fine grained tests at 48000 Hz, currently won't work as resampler commented out of test_tx.py and test_rx.py

test/001_highsnr_stdio_audio/test_rx.py

@@ -22,8 +22,8 @@ import codec2
 
 #--------------------------------------------GET PARAMETER INPUTS  
 parser = argparse.ArgumentParser(description='Simons TEST TNC')
-parser.add_argument('--bursts', dest="N_BURSTS", default=0, type=int)
+parser.add_argument('--bursts', dest="N_BURSTS", default=1, type=int)
-parser.add_argument('--framesperburst', dest="N_FRAMES_PER_BURST", default=0, type=int)
+parser.add_argument('--framesperburst', dest="N_FRAMES_PER_BURST", default=1, type=int)
 parser.add_argument('--mode', dest="FREEDV_MODE", type=str, choices=['datac0', 'datac1', 'datac3'])
 parser.add_argument('--audiodev', dest="AUDIO_INPUT_DEVICE", default=-1, type=int,
                     help="audio device number to use, use -2 to automatically select a loopback device")  
@@ -49,7 +49,7 @@ TIMEOUT = args.TIMEOUT
 # AUDIO PARAMETERS
 AUDIO_FRAMES_PER_BUFFER = 2048 # seems to be best if >=1024
 MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
-AUDIO_SAMPLE_RATE_RX = 48000
+AUDIO_SAMPLE_RATE_RX = 8000
 assert (AUDIO_SAMPLE_RATE_RX % MODEM_SAMPLE_RATE) == 0
 
 # check if we want to use an audio device then do an pyaudio init
@@ -104,13 +104,14 @@ receive = True
 audio_buffer = bytes()
 nbytes = 0
 BUF_SIZE = 160
+ratecv_state = None
 while receive and time.time() < timeout:
 
     if AUDIO_INPUT_DEVICE != -1: 
         
         data_in = stream_rx.read(AUDIO_FRAMES_PER_BUFFER,  exception_on_overflow = False)  
-        data_in = audioop.ratecv(data_in,2,1,AUDIO_SAMPLE_RATE_RX, MODEM_SAMPLE_RATE, None) 
+        #data_in, ratecv_state = audioop.ratecv(data_in,2,1,AUDIO_SAMPLE_RATE_RX, MODEM_SAMPLE_RATE, ratecv_state) 
-        audio_buffer += data_in[0]
+        audio_buffer += data_in
 
     else:
 
@@ -118,32 +119,30 @@ while receive and time.time() < timeout:
 
     nin = codec2.api.freedv_nin(freedv) * 2
     
-    rx_status = codec2.api.freedv_get_rx_status(freedv)       
-    if DEBUGGING_MODE:        
-        state = codec2.api.rx_sync_flags_to_text[rx_status]
-        print(f"NIN: {nin} STATE: {state}  BUFFER: {len(audio_buffer)}", file=sys.stderr)
-    
     #decode as soon as we filled the audio buffer    
     if len(audio_buffer) >= (nin):
         nbytes = codec2.api.freedv_rawdatarx(freedv, bytes_out, audio_buffer[:nin]) # demodulate audio
         audio_buffer = audio_buffer[nin:]
 
+        rx_status = codec2.api.freedv_get_rx_status(freedv)       
+        if rx_status & codec2.api.FREEDV_RX_BIT_ERRORS:
+            rx_errors = rx_errors + 1
+        if DEBUGGING_MODE:        
+          state = codec2.api.rx_sync_flags_to_text[rx_status]
+          print(f"NIN: {nin} RX_STATUS: {state}  BUFFER: {len(audio_buffer)}", file=sys.stderr)
 
-    total_n_bytes = total_n_bytes + nbytes
+        total_n_bytes = total_n_bytes + nbytes
             
-    if nbytes == bytes_per_frame:
+        if nbytes == bytes_per_frame:
-        rx_total_frames = rx_total_frames + 1
+            rx_total_frames = rx_total_frames + 1
-        rx_frames = rx_frames + 1
+            rx_frames = rx_frames + 1
 
         if rx_frames == N_FRAMES_PER_BURST:
             rx_frames = 0
             rx_bursts = rx_bursts + 1
                           
