mirror of
https://github.com/DJ2LS/FreeDATA
synced 2024-05-14 08:04:33 +00:00
moved code to class
less globals and we also need this later
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dj2ls
parent
84bf1970dd
commit
1c4bb7bfbc
1 changed files with 133 additions and 136 deletions
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@ -38,149 +38,146 @@ if args.LIST:
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for dev in range(0,p.get_device_count()):
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print("audiodev: ", dev, p.get_device_info_by_index(dev)["name"])
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quit()
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N_BURSTS = args.N_BURSTS
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N_FRAMES_PER_BURST = args.N_FRAMES_PER_BURST
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AUDIO_INPUT_DEVICE = args.AUDIO_INPUT_DEVICE
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MODE = codec2.FREEDV_MODE[args.FREEDV_MODE].value
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DEBUGGING_MODE = args.DEBUGGING_MODE
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TIMEOUT = args.TIMEOUT
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# AUDIO PARAMETERS
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AUDIO_FRAMES_PER_BUFFER = 2400*2 # <- consider increasing if you get nread_exceptions > 0
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MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
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AUDIO_SAMPLE_RATE_RX = 48000
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# make sure our resampler will work
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assert (AUDIO_SAMPLE_RATE_RX / MODEM_SAMPLE_RATE) == codec2.api.FDMDV_OS_48
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# DATA CHANNEL INITIALISATION
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# open codec2 instance
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freedv = cast(codec2.api.freedv_open(MODE), c_void_p)
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class Test():
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def __init__(self):
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self.N_BURSTS = args.N_BURSTS
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self.N_FRAMES_PER_BURST = args.N_FRAMES_PER_BURST
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self.AUDIO_INPUT_DEVICE = args.AUDIO_INPUT_DEVICE
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self.MODE = codec2.FREEDV_MODE[args.FREEDV_MODE].value
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self.DEBUGGING_MODE = args.DEBUGGING_MODE
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self.TIMEOUT = args.TIMEOUT
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# get number of bytes per frame for mode
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bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(freedv)/8)
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payload_bytes_per_frame = bytes_per_frame -2
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# AUDIO PARAMETERS
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self.AUDIO_FRAMES_PER_BUFFER = 2400*2 # <- consider increasing if you get nread_exceptions > 0
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self.MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
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self.AUDIO_SAMPLE_RATE_RX = 48000
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n_max_modem_samples = codec2.api.freedv_get_n_max_modem_samples(freedv)
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bytes_out = create_string_buffer(bytes_per_frame * 2)
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# make sure our resampler will work
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assert (self.AUDIO_SAMPLE_RATE_RX / self.MODEM_SAMPLE_RATE) == codec2.api.FDMDV_OS_48
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codec2.api.freedv_set_frames_per_burst(freedv,N_FRAMES_PER_BURST)
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total_n_bytes = 0
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rx_total_frames = 0
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rx_frames = 0
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rx_bursts = 0
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rx_errors = 0
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nread_exceptions = 0
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timeout = time.time() + TIMEOUT
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receive = True
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audio_buffer = codec2.audio_buffer(AUDIO_FRAMES_PER_BUFFER*2)
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resampler = codec2.resampler()
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# Copy received 48 kHz to a file. Listen to this file with:
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# aplay -r 48000 -f S16_LE rx48_callback.raw
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# Corruption of this file is a good way to detect audio card issues
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frx = open("rx48_callback.raw", mode='wb')
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# ------------------------------------------------ PYAUDIO CALLBACK
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def callback(data_in48k, frame_count, time_info, status):
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global total_n_bytes
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global rx_total_frames
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global rx_frames
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global rx_bursts
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global receive
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x = np.frombuffer(data_in48k, dtype=np.int16)
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x.tofile(frx)
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x = resampler.resample48_to_8(x)
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audio_buffer.push(x)
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# when we have enough samples call FreeDV Rx
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nin = codec2.api.freedv_nin(freedv)
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while audio_buffer.nbuffer >= nin:
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# demodulate audio
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nbytes = codec2.api.freedv_rawdatarx(freedv, bytes_out, audio_buffer.buffer.ctypes)
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audio_buffer.pop(nin)
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# call me on every loop!
