrun";
+	if(underrun) { text="underrun"; console.log("audio underrun, "+(++audio_underrun_cnt).toString()); }
 	if(overrun||underrun) 
 	{ 
 		audio_buffer_progressbar_update_disabled=true;
@@ -1377,13 +1408,26 @@ function webrx_set_param(what, value)
 	ws.send("SET "+what+"="+value.toString());
 }
 
-function audio_init()
+function parsehash()
 {
-	audio_debug_time_start=(new Date()).getTime();
+	if(h=window.location.hash) 
-	audio_debug_time_last_start=audio_debug_time_start;
+	{
+		h.substring(1).split(",").forEach(function(x){
+			harr=x.split("=");
+			console.log(harr);
+			if(harr[0]=="mod") starting_mod = harr[1];
+			if(harr[0]=="freq") {
+			console.log(parseInt(harr[1]));
+			console.log(center_freq);  
+			starting_offset_frequency = parseInt(harr[1])-center_freq;
+			}
+		});
+		
+	}
+}
 
-	//https://github.com/0xfe/experiments/blob/master/www/tone/js/sinewave.js
+function audio_preinit()
-	audio_initialized=1; // only tell on_ws_recv() not to call it again
+{
 	try 
 	{
 		window.AudioContext = window.AudioContext||window.webkitAudioContext;
@@ -1394,6 +1438,28 @@ function audio_init()
 		divlog('Your browser does not support Web Audio API, which is required for WebRX to run. Please upgrade to a HTML5 compatible browser.', 1);
 	}
 
+	//we send our setup packet
+	
+	parsehash();
+	 //needs audio_context.sampleRate to exist
+
+	audio_calculate_resampling(audio_context.sampleRate);
+	audio_resampler = new sdrjs.RationalResamplerFF(audio_client_resampling_factor,1);
+	ws.send("SET output_rate="+audio_server_output_rate.toString()+" action=start"); //now we'll get AUD packets as well
+
+}
+
+function audio_init()
+{
+	if(audio_client_resampling_factor==0) return; //if failed to find a valid resampling factor...
+
+	audio_debug_time_start=(new Date()).getTime();
+	audio_debug_time_last_start=audio_debug_time_start;
+
+	//https://github.com/0xfe/experiments/blob/master/www/tone/js/sinewave.js
+	audio_initialized=1; // only tell on_ws_recv() not to call it again
+
+
 	//on Chrome v36, createJavaScriptNode has been replaced by createScriptProcessor
 	createjsnode_function = (audio_context.createJavaScriptNode == undefined)?audio_context.createScriptProcessor.bind(audio_context):audio_context.createJavaScriptNode.bind(audio_context);
 	audio_node = createjsnode_function(audio_buffer_size, 0, 1);
@@ -1411,7 +1477,14 @@ function audio_init()
    audio_buffer = audio_context.createBuffer(xhr.response, false);
 	audio_source.buffer = buffer;
 	audio_source.noteOn(0);*/
-	demodulator_analog_replace('nfm'); //needs audio_context.sampleRate to exist
+	demodulator_analog_replace(starting_mod);
+	if(starting_offset_frequency)  
+	{ 
+		demodulators[0].offset_frequency = starting_offset_frequency;
+		demodulators[0].set();
+		mkscale();
+	}
+
 	//hide log panel in a second (if user has not hidden it yet)
 	window.setTimeout(function(){
 		if(typeof e("openwebrx-panel-log").openwebrxHidden == "undefined" && !was_error)

openwebrx.py

@@ -29,6 +29,7 @@ import plugins
 import plugins.dsp
 import thread
 import time
+import datetime
 import subprocess
 import os 
 from BaseHTTPServer import BaseHTTPRequestHandler, HTTPServer
@@ -47,7 +48,6 @@ import ctypes
 #import rtl_mus
 import rxws
 import uuid
-import config_webrx as cfg
 import signal
 import socket
 
@@ -75,11 +75,22 @@ class MultiThreadHTTPServer(ThreadingMixIn, HTTPServer):
     pass 
 
 def handle_signal(signal, frame):
+	global spectrum_dsp
 	print "[openwebrx] Ctrl+C: aborting."
+	cleanup_clients(True)
+	spectrum_dsp.stop()
 	os._exit(1) #not too graceful exit
 
+def access_log(data):
+	global logs
+	logs.access_log.write("["+datetime.datetime.now().isoformat()+"] "+data+"\n")
+	logs.access_log.flush()
+
+receiver_failed=spectrum_thread_watchdog_last_tick=rtl_thread=spectrum_dsp=server_fail=None
+
 def main():
-	global clients, clients_mutex, pypy, lock_try_time, avatar_ctime
+	global clients, clients_mutex, pypy, lock_try_time, avatar_ctime, cfg, logs
+	global serverfail, rtl_thread
 	print
 	print "OpenWebRX - Open Source SDR Web App for Everyone!  | for license see LICENSE file in the package"
 	print "_________________________________________________________________________________________________"
@@ -87,6 +98,15 @@ def main():
 	print "Author contact info:    Andras Retzler, HA7ILM <randras@sdr.hu>"
 	print 
 
+	no_arguments=len(sys.argv)==1
+	if no_arguments: print "[openwebrx-main] Configuration script not specified. I will use: \"config_webrx.py\""
+	cfg=__import__("config_webrx" if no_arguments else sys.argv[1])
+	for option in ("access_log","csdr_dynamic_bufsize","csdr_print_bufsizes","csdr_through"): 
+		if not option in dir(cfg): setattr(cfg, option, False) #initialize optional config parameters
+
+	#Open log files
+	logs = type("logs_class", (object,), {"access_log":open(cfg.access_log if cfg.access_log else "/dev/null","a"), "error_log":""})()
+	
 	#Set signal handler
 	signal.signal(signal.SIGINT, handle_signal) #http://stackoverflow.com/questions/1112343/how-do-i-capture-sigint-in-python
 
