Added some features.

2015-09-30 14:06:30 +00:00 · 2015-09-30 14:06:30 +00:00 · 4b3cc10924
parent c9bf26f1ac
commit 4b3cc10924
6 changed files with 199 additions and 57 deletions
--- a/README.md
+++ b/README.md
@ -14,7 +14,7 @@ It has the following features:
 - it works in Google Chrome, Chromium (above version 37) and Mozilla Firefox (above version 28),
 - currently only supports RTL-SDR, but 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`.)
@ -22,6 +22,13 @@ It has the following features:
 - Receivers can now be listed on <a href="http://sdr.hu/">sdr.hu</a>.
 - License for OpenWebRX is now Affero GPL v3. 

+**News (2015-09-01)**
+- The DDC in *csdr* has been hand-optimized for ARM NEON, so it runs 3× faster on the Raspberry Pi than before. 
+- Also we use *ncat* instead of *rtl_mus*, and it is also 3× faster.
+- OpenWebRX now supports URLs like: http://localhost:8073/#freq=145555000,mod=usb
+
+- When upgrading OpenWebRX, please make sure that you upgrade *csdr*, and install the new (optional) dependency *ncat*!
+
 ## Setup

 OpenWebRX currently requires Linux and python 2.7 to run. 
@ -30,6 +37,7 @@ 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). *(It is optional, but highly advised.)*

 After cloning this repository and connecting an RTL-SDR dongle to your computer, you can run the server:

@ -57,14 +65,10 @@ However, if you hold down the shift key, you can drag the center line (BFO) or t

 ## Configuration tips

-If you want to run OpenWebRX on a remote server instead of localhost, do not forget to set *server_hostname* in `config_webrx.py`, or you may get a WebSocket error.
+Now we have a %[Wiki](https://github.com/simonyiszk/openwebrx/wiki) with some how-tos. However, some quick tips:
+
+If you want to run OpenWebRX on a remote server instead of localhost, do not forget to set *server_hostname* in `config_webrx.py`.

 DSP CPU usage can be fine-tuned in `plugins/dsp/csdr/plugin.py`: you can set transition bandwidths higher (thus degrade filter performance by decreasing the length of the kernel, but also decrease CPU usage), and also set `fft_size` lower.

-If you constantly get *audio overrun* errors, you may change `audio_buffer_maximal_length_sec` in `openwebrx.js` from the default 1.7 to 3.
-
 If you want a chat-box to the top of the page, <a href="https://gist.github.com/ha7ilm/15c4c5e4c80cef9b3144">here is a snippet</a> for you to include in `config_webrx.py`.
-
-## Todo
-
-Currently, clients use up a lot of bandwidth. This will be improved later.
--- a/config_webrx.py
+++ b/config_webrx.py
@ -66,6 +66,7 @@ sdrhu_public_listing = False
 dsp_plugin="csdr"
 fft_fps=9
 fft_size=4096
+#samp_rate = 2048000
 samp_rate = 250000

 center_freq = 145525000
@ -103,7 +104,13 @@ format_conversion="csdr convert_u8_f"

 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
+# (if ncat is not available on your system, rtl_mus will be used, thus you will have to set the same port as "my_listening_port" in config_rtl.py as well)
--- a/htdocs/index.wrx
+++ b/htdocs/index.wrx
@ -23,11 +23,13 @@
 	<head>
 		<title>OpenWebRX | Open Source SDR Web App for Everyone!</title>
 		<script type="text/javascript">
-			//Local variables
-			client_id="%[CLIENT_ID]";
-			ws_url="%[WS_URL]";
-			rx_photo_height=%[RX_PHOTO_HEIGHT];
+			//Global variables
+			var client_id="%[CLIENT_ID]";
+			var ws_url="%[WS_URL]";
+			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>
--- a/htdocs/openwebrx.js
+++ b/htdocs/openwebrx.js
@ -959,6 +959,32 @@ 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;

