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optimized audio processing

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moved from queue to deque, decode only when not transmitting.
This commit is contained in:
dj2ls 2022-03-08 21:00:43 +01:00
parent a552109265
commit d0f6686899
1 changed files with 52 additions and 79 deletions
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131
tnc/modem.py
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@ -25,8 +25,7 @@ import queue
import codec2
import audio
from collections import deque
MODEM_STATS_NR_MAX = 320
MODEM_STATS_NC_MAX = 51
@ -92,7 +91,7 @@ class RF():
self.modem_received_queue = MODEM_RECEIVED_QUEUE
# init FIFO queue to store modulation out in
self.modoutqueue = queue.Queue()
self.modoutqueue = deque()
# define fft_data buffer
self.fft_data = bytes()
@ -229,54 +228,49 @@ class RF():
Returns:
"""
"""
x = np.frombuffer(data_in48k, dtype=np.int16)
time_sampler_start = time.time()
x = self.resampler.resample48_to_8(x)
time_sampler_end = time.time()
length_x = len(x)
time_buffer_start = time.time()
# avoid buffer overflow by filling only if buffer not full
if not self.datac0_buffer.nbuffer+len(x) > self.datac0_buffer.size:
self.datac0_buffer.push(x)
else:
static.BUFFER_OVERFLOW_COUNTER[0] += 1
# avoid buffer overflow by filling only if buffer not full and selected datachannel mode
if not self.datac1_buffer.nbuffer+len(x) > self.datac1_buffer.size:
if RECEIVE_DATAC1:
self.datac1_buffer.push(x)
else:
static.BUFFER_OVERFLOW_COUNTER[1] += 1
# avoid buffer overflow by filling only if buffer not full and selected datachannel mode
if not self.datac3_buffer.nbuffer+len(x) > self.datac3_buffer.size:
if RECEIVE_DATAC3:
self.datac3_buffer.push(x)
else:
static.BUFFER_OVERFLOW_COUNTER[2] += 1
if self.modoutqueue.empty() or self.mod_out_locked:
data_out48k = bytes(self.AUDIO_FRAMES_PER_BUFFER_TX*2)
# avoid decoding when transmitting to reduce CPU
if not static.TRANSMITTING:
# avoid buffer overflow by filling only if buffer not full
if not self.datac0_buffer.nbuffer+length_x > self.datac0_buffer.size:
self.datac0_buffer.push(x)
else:
static.BUFFER_OVERFLOW_COUNTER[0] += 1
# avoid buffer overflow by filling only if buffer not full and selected datachannel mode
if not self.datac1_buffer.nbuffer+length_x > self.datac1_buffer.size:
if RECEIVE_DATAC1:
self.datac1_buffer.push(x)
else:
static.BUFFER_OVERFLOW_COUNTER[1] += 1
# avoid buffer overflow by filling only if buffer not full and selected datachannel mode
if not self.datac3_buffer.nbuffer+length_x > self.datac3_buffer.size:
if RECEIVE_DATAC3:
self.datac3_buffer.push(x)
else:
static.BUFFER_OVERFLOW_COUNTER[2] += 1
if not self.modoutqueue or self.mod_out_locked:
data_out48k = np.zeros(frame_count, dtype=np.int16)
self.fft_data = bytes(x)
else:
data_out48k = self.modoutqueue.get()
data_out48k = self.modoutqueue.popleft()
self.fft_data = bytes(data_out48k)
time_buffer_end = time.time()
#print(f"SAMPLER {time_sampler_end - time_sampler_start} BUFFER {time_buffer_end - time_buffer_start}")
return (data_out48k, audio.pyaudio.paContinue)
# --------------------------------------------------------------------------------------------------------
def transmit(self, mode, repeats, repeat_delay, frames):
"""
@ -326,11 +320,8 @@ class RF():
for i in range(1,repeats+1):
# write preamble to txbuffer
codec2.api.freedv_rawdatapreambletx(freedv, mod_out_preamble)
#time.sleep(0.05)
#threading.Event().wait(0.05)
txbuffer += bytes(mod_out_preamble)
# create modulaton for n frames in list
for n in range(0,len(frames)):