@@ -107,16 +127,14 @@ def main():
 		pass
 
 	#Start rtl thread
+	if os.system("ncat --version > /dev/null") == 32512: #check for ncat
+		print "[openwebrx-main] Error: ncat not detected, please install it! The ncat tool is a netcat alternative, used for distributing the I/Q data stream. It is usually available in the nmap package (sudo apt-get install nmap). For more explanation, look into the README.md"
+		return 
 	if cfg.start_rtl_thread:
-		rtl_thread=threading.Thread(target = lambda:subprocess.Popen(cfg.start_rtl_command, shell=True), args=())
+		cfg.start_rtl_command += "| ncat -4l %d -k --send-only --allow 127.0.0.1" % cfg.iq_server_port
+		rtl_thread=threading.Thread(target = lambda:subprocess.Popen(cfg.start_rtl_command, shell=True),  args=())
 		rtl_thread.start()
-		print "[openwebrx-main] Started rtl thread: "+cfg.start_rtl_command
+		print "[openwebrx-main] Started rtl_thread: "+cfg.start_rtl_command
-
-	#Run rtl_mus.py in a different OS thread
-	python_command="pypy" if pypy else "python2"
-	rtl_mus_thread=threading.Thread(target = lambda:subprocess.Popen(python_command+" rtl_mus.py config_rtl", shell=True), args=())
-	rtl_mus_thread.start() # The new feature in GNU Radio 3.7: top_block() locks up ALL python threads until it gets the TCP connection.
-	print "[openwebrx-main] Started rtl_mus."
 	time.sleep(1) #wait until it really starts	
 
 	#Initialize clients
@@ -136,6 +154,8 @@ def main():
 	print "[openwebrx-main] Starting spectrum thread."
 	spectrum_thread=threading.Thread(target = spectrum_thread_function, args = ())
 	spectrum_thread.start()
+	spectrum_watchdog_thread=threading.Thread(target = spectrum_watchdog_thread_function, args = ())
+	spectrum_watchdog_thread.start()
 
 	get_cpu_usage()
 	bcastmsg_thread=threading.Thread(target = bcastmsg_thread_function, args = ())
@@ -146,13 +166,14 @@ def main():
 	#Start sdr.hu update thread
 	if sdrhu and cfg.sdrhu_key and cfg.sdrhu_public_listing: 
 		print "[openwebrx-main] Starting sdr.hu update thread..."
+		avatar_ctime=str(os.path.getctime("htdocs/gfx/openwebrx-avatar.png"))
 		sdrhu_thread=threading.Thread(target = sdrhu.run, args = ())
 		sdrhu_thread.start()
-		avatar_ctime=str(os.path.getctime("htdocs/gfx/openwebrx-avatar.png"))
 	
 	#Start HTTP thread
 	httpd = MultiThreadHTTPServer(('', cfg.web_port), WebRXHandler)
 	print('[openwebrx-main] Starting HTTP server.')
+	access_log("Starting OpenWebRX...")
 	httpd.serve_forever()
 
 
@@ -204,27 +225,61 @@ def mutex_watchdog_thread_function():
 	while True:
 		if lock_try_time != 0 and time.time()-lock_try_time > 3.0: 
 			#if 3 seconds pass without unlock
-			print "[openwebrx-watchdog] Mutex unlock timeout. Locker: \""+str(clients_mutex_locker)+"\" Now unlocking..." 
+			print "[openwebrx-mutex-watchdog] Mutex unlock timeout. Locker: \""+str(clients_mutex_locker)+"\" Now unlocking..." 
 			clients_mutex.release()
 		time.sleep(0.5)	
-		
+
+def spectrum_watchdog_thread_function():
+	global spectrum_thread_watchdog_last_tick, receiver_failed
+	while True:
+		time.sleep(60)
+		if spectrum_thread_watchdog_last_tick and time.time()-spectrum_thread_watchdog_last_tick > 60.0: 
+			print "[openwebrx-spectrum-watchdog] Spectrum timeout. Seems like no I/Q data is coming from the receiver.\nIf you're using RTL-SDR, the receiver hardware may randomly fail under some circumstances:\n1) high temperature,\n2) insufficient current available from the USB port."
+			print "[openwebrx-spectrum-watchdog] Deactivating receiver."
+			receiver_failed="spectrum"
+			return
+
+def check_server():
+	global spectrum_dsp, server_fail, rtl_thread
+	if server_fail: return server_fail	
+	#print spectrum_dsp.process.poll()
+	if spectrum_dsp and spectrum_dsp.process.poll()!=None: server_fail = "spectrum_thread dsp subprocess failed"
+	#if rtl_thread and not rtl_thread.is_alive(): server_fail = "rtl_thread failed"
+	if server_fail: print "[openwebrx-check_server] >>>>>>> ERROR:", server_fail
+	return server_fail
+
+def apply_csdr_cfg_to_dsp(dsp):
+	dsp.csdr_dynamic_bufsize = cfg.csdr_dynamic_bufsize
+	dsp.csdr_print_bufsizes = cfg.csdr_print_bufsizes
+	dsp.csdr_through = cfg.csdr_through	
+
 def spectrum_thread_function():
-	global clients
+	global clients, spectrum_dsp, spectrum_thread_watchdog_last_tick
-	dsp=getattr(plugins.dsp,cfg.dsp_plugin).plugin.dsp_plugin()
+	spectrum_dsp=dsp=getattr(plugins.dsp,cfg.dsp_plugin).plugin.dsp_plugin()
+	dsp.nc_port=cfg.iq_server_port
 	dsp.set_demodulator("fft")
 	dsp.set_samp_rate(cfg.samp_rate)
 	dsp.set_fft_size(cfg.fft_size)
 	dsp.set_fft_fps(cfg.fft_fps)
 	dsp.set_fft_compression(cfg.fft_compression)
 	dsp.set_format_conversion(cfg.format_conversion)
+	apply_csdr_cfg_to_dsp(dsp)
 	sleep_sec=0.87/cfg.fft_fps
 	print "[openwebrx-spectrum] Spectrum thread initialized successfully."
 	dsp.start()
+	if cfg.csdr_dynamic_bufsize: 
+		dsp.read(8) #dummy read to skip bufsize & preamble
+		print "[openwebrx-spectrum] Note: CSDR_DYNAMIC_BUFSIZE_ON = 1"
 	print "[openwebrx-spectrum] Spectrum thread started." 
 	bytes_to_read=int(dsp.get_fft_bytes_to_read())
+	spectrum_thread_counter=0
 	while True:
 		data=dsp.read(bytes_to_read)
 		#print "gotcha",len(data),"bytes of spectrum data via spectrum_thread_function()"
+		if spectrum_thread_counter >= cfg.fft_fps: 
+			spectrum_thread_counter=0
+			spectrum_thread_watchdog_last_tick = time.time() #once every second
+		else: spectrum_thread_counter+=1
 		cma("spectrum_thread")
 		correction=0
 		for i in range(0,len(clients)):
@@ -255,16 +310,17 @@ def get_client_by_id(client_id, use_mutex=True):
 def log_client(client, what):
 	print "[openwebrx-httpd] client {0}#{1} :: {2}".format(client.ip,client.id,what)
 