@ -974,6 +1000,7 @@ 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")
 	{
@ -1010,6 +1037,7 @@ function on_ws_recv(evt)
 				{
 					case "setup":
 						waterfall_init();
+						audio_preinit();
 						break;					
 					case "bandwidth":
 						bandwidth=parseInt(param[1]);
@ -1122,7 +1150,7 @@ var audio_initialized=0;

 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;
@ -1167,7 +1195,7 @@ 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; }
 }


@ -1246,6 +1274,8 @@ 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()
 {
@ -1255,8 +1285,8 @@ 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(underrun) text="underrun";
+	if(overrun) { text="overrun"; console.log("audio overrun, "+(++audio_overrun_cnt).toString()); }
+	if(underrun) { text="underrun"; console.log("audio underrun, "+(++audio_underrun_cnt).toString()); }
 	if(overrun||underrun) 
 	{ 
 		audio_buffer_progressbar_update_disabled=true;
@ -1345,13 +1375,26 @@ function webrx_set_param(what, value)
 	ws.send("SET "+what+"="+value.toString());
 }

-function audio_init()
+function parsehash()
 {
-	audio_debug_time_start=(new Date()).getTime();
-	audio_debug_time_last_start=audio_debug_time_start;
+	if(h=window.location.hash) 
+	{
+		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
-	audio_initialized=1; // only tell on_ws_recv() not to call it again
+function audio_preinit()
+{
 	try 
 	{
 		window.AudioContext = window.AudioContext||window.webkitAudioContext;
@ -1362,6 +1405,28 @@ 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);
@ -1379,7 +1444,14 @@ 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)
--- a/openwebrx.py
+++ b/openwebrx.py
@ -47,7 +47,6 @@ import ctypes
 #import rtl_mus
 import rxws
 import uuid
-import config_webrx as cfg
 import signal
 import socket

@ -75,11 +74,17 @@ 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

+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
+	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 +92,10 @@ def main():
 	print "Author contact info:    Andras Retzler, HA7ILM <randras@sdr.hu>"
 	print 

+	no_arguments=len(sys.argv)==1
+	if no_arguments: print "[openwebrx] Configuration script not specified. I will use: \"config_webrx.py\""
+	cfg=__import__("config_webrx" if no_arguments else sys.argv[1])
+
 	#Set signal handler
 	signal.signal(signal.SIGINT, handle_signal) #http://stackoverflow.com/questions/1112343/how-do-i-capture-sigint-in-python

@ -108,13 +117,18 @@ def main():

 	#Start rtl thread
 	if cfg.start_rtl_thread:
-		rtl_thread=threading.Thread(target = lambda:subprocess.Popen(cfg.start_rtl_command, shell=True), args=())
+		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_cmd = python_command+" rtl_mus.py config_rtl"
+	if os.system("ncat --version > /dev/null") != 32512:
+		print "[openwebrx-main] ncat detected, using it instead of rtl_mus:"
+		rtl_mus_cmd = "ncat localhost 8888 | ncat -4l %d -k --send-only --allow 127.0.0.1 " % cfg.iq_server_port
+		print rtl_mus_cmd
+	rtl_mus_thread=threading.Thread(target = lambda:subprocess.Popen(rtl_mus_cmd, 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	
@ -207,10 +221,20 @@ def mutex_watchdog_thread_function():
 			print "[openwebrx-watchdog] Mutex unlock timeout. Locker: \""+str(clients_mutex_locker)+"\" Now unlocking..." 
 			clients_mutex.release()
 		time.sleep(0.5)	
-		
+
+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 spectrum_thread_function():
-	global clients
-	dsp=getattr(plugins.dsp,cfg.dsp_plugin).plugin.dsp_plugin()
+	global clients, spectrum_dsp
+	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)
@ -220,6 +244,7 @@ def spectrum_thread_function():
 	sleep_sec=0.87/cfg.fft_fps
 	print "[openwebrx-spectrum] Spectrum thread initialized successfully."
 	dsp.start()
+	dsp.read(8) #dummy read to skip bufsize & preamble
 	print "[openwebrx-spectrum] Spectrum thread started." 
 	bytes_to_read=int(dsp.get_fft_bytes_to_read())
 	while True:
@ -255,16 +280,17 @@ 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():
-	# if client doesn't open websocket for too long time, we drop it
+def cleanup_clients(end_all=False):
+	# - 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:
-			print "[openwebrx] cleanup_clients :: client timeout to open WebSocket"
+		if end_all or ((not clients[i].ws_started) and (time.time()-clients[i].gen_time)>45):
+			if not end_all: print "[openwebrx] cleanup_clients :: client timeout to open WebSocket"
 			close_client(i, False)
 			correction+=1
 	cmr()
@ -363,13 +389,12 @@ class WebRXHandler(BaseHTTPRequestHandler):

