add basic support for audio over the network
This commit is contained in:
67
poetry.lock
generated
67
poetry.lock
generated
@@ -859,6 +859,71 @@ files = [
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{file = "multidict-6.2.0.tar.gz", hash = "sha256:0085b0afb2446e57050140240a8595846ed64d1cbd26cef936bfab3192c673b8"},
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]
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[[package]]
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name = "numpy"
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version = "2.2.6"
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description = "Fundamental package for array computing in Python"
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optional = false
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python-versions = ">=3.10"
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groups = ["main"]
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]
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[[package]]
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name = "packaging"
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version = "24.2"
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@@ -1631,4 +1696,4 @@ test = ["pytest", "pytest-asyncio"]
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[metadata]
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lock-version = "2.1"
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python-versions = ">=3.11"
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content-hash = "43fd58df49d1a116d9cacc305ee2dcee08521a6421aa4736916449c49ef42b2e"
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content-hash = "d4fd7c027611a5a3c66d88cc4bff8735f8b867a1eda731ef34aa39c3bf49f289"
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@@ -12,7 +12,8 @@ dependencies = [
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"uvicorn==0.34.0",
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"aiohttp==3.9.3",
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"sounddevice (>=0.5.1,<0.6.0)",
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"aioconsole (>=0.8.1,<0.9.0)"
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"aioconsole (>=0.8.1,<0.9.0)",
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"numpy (>=2.2.6,<3.0.0)"
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]
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[project.optional-dependencies]
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@@ -49,6 +49,7 @@ from bumble.audio import io as audio_io
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from auracast import auracast_config
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from auracast.utils.read_lc3_file import read_lc3_file
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from auracast.network_audio_receiver import NetworkAudioReceiverUncoded
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# modified from bumble
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@@ -339,8 +340,25 @@ class Streamer():
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audio_source = big_config[i].audio_source
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input_format = big_config[i].input_format
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# --- New: network_uncoded mode using NetworkAudioReceiver ---
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if isinstance(audio_source, NetworkAudioReceiverUncoded):
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# Start the UDP receiver coroutine so packets are actually received
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asyncio.create_task(audio_source.receive())
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encoder = lc3.Encoder(
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frame_duration_us=global_config.frame_duration_us,
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sample_rate_hz=global_config.auracast_sampling_rate_hz,
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num_channels=1,
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input_sample_rate_hz=audio_source.samplerate,
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)
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lc3_frame_samples = encoder.get_frame_samples()
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big['pcm_bit_depth'] = 16
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big['lc3_frame_samples'] = lc3_frame_samples
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big['audio_input'] = audio_source
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big['encoder'] = encoder
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big['precoded'] = False
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# precoded lc3 from ram
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if isinstance(big_config[i].audio_source, bytes):
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elif isinstance(big_config[i].audio_source, bytes):
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big['precoded'] = True
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lc3_frames = iter(big_config[i].audio_source)
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@@ -536,16 +554,19 @@ if __name__ == "__main__":
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#config.transport='serial:/dev/serial/by-id/usb-SEGGER_J-Link_001057705357-if02,1000000,rtscts' # transport for nrf54l15dk
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#config.transport='serial:/dev/serial/by-id/usb-ZEPHYR_Zephyr_HCI_UART_sample_95A087EADB030B24-if00,115200,rtscts' #nrf52dongle hci_uart usb cdc
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#config.transport='usb:2fe3:000b' #nrf52dongle hci_usb # TODO: iso packet over usb not supported
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#config.transport= 'auto'
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config.transport='serial:/dev/ttyAMA2,1000000,rtscts' # transport for raspberry pi
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config.transport= 'auto'
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#config.transport='serial:/dev/ttyAMA2,1000000,rtscts' # transport for raspberry pi
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for big in config.bigs: # TODO: encrypted streams are not working
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#big.code = 'ff'*16 # returns hci/HCI_ENCRYPTION_MODE_NOT_ACCEPTABLE_ERROR
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#big.code = '78 e5 dc f1 34 ab 42 bf c1 92 ef dd 3a fd 67 ae'
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big.precode_wav = True
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big.audio_source = big.audio_source.replace('.wav', '_10_16_32.lc3') #lc3 precoded files
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big.audio_source = read_lc3_file(big.audio_source) # load files in advance
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#big.audio_source = big.audio_source.replace('.wav', '_10_16_32.lc3') #lc3 precoded files
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#big.audio_source = read_lc3_file(big.audio_source) # load files in advance
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# --- Network_uncoded mode using NetworkAudioReceiver ---
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big.audio_source = NetworkAudioReceiverUncoded(port=50007, samplerate=16000, channels=1, chunk_size=1024)
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# 16kHz works reliably with 3 streams
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# 24kHz is only working with 2 streams - probably airtime constraint
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67
src/auracast/network_audio_receiver.py
Normal file
67
src/auracast/network_audio_receiver.py
Normal file
@@ -0,0 +1,67 @@
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import asyncio
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import socket
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import logging
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import numpy as np
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from typing import AsyncGenerator
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class NetworkAudioReceiverUncoded:
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"""
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Receives PCM audio over UDP and provides an async generator interface for uncoded PCM frames.
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Combines network receiving and input logic for use with Auracast streamer.
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"""
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def __init__(self, port: int = 50007, samplerate: int = 16000, channels: int = 1, chunk_size: int = 1024):
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self.port = port
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self.samplerate = samplerate
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self.channels = channels
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self.chunk_size = chunk_size
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self.sock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
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self.sock.bind(('0.0.0.0', self.port))
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self.sock.setblocking(False)
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self._running = False
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# Reduce queue size for lower latency (less buffering)
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self._queue = asyncio.Queue(maxsize=2) # Was 20
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async def receive(self):
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self._running = True
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logging.info(f"NetworkAudioReceiver listening on UDP port {self.port}")
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try:
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while self._running:
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try:
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data, _ = await asyncio.get_event_loop().sock_recvfrom(self.sock, self.chunk_size * 2)
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await self._queue.put(data)
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except Exception:
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await asyncio.sleep(0.01)
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finally:
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self.sock.close()
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logging.info("NetworkAudioReceiver stopped.")
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def stop(self):
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self._running = False
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async def open(self):
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# Dummy PCM format object
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class PCMFormat:
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channels = self.channels
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sample_type = 'int16'
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sample_rate = self.samplerate
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return PCMFormat()
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def rewind(self):
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pass # Not supported for live network input
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async def frames(self, samples_per_frame: int) -> AsyncGenerator[np.ndarray, None]:
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bytes_per_frame = samples_per_frame * 2 * self.channels # 2 bytes for int16
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buf = bytearray()
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while True:
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data = await self._queue.get()
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||||
# Optional: log queue size for latency debugging
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||||
# logging.debug(f'NetworkAudioReceiver queue size: {self._queue.qsize()}')
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if data is None:
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break
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buf.extend(data)
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while len(buf) >= bytes_per_frame:
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frame = np.frombuffer(buf[:bytes_per_frame], dtype=np.int16).reshape(-1, self.channels)
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# Optional: log when a frame is yielded
|
||||
# logging.debug(f'Yielding frame of shape {frame.shape}')
|
||||
yield frame
|
||||
buf = buf[bytes_per_frame:]
|
||||
Reference in New Issue
Block a user