-def cleanup_clients():
+def cleanup_clients(end_all=False):
-	# if client doesn't open websocket for too long time, we drop it
+	# - if a client doesn't open websocket for too long time, we drop it
+	# - or if end_all is true, we drop all clients
 	global clients
 	cma("cleanup_clients")
 	correction=0
 	for i in range(0,len(clients)):
 		i-=correction
 		#print "cleanup_clients:: len(clients)=", len(clients), "i=", i
-		if (not clients[i].ws_started) and (time.time()-clients[i].gen_time)>45:
+		if end_all or ((not clients[i].ws_started) and (time.time()-clients[i].gen_time)>45):
-			print "[openwebrx] cleanup_clients :: client timeout to open WebSocket"
+			if not end_all: print "[openwebrx] cleanup_clients :: client timeout to open WebSocket"
 			close_client(i, False)
 			correction+=1
 	cmr()
@@ -299,6 +355,7 @@ def close_client(i, use_mutex=True):
 		print "[openwebrx] close_client dsp.stop() :: error -",exc_type,exc_value
 		traceback.print_tb(exc_traceback)
 	clients[i].closed[0]=True
+	access_log("Stopped streaming to client: "+clients[i].ip+"#"+str(clients[i].id)+" (users now: "+str(len(clients)-1)+")")
 	del clients[i]
 	if use_mutex: cmr()
 
@@ -319,18 +376,20 @@ class WebRXHandler(BaseHTTPRequestHandler):
 
 	def do_GET(self):
 		self.connection.setsockopt(socket.IPPROTO_TCP, socket.TCP_NODELAY, 1)
-		global dsp_plugin, clients_mutex, clients, avatar_ctime, sw_version
+		global dsp_plugin, clients_mutex, clients, avatar_ctime, sw_version, receiver_failed
 		rootdir = 'htdocs' 
 		self.path=self.path.replace("..","")
 		path_temp_parts=self.path.split("?")
 		self.path=path_temp_parts[0]
 		request_param=path_temp_parts[1] if(len(path_temp_parts)>1) else "" 
+		access_log("GET "+self.path+" from "+self.client_address[0])
 		try:
 			if self.path=="/":
 				self.path="/index.wrx"
 			# there's even another cool tip at http://stackoverflow.com/questions/4419650/how-to-implement-timeout-in-basehttpserver-basehttprequesthandler-python
 			#if self.path[:5]=="/lock": cma("do_GET /lock/") # to test mutex_watchdog_thread. Do not uncomment in production environment!
 			if self.path[:4]=="/ws/":
+				if receiver_failed: self.send_error(500,"Internal server error")
 				try:
 					# ========= WebSocket handshake  =========
 					ws_success=True
@@ -363,23 +422,26 @@ class WebRXHandler(BaseHTTPRequestHandler):
 
 					# ========= Initialize DSP =========
 					dsp=getattr(plugins.dsp,cfg.dsp_plugin).plugin.dsp_plugin()
-					dsp.set_samp_rate(cfg.samp_rate)
+					dsp_initialized=False
-					dsp.set_demodulator("nfm")
-					dsp.set_offset_freq(0)
-					dsp.set_bpf(-4000,4000)
 					dsp.set_audio_compression(cfg.audio_compression)
 					dsp.set_format_conversion(cfg.format_conversion)
-					dsp.start()
+					dsp.set_offset_freq(0)
+					dsp.set_bpf(-4000,4000)
+					dsp.nc_port=cfg.iq_server_port
+					apply_csdr_cfg_to_dsp(dsp)
 					myclient.dsp=dsp
 					
+					access_log("Started streaming to client: "+self.client_address[0]+"#"+myclient.id+" (users now: "+str(len(clients))+")")
+
 					while True:
 						if myclient.closed[0]: 
 							print "[openwebrx-httpd:ws] client closed by other thread"
 							break
 
 						# ========= send audio =========
-						temp_audio_data=dsp.read(256)
+						if dsp_initialized:
-						rxws.send(self, temp_audio_data, "AUD ")
+							temp_audio_data=dsp.read(256)
+							rxws.send(self, temp_audio_data, "AUD ")
 