 					# ========= Initialize DSP =========
 					dsp=getattr(plugins.dsp,cfg.dsp_plugin).plugin.dsp_plugin()
-					dsp.set_samp_rate(cfg.samp_rate)
-					dsp.set_demodulator("nfm")
-					dsp.set_offset_freq(0)
-					dsp.set_bpf(-4000,4000)
+					dsp_initialized=False
 					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
 					myclient.dsp=dsp
 					
 					while True:
@ -378,8 +403,9 @@ class WebRXHandler(BaseHTTPRequestHandler):
 							break

 						# ========= send audio =========
-						temp_audio_data=dsp.read(256)
-						rxws.send(self, temp_audio_data, "AUD ")
+						if dsp_initialized:
+							temp_audio_data=dsp.read(256)
+							rxws.send(self, temp_audio_data, "AUD ")

 						# ========= send spectrum =========
 						while not myclient.spectrum_queue.empty():
@ -417,9 +443,17 @@ 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:
@ -464,6 +498,13 @@ class WebRXHandler(BaseHTTPRequestHandler):
 				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:
+					self.send_response(500)
+					self.send_header('Content-type','text/html')
+					self.end_headers()
+					self.wfile.write("<html><body><h1>OpenWebRX Internal Server Error</h1>Please check the server log for details.</body></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 +533,9 @@ 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:
--- a/plugins/dsp/csdr/plugin.py
+++ b/plugins/dsp/csdr/plugin.py
@ -44,13 +44,15 @@ class dsp_plugin:
 		self.demodulator = "nfm"
 		self.name = "csdr"
 		self.format_conversion = "csdr convert_u8_f"
+		self.base_bufsize = 512
+		self.nc_port = 4951
 		try:	
 			subprocess.Popen("nc",stdout=subprocess.PIPE,stderr=subprocess.PIPE).kill()
 		except:
 			print "[openwebrx-plugin:csdr] error: netcat not found, please install netcat!"

 	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="nc -v localhost {nc_port} | csdr setbuf {start_bufsize} | csdr through | "+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 +63,9 @@ 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 +94,10 @@ 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 +159,16 @@ 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()
+		my_env["CSDR_DYNAMIC_BUFSIZE_ON"]="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 +191,16 @@ class dsp_plugin:
 		#	os.killpg(self.process.pid, signal.SIGTERM) 
 		#	
 		#	time.sleep(0.1)
-		try:
-			os.unlink(self.bpf_pipe)
-		except:
-			print "[openwebrx-dsp-plugin:csdr] stop() :: unlink failed: " + self.bpf_pipe
-		try:
-			os.unlink(self.shift_pipe)
-		except:
-			print "[openwebrx-dsp-plugin:csdr] stop() :: unlink failed: " + self.bpf_pipe
+		if self.bpf_pipe:
+			try:
+				os.unlink(self.bpf_pipe)
+			except:
+				print "[openwebrx-dsp-plugin:csdr] stop() :: unlink failed: " + self.bpf_pipe
+		if self.shift_pipe:
+			try:
+				os.unlink(self.shift_pipe)
+			except:
+				print "[openwebrx-dsp-plugin:csdr] stop() :: unlink failed: " + self.bpf_pipe
 		self.running = False

 	def restart(self):