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nin = codec2.api.freedv_nin(freedv)
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rx_status = codec2.api.freedv_get_rx_status(freedv)
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if rx_status & codec2.api.FREEDV_RX_BIT_ERRORS:
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rx_errors = rx_errors + 1
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if DEBUGGING_MODE:
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rx_status = codec2.api.rx_sync_flags_to_text[rx_status]
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print("nin: %5d rx_status: %4s naudio_buffer: %4d" % \
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(nin,rx_status,audio_buffer.nbuffer), file=sys.stderr)
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if nbytes:
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total_n_bytes = total_n_bytes + nbytes
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# check if we want to use an audio device then do an pyaudio init
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if self.AUDIO_INPUT_DEVICE != -1:
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self.p = pyaudio.PyAudio()
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# auto search for loopback devices
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if self.AUDIO_INPUT_DEVICE == -2:
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loopback_list = []
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for dev in range(0,self.p.get_device_count()):
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if 'Loopback: PCM' in self.p.get_device_info_by_index(dev)["name"]:
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loopback_list.append(dev)
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if len(loopback_list) >= 2:
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self.AUDIO_INPUT_DEVICE = loopback_list[0] #0 = RX 1 = TX
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print(f"loopback_list rx: {loopback_list}", file=sys.stderr)
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else:
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quit()
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print(f"AUDIO INPUT DEVICE: {self.AUDIO_INPUT_DEVICE} DEVICE: {self.p.get_device_info_by_index(self.AUDIO_INPUT_DEVICE)['name']} \
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AUDIO SAMPLE RATE: {self.AUDIO_SAMPLE_RATE_RX}", file=sys.stderr)
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if nbytes == bytes_per_frame:
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rx_total_frames = rx_total_frames + 1
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rx_frames = rx_frames + 1
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self.stream_rx = self.p.open(format=pyaudio.paInt16,
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channels=1,
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rate=self.AUDIO_SAMPLE_RATE_RX,
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frames_per_buffer=self.AUDIO_FRAMES_PER_BUFFER,
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input=True,
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output=False,
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input_device_index=self.AUDIO_INPUT_DEVICE,
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stream_callback=self.callback
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)
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if rx_frames == N_FRAMES_PER_BURST:
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rx_frames = 0
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rx_bursts = rx_bursts + 1
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if rx_bursts == N_BURSTS:
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receive = False
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return (None, pyaudio.paContinue)
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# check if we want to use an audio device then do an pyaudio init
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if AUDIO_INPUT_DEVICE != -1:
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p = pyaudio.PyAudio()
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# auto search for loopback devices
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if AUDIO_INPUT_DEVICE == -2:
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loopback_list = []
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for dev in range(0,p.get_device_count()):
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if 'Loopback: PCM' in p.get_device_info_by_index(dev)["name"]:
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loopback_list.append(dev)
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if len(loopback_list) >= 2:
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AUDIO_INPUT_DEVICE = loopback_list[0] #0 = RX 1 = TX
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print(f"loopback_list rx: {loopback_list}", file=sys.stderr)
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else:
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quit()
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print(f"AUDIO INPUT DEVICE: {AUDIO_INPUT_DEVICE} DEVICE: {p.get_device_info_by_index(AUDIO_INPUT_DEVICE)['name']} \
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AUDIO SAMPLE RATE: {AUDIO_SAMPLE_RATE_RX}", file=sys.stderr)
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stream_rx = p.open(format=pyaudio.paInt16,
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channels=1,
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rate=AUDIO_SAMPLE_RATE_RX,
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frames_per_buffer=AUDIO_FRAMES_PER_BUFFER,
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input=True,
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output=False,
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input_device_index=AUDIO_INPUT_DEVICE,
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stream_callback=callback
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)
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try:
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print(f"starting pyaudio callback", file=sys.stderr)
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stream_rx.start_stream()
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except Exception as e:
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print(f"pyAudio error: {e}", file=sys.stderr)
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# ----------------------------------------------------------------
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while receive and time.time() < timeout:
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time.sleep(1)
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if time.time() >= timeout:
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print("TIMEOUT REACHED")
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# open codec2 instance
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self.freedv = cast(codec2.api.freedv_open(self.MODE), c_void_p)