 						# ========= send spectrum =========
 						while not myclient.spectrum_queue.empty():
@@ -417,9 +479,17 @@ class WebRXHandler(BaseHTTPRequestHandler):
 										dsp.set_offset_freq(int(param_value))
 									elif param_name=="mod":
 										if (dsp.get_demodulator()!=param_value):
-											dsp.stop()
+											if dsp_initialized: dsp.stop()
 											dsp.set_demodulator(param_value)
+											if dsp_initialized: dsp.start()
+									elif param_name == "output_rate":
+										if not dsp_initialized: 
+											dsp.set_output_rate(int(param_value))
+											dsp.set_samp_rate(cfg.samp_rate)
+									elif param_name=="action" and param_value=="start":
+										if not dsp_initialized: 
 											dsp.start()
+											dsp_initialized=True
 									else:
 										print "[openwebrx-httpd:ws] invalid parameter"
 								if bpf_set:
@@ -457,13 +527,17 @@ class WebRXHandler(BaseHTTPRequestHandler):
 			elif self.path in ("/status", "/status/"):
 				#self.send_header('Content-type','text/plain')
 				getbands=lambda: str(int(cfg.shown_center_freq-cfg.samp_rate/2))+"-"+str(int(cfg.shown_center_freq+cfg.samp_rate/2))
-				self.wfile.write("status=active\nname="+cfg.receiver_name+"\nsdr_hw="+cfg.receiver_device+"\nop_email="+cfg.receiver_admin+"\nbands="+getbands()+"\nusers="+str(len(clients))+"\navatar_ctime="+avatar_ctime+"\ngps="+str(cfg.receiver_gps)+"\nasl="+str(cfg.receiver_asl)+"\nloc="+cfg.receiver_location+"\nsw_version="+sw_version+"\nantenna="+cfg.receiver_ant+"\n")
+				self.wfile.write("status="+("inactive" if receiver_failed else "active")+"\nname="+cfg.receiver_name+"\nsdr_hw="+cfg.receiver_device+"\nop_email="+cfg.receiver_admin+"\nbands="+getbands()+"\nusers="+str(len(clients))+"\nusers_max="+str(cfg.max_clients)+"\navatar_ctime="+avatar_ctime+"\ngps="+str(cfg.receiver_gps)+"\nasl="+str(cfg.receiver_asl)+"\nloc="+cfg.receiver_location+"\nsw_version="+sw_version+"\nantenna="+cfg.receiver_ant+"\n")
 				print "[openwebrx-httpd] GET /status/ from",self.client_address[0]			
 			else:
 				f=open(rootdir+self.path)
 				data=f.read()
 				extension=self.path[(len(self.path)-4):len(self.path)]
 				extension=extension[2:] if extension[1]=='.' else extension[1:]
+				checkresult=check_server()
+				if extension == "wrx" and (checkresult or receiver_failed):
+					self.send_302("inactive.html")
+					return
 				if extension == "wrx" and ((self.headers['user-agent'].count("Chrome")==0 and self.headers['user-agent'].count("Firefox")==0 and (not "Googlebot" in self.headers['user-agent'])) if 'user-agent' in self.headers.keys() else True) and (not request_param.count("unsupported")):
 					self.send_302("upgrade.html")
 					return
@@ -492,7 +566,9 @@ class WebRXHandler(BaseHTTPRequestHandler):
 						("%[RX_ADMIN]",cfg.receiver_admin),
 						("%[RX_ANT]",cfg.receiver_ant),
 						("%[RX_DEVICE]",cfg.receiver_device),
-						("%[AUDIO_BUFSIZE]",str(cfg.client_audio_buffer_size))
+						("%[AUDIO_BUFSIZE]",str(cfg.client_audio_buffer_size)),
+						("%[START_OFFSET_FREQ]",str(cfg.start_freq-cfg.center_freq)),
+						("%[START_MOD]",cfg.start_mod)
 					)
 					for rule in replace_dictionary:
 						while data.find(rule[0])!=-1:

plugins/dsp/csdr/plugin.py

@@ -44,13 +44,17 @@ class dsp_plugin:
 		self.demodulator = "nfm"
 		self.name = "csdr"
 		self.format_conversion = "csdr convert_u8_f"
-		try:	
+		self.base_bufsize = 512
-			subprocess.Popen("nc",stdout=subprocess.PIPE,stderr=subprocess.PIPE).kill()
+		self.nc_port = 4951
-		except:
+		self.csdr_dynamic_bufsize = False
-			print "[openwebrx-plugin:csdr] error: netcat not found, please install netcat!"
+		self.csdr_print_bufsizes = False
+		self.csdr_through = False
 
 	def chain(self,which):
-		any_chain_base="nc -v localhost 4951 | "+self.format_conversion+(" | " if  self.format_conversion!="" else "")+"csdr flowcontrol {flowcontrol} 10 | "
+		any_chain_base="ncat -v 127.0.0.1 {nc_port} | "
+		if self.csdr_dynamic_bufsize: any_chain_base+="csdr setbuf {start_bufsize} | "
+		if self.csdr_through: any_chain_base+="csdr through | "
+		any_chain_base+=self.format_conversion+(" | " if  self.format_conversion!="" else "") ##"csdr flowcontrol {flowcontrol} auto 1.5 10 | "
 		if which == "fft": 
 			fft_chain_base = "sleep 1; "+any_chain_base+"csdr fft_cc {fft_size} {fft_block_size} | csdr logpower_cf -70 | csdr fft_exchange_sides_ff {fft_size}"
 			if self.fft_compression=="adpcm":
@@ -61,9 +65,9 @@ class dsp_plugin:
 		chain_end = "" 
 		if self.audio_compression=="adpcm":
 			chain_end = " | csdr encode_ima_adpcm_i16_u8"
-		if which == "nfm": return chain_begin + "csdr fmdemod_quadri_cf | csdr limit_ff | csdr fractional_decimator_ff {last_decimation} | csdr deemphasis_nfm_ff 11025 | csdr fastagc_ff | csdr convert_f_i16"+chain_end
+		if which == "nfm": return chain_begin + "csdr fmdemod_quadri_cf | csdr limit_ff | csdr fractional_decimator_ff {last_decimation} | csdr deemphasis_nfm_ff 11025 | csdr fastagc_ff 1024 | csdr convert_f_i16"+chain_end
 		elif which == "am":	return chain_begin + "csdr amdemod_cf | csdr fastdcblock_ff | csdr fractional_decimator_ff {last_decimation} | csdr agc_ff | csdr limit_ff | csdr convert_f_i16"+chain_end
-		elif which == "ssb": return chain_begin + "csdr realpart_cf | csdr fractional_decimator_ff {last_decimation} | csdr agc_ff | csdr limit_ff | csdr convert_f_i16"+chain_end
+		elif which == "ssb": return chain_begin + "csdr realpart_cf | csdr fractional_decimator_ff {last_decimation} | csdr agc_ff | csdr clipdetect_ff | csdr limit_ff | csdr convert_f_i16"+chain_end
 
 	def set_audio_compression(self,what):
 		self.audio_compression = what
@@ -92,6 +96,10 @@ class dsp_plugin:
 	def get_output_rate(self):
 		return self.output_rate
 
