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

This commit is contained in:
drowe67 2021-12-15 11:12:31 +10:30 committed by David Rowe
parent 8a93ec4896
commit 8c9f1c5761
7 changed files with 119 additions and 30 deletions

View file

@ -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('--framesperburst', dest="N_FRAMES_PER_BURST", 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=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)
audio_buffer += data_in[0]
#data_in, ratecv_state = audioop.ratecv(data_in,2,1,AUDIO_SAMPLE_RATE_RX, MODEM_SAMPLE_RATE, ratecv_state)
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:
rx_total_frames = rx_total_frames + 1
rx_frames = rx_frames + 1
if nbytes == bytes_per_frame:
rx_total_frames = rx_total_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:
rx_errors = rx_errors + 1
if rx_bursts == N_BURSTS:
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)

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@ -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('--framesperburst', dest="N_FRAMES_PER_BURST", default=0, type=int)
parser.add_argument('--delay', dest="DELAY_BETWEEN_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=1, 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)
stream_tx.write(audio[0])
#audio = audioop.ratecv(txbuffer,2,1,MODEM_SAMPLE_RATE, AUDIO_SAMPLE_RATE_TX, None)
stream_tx.write(txbuffer)
else:
# print data to terminal for piping the output to other programs

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

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@ -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}

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@ -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}

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@ -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}

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@ -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