-    if rx_status & codec2.api.FREEDV_RX_BIT_ERRORS:
+        if rx_bursts == N_BURSTS:
-        rx_errors = rx_errors + 1
+            receive = False   
-                
-    if rx_bursts == N_BURSTS:
-        receive = False   
                    
 
 print(f"RECEIVED BURSTS: {rx_bursts} RECEIVED FRAMES: {rx_total_frames} RX_ERRORS: {rx_errors}", file=sys.stderr)

test/001_highsnr_stdio_audio/test_tx.py

@@ -17,9 +17,10 @@ import codec2
 
 # GET PARAMETER INPUTS  
 parser = argparse.ArgumentParser(description='Simons TEST TNC')
-parser.add_argument('--bursts', dest="N_BURSTS", default=0, type=int)
+parser.add_argument('--bursts', dest="N_BURSTS", default=1, type=int)
-parser.add_argument('--framesperburst', dest="N_FRAMES_PER_BURST", default=0, type=int)
+parser.add_argument('--framesperburst', dest="N_FRAMES_PER_BURST", default=1, type=int)
-parser.add_argument('--delay', dest="DELAY_BETWEEN_BURSTS", default=0, type=int)
+parser.add_argument('--delay', dest="DELAY_BETWEEN_BURSTS", default=500, type=int,
+                    help="delay between bursts in ms")
 parser.add_argument('--mode', dest="FREEDV_MODE", type=str, choices=['datac0', 'datac1', 'datac3'])
 parser.add_argument('--audiodev', dest="AUDIO_OUTPUT_DEVICE", default=-1, type=int,
                     help="audio output device number to use, use -2 to automatically select a loopback device") 
@@ -44,7 +45,7 @@ MODE = codec2.FREEDV_MODE[args.FREEDV_MODE].value
 # AUDIO PARAMETERS
 AUDIO_FRAMES_PER_BUFFER = 2048 
 MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
-AUDIO_SAMPLE_RATE_TX = 48000
+AUDIO_SAMPLE_RATE_TX = 8000
 assert (AUDIO_SAMPLE_RATE_TX % MODEM_SAMPLE_RATE) == 0
 
 # check if we want to use an audio device then do an pyaudio init
@@ -136,7 +137,7 @@ for i in range(1,N_BURSTS+1):
     txbuffer += bytes(mod_out_postamble)
 
     # append a delay between bursts as audio silence
-    samples_delay = int(MODEM_SAMPLE_RATE*DELAY_BETWEEN_BURSTS)
+    samples_delay = int(MODEM_SAMPLE_RATE*DELAY_BETWEEN_BURSTS*2)
     mod_out_silence = create_string_buffer(samples_delay)    
     txbuffer += bytes(mod_out_silence)
     print(f"samples_delay: {samples_delay} DELAY_BETWEEN_BURSTS: {DELAY_BETWEEN_BURSTS}", file=sys.stderr)
@@ -145,8 +146,8 @@ for i in range(1,N_BURSTS+1):
     if AUDIO_OUTPUT_DEVICE != -1: 
         
         # sample rate conversion from 8000Hz to 48000Hz
-        audio = audioop.ratecv(txbuffer,2,1,MODEM_SAMPLE_RATE, AUDIO_SAMPLE_RATE_TX, None)                                           
+        #audio = audioop.ratecv(txbuffer,2,1,MODEM_SAMPLE_RATE, AUDIO_SAMPLE_RATE_TX, None)                                           
-        stream_tx.write(audio[0])
+        stream_tx.write(txbuffer)
 
     else:
         # print data to terminal for piping the output to other programs

test/001_highsnr_stdio_audio/test_virtual1.sh

@@ -1,6 +1,6 @@
 #!/bin/bash -x
-# Run a test using the virtual sound cards, but sound I/O performed by aplay/arecord, and
+# Run a test using the virtual sound cards, sound I/O performed by aplay/arecord at
-# we pipe to Python utilities
+# Fs=8000 Hz, and we pipe to Python utilities
 