+	def set_output_rate(self,output_rate):
+		self.output_rate=output_rate
+		self.set_samp_rate(self.samp_rate) #as it depends on output_rate
+
 	def set_demodulator(self,demodulator):
 		#to change this, restart is required
 		self.demodulator=demodulator
@@ -153,10 +161,17 @@ class dsp_plugin:
 			self.mkfifo(self.shift_pipe)
 
 		#run the command
-		command=command_base.format(bpf_pipe=self.bpf_pipe,shift_pipe=self.shift_pipe,decimation=self.decimation,last_decimation=self.last_decimation,fft_size=self.fft_size,fft_block_size=self.fft_block_size(),bpf_transition_bw=float(self.bpf_transition_bw)/self.if_samp_rate(),ddc_transition_bw=self.ddc_transition_bw(),flowcontrol=int(self.samp_rate*4*2*1.5))
+		command=command_base.format( bpf_pipe=self.bpf_pipe, shift_pipe=self.shift_pipe, decimation=self.decimation, \
+			last_decimation=self.last_decimation, fft_size=self.fft_size, fft_block_size=self.fft_block_size(), \
+			bpf_transition_bw=float(self.bpf_transition_bw)/self.if_samp_rate(), ddc_transition_bw=self.ddc_transition_bw(), \
+			flowcontrol=int(self.samp_rate*2), start_bufsize=self.base_bufsize*self.decimation, nc_port=self.nc_port)
+
 		print "[openwebrx-dsp-plugin:csdr] Command =",command
 		#code.interact(local=locals())
-		self.process = subprocess.Popen(command, stdout=subprocess.PIPE, shell=True, preexec_fn=os.setpgrp)
+		my_env=os.environ.copy()
+		if self.csdr_dynamic_bufsize: my_env["CSDR_DYNAMIC_BUFSIZE_ON"]="1";
+		if self.csdr_print_bufsizes: my_env["CSDR_PRINT_BUFSIZES"]="1";
+		self.process = subprocess.Popen(command, stdout=subprocess.PIPE, shell=True, preexec_fn=os.setpgrp, env=my_env)
 		self.running = True
 
 		#open control pipes for csdr and send initialization data
@@ -179,14 +194,16 @@ class dsp_plugin:
 		#	os.killpg(self.process.pid, signal.SIGTERM) 
 		#	
 		#	time.sleep(0.1)
-		try:
+		if self.bpf_pipe:
-			os.unlink(self.bpf_pipe)
+			try:
-		except:
+				os.unlink(self.bpf_pipe)
-			print "[openwebrx-dsp-plugin:csdr] stop() :: unlink failed: " + self.bpf_pipe
+			except:
-		try:
+				print "[openwebrx-dsp-plugin:csdr] stop() :: unlink failed: " + self.bpf_pipe
-			os.unlink(self.shift_pipe)
+		if self.shift_pipe:
-		except:
+			try:
-			print "[openwebrx-dsp-plugin:csdr] stop() :: unlink failed: " + self.bpf_pipe
+				os.unlink(self.shift_pipe)
+			except:
+				print "[openwebrx-dsp-plugin:csdr] stop() :: unlink failed: " + self.bpf_pipe
 		self.running = False
 
 	def restart(self):