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if nread_exceptions:
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print("nread_exceptions %d - receive audio lost! Consider increasing Pyaudio frames_per_buffer..." % \
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nread_exceptions, file=sys.stderr)
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print(f"RECEIVED BURSTS: {rx_bursts} RECEIVED FRAMES: {rx_total_frames} RX_ERRORS: {rx_errors}", file=sys.stderr)
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frx.close()
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# get number of bytes per frame for mode
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self.bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(self.freedv)/8)
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self.bytes_out = create_string_buffer(self.bytes_per_frame * 2)
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codec2.api.freedv_set_frames_per_burst(self.freedv,self.N_FRAMES_PER_BURST)
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self.total_n_bytes = 0
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self.rx_total_frames = 0
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self.rx_frames = 0
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self.rx_bursts = 0
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self.rx_errors = 0
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self.nread_exceptions = 0
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self.timeout = time.time() + self.TIMEOUT
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self.receive = True
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self.audio_buffer = codec2.audio_buffer(self.AUDIO_FRAMES_PER_BUFFER*2)
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self.resampler = codec2.resampler()
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# Copy received 48 kHz to a file. Listen to this file with:
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# aplay -r 48000 -f S16_LE rx48_callback.raw
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# Corruption of this file is a good way to detect audio card issues
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self.frx = open("rx48_callback.raw", mode='wb')
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def callback(self, data_in48k, frame_count, time_info, status):
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x = np.frombuffer(data_in48k, dtype=np.int16)
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x.tofile(self.frx)
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x = self.resampler.resample48_to_8(x)
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self.audio_buffer.push(x)
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# cloese pyaudio instance
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stream_rx.close()
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p.terminate()
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# when we have enough samples call FreeDV Rx
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nin = codec2.api.freedv_nin(self.freedv)
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while self.audio_buffer.nbuffer >= nin:
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# demodulate audio
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nbytes = codec2.api.freedv_rawdatarx(self.freedv, self.bytes_out, self.audio_buffer.buffer.ctypes)
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self.audio_buffer.pop(nin)
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# call me on every loop!
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nin = codec2.api.freedv_nin(self.freedv)
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rx_status = codec2.api.freedv_get_rx_status(self.freedv)
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if rx_status & codec2.api.FREEDV_RX_BIT_ERRORS:
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self.rx_errors = self.rx_errors + 1
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if self.DEBUGGING_MODE:
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rx_status = codec2.api.rx_sync_flags_to_text[rx_status]
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print("nin: %5d rx_status: %4s naudio_buffer: %4d" % \
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(nin,rx_status,self.audio_buffer.nbuffer), file=sys.stderr)
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if nbytes:
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self.total_n_bytes = self.total_n_bytes + nbytes
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if nbytes == self.bytes_per_frame:
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self.rx_total_frames = self.rx_total_frames + 1
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self.rx_frames = self.rx_frames + 1
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if self.rx_frames == self.N_FRAMES_PER_BURST:
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self.rx_frames = 0
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self.rx_bursts = self.rx_bursts + 1
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if self.rx_bursts == self.N_BURSTS:
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self.receive = False
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return (None, pyaudio.paContinue)
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def run_audio(self):
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try:
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print(f"starting pyaudio callback", file=sys.stderr)
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self.stream_rx.start_stream()
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except Exception as e:
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print(f"pyAudio error: {e}", file=sys.stderr)
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while self.receive and time.time() < self.timeout:
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time.sleep(1)
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if time.time() >= self.timeout:
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print("TIMEOUT REACHED")
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if self.nread_exceptions:
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print("nread_exceptions %d - receive audio lost! Consider increasing Pyaudio frames_per_buffer..." % \
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self.nread_exceptions, file=sys.stderr)
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print(f"RECEIVED BURSTS: {self.rx_bursts} RECEIVED FRAMES: {self.rx_total_frames} RX_ERRORS: {self.rx_errors}", file=sys.stderr)
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self.frx.close()
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# cloese pyaudio instance
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self.stream_rx.close()
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self.p.terminate()
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test = Test()
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test.run_audio()
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