 function check_alsa_loopback {
     lsmod | grep snd_aloop >> /dev/null

test/001_highsnr_stdio_audio/test_virtual1a.sh

@@ -0,0 +1,31 @@
+#!/bin/bash -x
+# Run a test using the virtual sound cards, tx sound I/O performed by aplay
+# and arecord at Fs=48000Hz, we pipe to Python utilities
+
+function check_alsa_loopback {
+    lsmod | grep snd_aloop >> /dev/null
+    if [ $? -eq 1 ]; then
+      echo "ALSA loopback device not present.  Please install with:"
+      echo
+      echo "  sudo modprobe snd-aloop index=1,2 enable=1,1 pcm_substreams=1,1 id=CHAT1,CHAT2"
+      exit 1
+    fi  
+}
+
+check_alsa_loopback
+
+RX_LOG=$(mktemp)
+MAX_RUN_TIME=2600
+
+# make sure all child processes are killed when we exit
+trap 'jobs -p | xargs -r kill' EXIT
+
+arecord -r 48000 --device="plughw:CARD=CHAT1,DEV=0" -f S16_LE -d $MAX_RUN_TIME | \
+    sox -t .s16 -r 48000 -c 1 - -t .s16 -r 8000 -c 1 - | \
+    python3 test_rx.py --mode datac0 --frames 2 --bursts 5 --debug &
+rx_pid=$!
+sleep 1
+python3 test_tx.py --mode datac0 --frames 2 --bursts 5 --delay 500 | \
+    sox -t .s16 -r 8000 -c 1 - -t .s16 -r 48000 -c 1 - | \
+    aplay -r 48000 --device="plughw:CARD=CHAT1,DEV=1" -f S16_LE
+wait ${rx_pid}

test/001_highsnr_stdio_audio/test_virtual1b.sh

@@ -0,0 +1,29 @@
+#!/bin/bash -x
+# Run a test using the virtual sound cards, tx sound I/O performed by Python,
+# rx using arecord, Fs=48000Hz
+
+function check_alsa_loopback {
+    lsmod | grep snd_aloop >> /dev/null
+    if [ $? -eq 1 ]; then
+      echo "ALSA loopback device not present.  Please install with:"
+      echo
+      echo "  sudo modprobe snd-aloop index=1,2 enable=1,1 pcm_substreams=1,1 id=CHAT1,CHAT2"
+      exit 1
+    fi  
+}
+
+check_alsa_loopback
+
+RX_LOG=$(mktemp)
+MAX_RUN_TIME=2600
+
+# make sure all child processes are killed when we exit
+trap 'jobs -p | xargs -r kill' EXIT
+
+arecord -r 48000 --device="plughw:CARD=CHAT1,DEV=0" -f S16_LE -d $MAX_RUN_TIME | \
+    sox -t .s16 -r 48000 -c 1 - -t .s16 -r 8000 -c 1 - | \
+    python3 test_rx.py --mode datac0 --frames 2 --bursts 5 --debug &
+rx_pid=$!
+sleep 1
+python3 test_tx.py --mode datac0 --frames 2 --bursts 5 --delay 500 --audiodev -2
+wait ${rx_pid}

test/001_highsnr_stdio_audio/test_virtual1c.sh

@@ -0,0 +1,29 @@
+#!/bin/bash -x
+# Run a test using the virtual sound cards, tx sound I/O performed by aplay,
+# rx sound I/O by Python, Fs=48000Hz.
+
+function check_alsa_loopback {
+    lsmod | grep snd_aloop >> /dev/null
+    if [ $? -eq 1 ]; then
+      echo "ALSA loopback device not present.  Please install with:"
+      echo
+      echo "  sudo modprobe snd-aloop index=1,2 enable=1,1 pcm_substreams=1,1 id=CHAT1,CHAT2"
+      exit 1
+    fi  
+}
+
+check_alsa_loopback
+
+RX_LOG=$(mktemp)
+MAX_RUN_TIME=2600
+
+# make sure all child processes are killed when we exit
+trap 'jobs -p | xargs -r kill' EXIT
+
+python3 test_rx.py --mode datac0 --frames 2 --bursts 5 --debug --audiodev -2 &
+rx_pid=$!
+sleep 1
+python3 test_tx.py --mode datac0 --frames 2 --bursts 5 --delay 500 | \
+    sox -t .s16 -r 8000 -c 1 - -t .s16 -r 48000 -c 1 - | \
+    aplay -r 48000 --device="plughw:CARD=CHAT1,DEV=1" -f S16_LE
+wait ${rx_pid}

test/001_highsnr_stdio_audio/test_virtual2.sh

@@ -1,5 +1,5 @@
 #!/bin/bash -x
-# Run a test using the virtual sound cards
+# Run a test using the virtual sound cards, Python audio I/O
 
 function check_alsa_loopback {
     lsmod | grep snd_aloop >> /dev/null