rtl_mus.py

@@ -1,539 +0,0 @@
-'''
-	This file is part of RTL Multi-User Server, 
-	that makes multi-user access to your DVB-T dongle used as an SDR.
-	Copyright (c) 2013-2015 by Andras Retzler <randras@sdr.hu>
-
-    This program is free software: you can redistribute it and/or modify
-    it under the terms of the GNU Affero General Public License as
-    published by the Free Software Foundation, either version 3 of the
-    License, or (at your option) any later version.
-
-    This program is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU Affero General Public License for more details.
-
-    You should have received a copy of the GNU Affero General Public License
-    along with this program.  If not, see <http://www.gnu.org/licenses/>.
-
------
-
-2013-11?  Asyncore version
-2014-03   Fill with null on no data
-
-'''
-
-import socket 
-import sys
-import array
-import time
-import logging
-import os
-import time
-import subprocess
-import fcntl
-import thread
-import pdb
-import asyncore
-import multiprocessing
-import signal
-#pypy compatiblity
-try: import dl
-except: pass
-
-import code
-import traceback
-
-def handle_signal(signal, frame):
-	log.info("Ctrl+C: aborting.")
-	os._exit(1) #not too graceful exit
-
-def ip_match(this,ip_ranges,for_allow):
-	if not len(ip_ranges):
-		return 1 #empty list matches all ip addresses
-	for ip_range in ip_ranges:
-		#print this[0:len(ip_range)], ip_range
-		if this[0:len(ip_range)]==ip_range:
-			return 1
-	return 0
-
-def ip_access_control(ip):
-	if(not cfg.use_ip_access_control): return 1
-	allowed=0
-	if(cfg.order_allow_deny):
-		if ip_match(ip,cfg.allowed_ip_ranges,1): allowed=1
-		if ip_match(ip,cfg.denied_ip_ranges,0): allowed=0
-	else:
-		if ip_match(ip,cfg.denied_ip_ranges,0): 
-			allowed=0
-		if ip_match(ip,cfg.allowed_ip_ranges,1): 
-			allowed=1
-	return allowed
-
-def add_data_to_clients(new_data):
-	# might be called from:
-	# -> dsp_read
-	# -> rtl_tcp_asyncore.handle_read
-	global clients
-	global clients_mutex
-	clients_mutex.acquire()
-	for client in clients:
-		#print "client %d size: %d"%(client[0].ident,client[0].waiting_data.qsize())
-		if(client[0].waiting_data.full()):
-			if cfg.cache_full_behaviour == 0:
-				log.error("client cache full, dropping samples: "+str(client[0].ident)+"@"+client[0].socket[1][0])
-				while not client[0].waiting_data.empty(): # clear queue
-					client[0].waiting_data.get(False, None)
-			elif cfg.cache_full_behaviour == 1:
-				#rather closing client:
-				log.error("client cache full, dropping client: "+str(client[0].ident)+"@"+client[0].socket[1][0])
-				client[0].close(False)
-			elif cfg.cache_full_behaviour == 2:
-				pass #client cache full, just not taking care
-			else: log.error("invalid value for cfg.cache_full_behaviour")
-		else:
-			client[0].waiting_data.put(new_data)
-	clients_mutex.release()
-
-def dsp_read_thread():
-	global proc
-	global dsp_data_count
-	while True:
-		try:
-			my_buffer=proc.stdout.read(1024)
-		except IOError:
-			log.error("DSP subprocess is not ready for reading.")
-			time.sleep(1)
-			continue
-		add_data_to_clients(my_buffer)
-		if cfg.debug_dsp_command:
-			dsp_data_count+=len(my_buffer)	
-
-def dsp_write_thread():
-	global proc
-	global dsp_input_queue
-	global original_data_count
-	while True:
-		try:
-			my_buffer=dsp_input_queue.get(timeout=0.3)
-		except:
-			continue
-		proc.stdin.write(my_buffer)
-		proc.stdin.flush()
-		if cfg.debug_dsp_command:
-			original_data_count+=len(my_buffer)
-
-class client_handler(asyncore.dispatcher):
-
-	def __init__(self,client_param):
-		self.client=client_param
-		self.client[0].asyncore=self
-		self.sent_dongle_id=False
-		self.last_waiting_buffer=""
-		asyncore.dispatcher.__init__(self, self.client[0].socket[0])
-		self.socket.setsockopt(socket.IPPROTO_TCP, socket.TCP_NODELAY, 1)
-
-	def handle_read(self):
-		global commands
-		new_command = self.recv(5)
-		if len(new_command)>=5:
-			if handle_command(new_command, self.client):
-				commands.put(new_command)
-
-	def handle_error(self):
-		exc_type, exc_value, exc_traceback = sys.exc_info()
-		log.info("client error: "+str(self.client[0].ident)+"@"+self.client[0].socket[1][0])
-		traceback.print_tb(exc_traceback)
-		self.close()
-
-	def handle_close(self):
-		self.client[0].close()
-		log.info("client disconnected: "+str(self.client[0].ident)+"@"+self.client[0].socket[1][0])
-
-	def writable(self):
-		#print "queryWritable",not self.client[0].waiting_data.empty()
-		return not self.client[0].waiting_data.empty()
-
-	def handle_write(self):
-		global last_waiting
-		global rtl_dongle_identifier
-		global sample_rate
-		if not self.sent_dongle_id:
-			self.send(rtl_dongle_identifier)
-			self.sent_dongle_id=True
-			return
-		#print "write2client",self.client[0].waiting_data.qsize()
-		next=self.last_waiting_buffer+self.client[0].waiting_data.get()
-		sent=asyncore.dispatcher.send(self, next)
-		self.last_waiting_buffer=next[sent:]
-
-class server_asyncore(asyncore.dispatcher):
-
-	def __init__(self):
-		asyncore.dispatcher.__init__(self)
-		self.create_socket(socket.AF_INET, socket.SOCK_STREAM)
-		self.set_reuse_addr()
-		self.bind((cfg.my_ip, cfg.my_listening_port))
-		self.listen(5)
-		self.socket.setsockopt(socket.IPPROTO_TCP, socket.TCP_NODELAY, 1)
-		log.info("Server listening on port: "+str(cfg.my_listening_port))
-
-	def handle_accept(self):
-		global max_client_id
-		global clients_mutex
-		global clients
-		my_client=[client()]
-		my_client[0].socket=self.accept()