@ -346,23 +337,16 @@ class RF():
data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer)
codec2.api.freedv_rawdatatx(freedv,mod_out,data) # modulate DATA and save it into mod_out pointer
#time.sleep(0.05)
#threading.Event().wait(0.05)
txbuffer += bytes(mod_out)
# append postamble to txbuffer
codec2.api.freedv_rawdatapostambletx(freedv, mod_out_postamble)
txbuffer += bytes(mod_out_postamble)
#time.sleep(0.05)
#threading.Event().wait(0.05)
# add delay to end of frames
samples_delay = int(self.MODEM_SAMPLE_RATE*(repeat_delay/1000))
mod_out_silence = create_string_buffer(samples_delay*2)
txbuffer += bytes(mod_out_silence)
#time.sleep(0.05)
# resample up to 48k (resampler works on np.int16)
x = np.frombuffer(txbuffer, dtype=np.int16)
@ -370,29 +354,22 @@ class RF():
# explicitly lock our usage of mod_out_queue
self.mod_out_locked = True
# split modualted audio to chunks
#https://newbedev.com/how-to-split-a-byte-string-into-separate-bytes-in-python
txbuffer_48k = bytes(txbuffer_48k)
chunk = [txbuffer_48k[i:i+self.AUDIO_FRAMES_PER_BUFFER_RX*2] for i in range(0, len(txbuffer_48k), self.AUDIO_FRAMES_PER_BUFFER_RX*2)]
# add modulated chunks to fifo buffer
chunk_length = self.AUDIO_FRAMES_PER_BUFFER_TX #4800
chunk = [txbuffer_48k[i:i+chunk_length] for i in range(0, len(txbuffer_48k), chunk_length)]
for c in chunk:
# if data is shorter than the expcected audio frames per buffer we need to append 0
# to prevent the callback from stucking/crashing
if len(c) < self.AUDIO_FRAMES_PER_BUFFER_RX*2:
delta = bytes(self.AUDIO_FRAMES_PER_BUFFER_RX*2 - len(c))
c += delta
structlog.get_logger("structlog").debug("[TNC] mod out shorter than audio buffer", delta=len(delta))
self.modoutqueue.put(c)
if len(c) < chunk_length:
delta = chunk_length - len(c)
delta_zeros = np.zeros(delta, dtype=np.int16)
c = np.append(c, delta_zeros)
structlog.get_logger("structlog").debug("[TNC] mod out shorter than audio buffer", delta=delta)
self.modoutqueue.append(c)
# Release our mod_out_lock so we can use the queue
self.mod_out_locked = False
while not self.modoutqueue.empty():
pass
while self.modoutqueue:
time.sleep(0.01)
static.PTT_STATE = self.hamlib.set_ptt(False)
@ -430,14 +407,14 @@ class RF():
while self.audio_stream.is_active():
threading.Event().wait(0.01)
while self.datac1_buffer.nbuffer >= self.datac1_nin:
# demodulate audio
nbytes_datac1 = codec2.api.freedv_rawdatarx(self.datac1_freedv, self.datac1_bytes_out, self.datac1_buffer.buffer.ctypes)
self.datac1_buffer.pop(self.datac1_nin)
self.datac1_nin = codec2.api.freedv_nin(self.datac1_freedv)
if nbytes_datac1 == self.datac1_bytes_per_frame:
self.modem_received_queue.put([self.datac1_bytes_out, self.datac1_freedv ,self.datac1_bytes_per_frame])
self.get_scatter(self.datac1_freedv)
self.calculate_snr(self.datac1_freedv)
# demodulate audio
nbytes_datac1 = codec2.api.freedv_rawdatarx(self.datac1_freedv, self.datac1_bytes_out, self.datac1_buffer.buffer.ctypes)
self.datac1_buffer.pop(self.datac1_nin)
self.datac1_nin = codec2.api.freedv_nin(self.datac1_freedv)
if nbytes_datac1 == self.datac1_bytes_per_frame:
self.modem_received_queue.put([self.datac1_bytes_out, self.datac1_freedv ,self.datac1_bytes_per_frame])
self.get_scatter(self.datac1_freedv)
self.calculate_snr(self.datac1_freedv)
def audio_datac3(self):
""" """
@ -578,11 +555,7 @@ class RF():
data_in = self.fft_data
# delete fft_buffer
self.fft_data = bytes()
self.fft_data = bytes()
# https://gist.github.com/ZWMiller/53232427efc5088007cab6feee7c6e4c
audio_data = np.fromstring(data_in, np.int16)