-		if (my_client[0].socket is None): # not sure if required
-			return 
-		if (ip_access_control(my_client[0].socket[1][0])):
-			my_client[0].ident=max_client_id
-			max_client_id+=1
-			my_client[0].start_time=time.time()
-			my_client[0].waiting_data=multiprocessing.Queue(500)
-			clients_mutex.acquire()
-			clients.append(my_client)
-			clients_mutex.release()
-			handler = client_handler(my_client)
-			log.info("client accepted: "+str(len(clients)-1)+"@"+my_client[0].socket[1][0]+":"+str(my_client[0].socket[1][1])+"  users now: "+str(len(clients)))
-		else:
-			log.info("client denied: "+str(len(clients)-1)+"@"+my_client[0].socket[1][0]+":"+str(my_client[0].socket[1][1])+" blocked by ip")
-			my_client.socket.close()
-
-rtl_tcp_resetting=False #put me away
-
-def rtl_tcp_asyncore_reset(timeout):
-	global rtl_tcp_core
-	global rtl_tcp_resetting
-	if rtl_tcp_resetting: return
-	#print "rtl_tcp_asyncore_reset"
-	rtl_tcp_resetting=True
-	time.sleep(timeout)
-	try:
-		rtl_tcp_core.close()
-	except:
-		pass
-	try:
-		del rtl_tcp_core
-	except:
-		pass
-	rtl_tcp_core=rtl_tcp_asyncore()
-	#print asyncore.socket_map
-	rtl_tcp_resetting=False
-
-class rtl_tcp_asyncore(asyncore.dispatcher):
-	def __init__(self):
-		global server_missing_logged
-		asyncore.dispatcher.__init__(self)
-		self.password_sent = False
-		self.ok=True
-		self.create_socket(socket.AF_INET, socket.SOCK_STREAM)
-		try:		
-			self.connect((cfg.rtl_tcp_host, cfg.rtl_tcp_port))
-			self.socket.settimeout(0.1)
-		except:
-			log.error("rtl_tcp connection refused. Retrying.")
-			thread.start_new_thread(rtl_tcp_asyncore_reset, (1,))
-			self.close()
-			return
-
-	def handle_error(self):
-		global server_missing_logged
-		global rtl_tcp_connected
-		rtl_tcp_connected=False
-		exc_type, exc_value, exc_traceback = sys.exc_info()
-		self.ok=False
-		server_is_missing=hasattr(exc_value,"errno") and exc_value.errno==111
-		if (not server_is_missing) or (not server_missing_logged):
-			log.error("with rtl_tcp host connection: "+str(exc_value))
-			#traceback.print_tb(exc_traceback)
-			server_missing_logged|=server_is_missing
-		try:
-			self.close()
-		except:
-			pass
-		thread.start_new_thread(rtl_tcp_asyncore_reset, (2,))
-
-	def handle_connect(self):
-		global server_missing_logged
-		global rtl_tcp_connected
-		self.socket.settimeout(0.1)
-		self.password_sent = False
-		rtl_tcp_connected=True
-		if self.ok:
-			log.info("rtl_tcp host connection estabilished")
-			server_missing_logged=False
-
-	def handle_close(self):
-		global rtl_tcp_connected
-		global rtl_tcp_core
-		rtl_tcp_connected=False
-		log.error("rtl_tcp host connection has closed, now trying to reopen")
-		try:
-			self.close()
-		except:
-			pass
-		thread.start_new_thread(rtl_tcp_asyncore_reset, (2,))
-
-	def handle_read(self):
-		global rtl_dongle_identifier
-		global dsp_input_queue
-		global watchdog_data_count
-		if(len(rtl_dongle_identifier)==0):
-			rtl_dongle_identifier=self.recv(12)
-			return
-		new_data_buffer=self.recv(1024*16)
-		if cfg.watchdog_interval:
-			watchdog_data_count+=1024*16
-		if cfg.use_dsp_command:
-			dsp_input_queue.put(new_data_buffer)
-			#print "did put anyway"
-		else:
-			add_data_to_clients(new_data_buffer)
-
-	def writable(self):
-		
-		#check if any new commands to write
-		global commands
-		return (not self.password_sent and cfg.rtl_tcp_password != None) or not commands.empty()
-
-	def handle_write(self):
-		if(not self.password_sent and cfg.rtl_tcp_password != None):
-			log.info("Sending rtl_tcp_password...")
-			self.send(cfg.rtl_tcp_password)
-			self.password_sent = True
-		global commands
-		while not commands.empty():
-			mcmd=commands.get()
-			self.send(mcmd)
-
-def xxd(data):
-	#diagnostic purposes only
-	output=""
-	for d in data:
-		output+=hex(ord(d))[2:].zfill(2)+" " 
-	return output
-
-def handle_command(command, client_param):
-	global sample_rate
-	client=client_param[0]
-	param=array.array("I", command[1:5])[0]
-	param=socket.ntohl(param)
-	command_id=ord(command[0])
-	client_info=str(client.ident)+"@"+client.socket[1][0]+":"+str(client.socket[1][1])
-	if(time.time()-client.start_time<cfg.client_cant_set_until and not (cfg.first_client_can_set and client.ident==0) ):
-		log.info("deny: "+client_info+" -> client can't set anything until "+str(cfg.client_cant_set_until)+" seconds")
-		return 0
-	if command_id == 1:
-		if max(map((lambda r: param>=r[0] and param<=r[1]),cfg.freq_allowed_ranges)):
-			log.debug("allow: "+client_info+" -> set freq "+str(param))
-			return 1
-		else:
-			log.debug("deny: "+client_info+" -> set freq - out of range: "+str(param))
-	elif command_id == 2:
-		log.debug("deny: "+client_info+" -> set sample rate: "+str(param))
-		sample_rate=param
-		return 0 # ordinary clients are not allowed to do this
-	elif command_id == 3:
-		log.debug("deny/allow: "+client_info+" -> set gain mode: "+str(param))
-		return cfg.allow_gain_set
-	elif command_id == 4:
-		log.debug("deny/allow: "+client_info+" -> set gain: "+str(param))
-		return cfg.allow_gain_set 
-	elif command_id == 5:
-		log.debug("deny: "+client_info+" -> set freq correction: "+str(param))
-		return 0 
-	elif command_id == 6:
-		log.debug("deny/allow: set if stage gain")
-		return cfg.allow_gain_set
-	elif command_id == 7:
-		log.debug("deny: set test mode")
-		return 0
-	elif command_id == 8:
-		log.debug("deny/allow: set agc mode")
-		return cfg.allow_gain_set
-	elif command_id == 9:
-		log.debug("deny: set direct sampling")
-		return 0
-	elif command_id == 10:
-		log.debug("deny: set offset tuning")
-		return 0
-	elif command_id == 11:
-		log.debug("deny: set rtl xtal")
-		return 0
-	elif command_id == 12:
-		log.debug("deny: set tuner xtal")
-		return 0
-	elif command_id == 13:
-		log.debug("deny/allow: set tuner gain by index")
-		return cfg.allow_gain_set
-	else:
-		log.debug("deny: "+client_info+" sent an ivalid command: "+str(param))
-	return 0
-
-def watchdog_thread():
-	global rtl_tcp_connected
-	global rtl_tcp_core	
-	global watchdog_data_count
-	global sample_rate
-	zero_buffer_size=16348
-	second_frac=10
-	zero_buffer='\x7f'*zero_buffer_size
-	watchdog_data_count=0
-	rtl_tcp_connected=False
-	null_fill=False
-	time.sleep(4) # wait before activating this thread
-	log.info("watchdog started")
-	first_start=True
-	n=0
-	while True:
-		wait_altogether=cfg.watchdog_interval if rtl_tcp_connected or first_start else cfg.reconnect_interval	
-		first_start=False
-		if null_fill:
-			log.error("watchdog: filling buffer with zeros.")	
-			while wait_altogether>0:
-				wait_altogether-=1.0/second_frac
-				for i in range(0,((2*sample_rate)/second_frac)/zero_buffer_size):	
-					add_data_to_clients(zero_buffer)
-					n+=len(zero_buffer)
-					time.sleep(0) #yield
-					if watchdog_data_count: break
-				if watchdog_data_count: break
-				time.sleep(1.0/second_frac)
-				#print "sent altogether",n
-		else:
-			time.sleep(wait_altogether)
-		null_fill=not watchdog_data_count
-		if not watchdog_data_count:
-			log.error("watchdog: restarting rtl_tcp_asyncore() now.")
-			rtl_tcp_asyncore_reset(0)
-		watchdog_data_count=0
-			
-		
-
-def dsp_debug_thread():
-	global dsp_data_count
-	global original_data_count
-	while 1:	
-		time.sleep(1)
-		print "[rtl-mus] DSP | Original data: "+str(int(original_data_count/1000))+"kB/sec | Processed data: "+str(int(dsp_data_count/1000))+"kB/sec"
-		dsp_data_count = original_data_count=0
-		
-class client:
-	ident=None #id
-	to_close=False
-	waiting_data=None
-	start_time=None	
-	socket=None
-	asyncore=None
-
-	def close(self, use_mutex=True):
-		global clients_mutex
-		global clients
-		if use_mutex: clients_mutex.acquire()
-		correction=0
-		for i in range(0,len(clients)):
-			i-=correction
-			if clients[i][0].ident==self.ident:
-				try:
-					self.socket.close()
-				except:
-					pass
-				try:
-					self.asyncore.close()
-					del self.asyncore
-				except:
-					pass
-				del clients[i]
-				correction+=1
-		if use_mutex: clients_mutex.release()
-
-
-def main():
-	global server_missing_logged
-	global rtl_dongle_identifier
-	global log
-	global clients
-	global clients_mutex
-	global original_data_count
-	global dsp_input_queue
-	global dsp_data_count
-	global proc
-	global commands
-	global max_client_id
-	global rtl_tcp_core
-	global sample_rate
-
-	#Set signal handler
-	signal.signal(signal.SIGINT, handle_signal) #http://stackoverflow.com/questions/1112343/how-do-i-capture-sigint-in-python
-
-	# set up logging
-	log = logging.getLogger("rtl_mus")
-	log.setLevel(logging.DEBUG)
-	formatter = logging.Formatter('%(asctime)s %(levelname)s %(message)s')
-	stream_handler = logging.StreamHandler()
-	stream_handler.setLevel(logging.DEBUG)
-	stream_handler.setFormatter(formatter)
-	log.addHandler(stream_handler)
-	file_handler = logging.FileHandler(cfg.log_file_path)
-	file_handler.setLevel(logging.INFO)
-	file_handler.setFormatter(formatter)
-	log.addHandler(file_handler)
-	log.info("Server is UP")
-	
-	server_missing_logged=0	# Not to flood the screen with messages related to rtl_tcp disconnect
-	rtl_dongle_identifier='' # rtl_tcp sends some identifier on dongle type and gain values in the first few bytes right after connection
-	clients=[]
-	dsp_data_count=original_data_count=0
-	commands=multiprocessing.Queue()
-	dsp_input_queue=multiprocessing.Queue()
-	clients_mutex=multiprocessing.Lock()
-	max_client_id=0
-	sample_rate=250000 # so far only watchdog thread uses it to fill buffer up with zeros on missing input
-
-	# start dsp threads
-	if cfg.use_dsp_command:
-		print "[rtl_mus] Opening DSP process..."
-		proc = subprocess.Popen (cfg.dsp_command.split(" "), stdin = subprocess.PIPE, stdout = subprocess.PIPE) #!! should fix the split :-S
-		dsp_read_thread_v=thread.start_new_thread(dsp_read_thread, ())
-		dsp_write_thread_v=thread.start_new_thread(dsp_write_thread, ())
-		if cfg.debug_dsp_command:
-			dsp_debug_thread_v=thread.start_new_thread(dsp_debug_thread,())
-
-	# start watchdog thread
-	if cfg.watchdog_interval != 0:
-		watchdog_thread_v=thread.start_new_thread(watchdog_thread,())
-
-	# start asyncores
-	rtl_tcp_core = rtl_tcp_asyncore()
-	server_core = server_asyncore()
-
-	asyncore.loop(0.1)
-
-
-if __name__=="__main__":
-	print
-	print "rtl_mus: Multi-User I/Q Data Server for RTL-SDR v0.22, made at HA5KFU Amateur Radio Club (http://ha5kfu.hu)"
-	print "    code by Andras Retzler, HA7ILM <randras@sdr.hu>"
-	print "    distributed under GNU GPL v3"
-	print 
-
-	try:
-		for libcpath in ["/lib/i386-linux-gnu/libc.so.6","/lib/libc.so.6"]:
-			if os.path.exists(libcpath):
-				libc = dl.open(libcpath)
-				libc.call("prctl", 15, "rtl_mus", 0, 0, 0)
-				break
-	except:
-		pass
-
-	# === Load configuration script ===
-	if len(sys.argv)==1:
-		print "[rtl_mus] Warning! Configuration script not specified. I will use: \"config_rtl.py\""
-		config_script="config_rtl"
-	else:
-		config_script=sys.argv[1]
-	cfg=__import__(config_script)
-	if cfg.setuid_on_start:
-		os.setuid(cfg.uid)
-	main()