696 lines
29 KiB
Python
696 lines
29 KiB
Python
# Copyright 2024 Google LLC
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# https://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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# -----------------------------------------------------------------------------
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# Imports
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# -----------------------------------------------------------------------------
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from __future__ import annotations
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import pprint
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import asyncio
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import contextlib
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import logging
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import wave
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import itertools
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import struct
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from typing import cast, Any, AsyncGenerator, Coroutine, List
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import itertools
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import glob
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try:
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import lc3 # type: ignore # pylint: disable=E0401
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except ImportError as e:
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raise ImportError("Try `python -m pip install \".[lc3]\"`.") from e
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from bumble.colors import color
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from bumble import company_ids
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from bumble import core
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from bumble import gatt
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from bumble import hci
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from bumble.profiles import bap
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from bumble.profiles import le_audio
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from bumble.profiles import pbp
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from bumble.profiles import bass
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import bumble.device
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import bumble.transport
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import bumble.utils
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import numpy as np # for audio down-mix
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from bumble.device import Host, BIGInfoAdvertisement, AdvertisingChannelMap
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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.utils.network_audio_receiver import NetworkAudioReceiverUncoded
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from auracast.utils.webrtc_audio_input import WebRTCAudioInput
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# Instantiate WebRTC audio input for streaming (can be used per-BIG or globally)
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# modified from bumble
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class ModWaveAudioInput(audio_io.ThreadedAudioInput):
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"""Audio input that reads PCM samples from a .wav file."""
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def __init__(self, filename: str) -> None:
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super().__init__()
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self._filename = filename
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self._wav: wave.Wave_read | None = None
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self._bytes_read = 0
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self.rewind=True
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def _open(self) -> audio_io.PcmFormat:
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self._wav = wave.open(self._filename, 'rb')
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if self._wav.getsampwidth() != 2:
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raise ValueError('sample width not supported')
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return audio_io.PcmFormat(
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audio_io.PcmFormat.Endianness.LITTLE,
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audio_io.PcmFormat.SampleType.INT16,
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self._wav.getframerate(),
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self._wav.getnchannels(),
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)
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def _read(self, frame_size: int) -> bytes:
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if not self._wav:
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return b''
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pcm_samples = self._wav.readframes(frame_size)
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if not pcm_samples and self._bytes_read:
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if not self.rewind:
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return None
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# Loop around.
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self._wav.rewind()
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self._bytes_read = 0
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pcm_samples = self._wav.readframes(frame_size)
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self._bytes_read += len(pcm_samples)
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return pcm_samples
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def _close(self) -> None:
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if self._wav:
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self._wav.close()
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audio_io.WaveAudioInput = ModWaveAudioInput
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# -----------------------------------------------------------------------------
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# Logging
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# -----------------------------------------------------------------------------
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logger = logging.getLogger(__name__)
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@contextlib.asynccontextmanager
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async def create_device(config: auracast_config.AuracastGlobalConfig) -> AsyncGenerator[bumble.device.Device, Any]:
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async with await bumble.transport.open_transport(config.transport) as (
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hci_source,
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hci_sink,
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):
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device_config = bumble.device.DeviceConfiguration(
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name=config.device_name,
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address= hci.Address(config.auracast_device_address),
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keystore='JsonKeyStore',
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#le_simultaneous_enabled=True #TODO: What is this doing ?
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)
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device = bumble.device.Device.from_config_with_hci(
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device_config,
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hci_source,
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hci_sink,
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)
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await device.power_on()
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yield device
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def run_async(async_command: Coroutine) -> None:
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try:
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asyncio.run(async_command)
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except core.ProtocolError as error:
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if error.error_namespace == 'att' and error.error_code in list(
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bass.ApplicationError
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):
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message = bass.ApplicationError(error.error_code).name
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else:
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message = str(error)
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print(
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color('!!! An error occurred while executing the command:', 'red'), message
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)
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async def init_broadcast(
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device,
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global_config : auracast_config.AuracastGlobalConfig,
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big_config: List[auracast_config.AuracastBigConfig]
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) -> dict:
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bap_sampling_freq = getattr(bap.SamplingFrequency, f"FREQ_{global_config.auracast_sampling_rate_hz}")
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bigs = {}
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for i, conf in enumerate(big_config):
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bigs[f'big{i}'] = {}
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# Config advertising set
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bigs[f'big{i}']['basic_audio_announcement'] = bap.BasicAudioAnnouncement(
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presentation_delay=global_config.presentation_delay_us,
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subgroups=[
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bap.BasicAudioAnnouncement.Subgroup(
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codec_id=hci.CodingFormat(codec_id=hci.CodecID.LC3),
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codec_specific_configuration=bap.CodecSpecificConfiguration(
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sampling_frequency=bap_sampling_freq,
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frame_duration=bap.FrameDuration.DURATION_10000_US,
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octets_per_codec_frame=global_config.octets_per_frame,
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),
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metadata=le_audio.Metadata(
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[
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le_audio.Metadata.Entry(
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tag=le_audio.Metadata.Tag.LANGUAGE, data=conf.language.encode()
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),
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le_audio.Metadata.Entry(
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tag=le_audio.Metadata.Tag.PROGRAM_INFO, data=conf.program_info.encode()
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),
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le_audio.Metadata.Entry(
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tag=le_audio.Metadata.Tag.BROADCAST_NAME, data=conf.name.encode()
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),
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]
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),
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bis=[
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bap.BasicAudioAnnouncement.BIS(
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index=1,
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codec_specific_configuration=bap.CodecSpecificConfiguration(
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audio_channel_allocation=bap.AudioLocation.FRONT_LEFT
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),
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),
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],
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)
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],
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)
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logger.info('Setup Advertising')
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advertising_manufacturer_data = (
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b''
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if global_config.manufacturer_data == (None, None)
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else bytes(
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core.AdvertisingData(
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[
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(
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core.AdvertisingData.MANUFACTURER_SPECIFIC_DATA,
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struct.pack('<H', global_config.manufacturer_data[0])
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+ global_config.manufacturer_data[1],
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)
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]
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)
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)
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)
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bigs[f'big{i}']['broadcast_audio_announcement'] = bap.BroadcastAudioAnnouncement(conf.id)
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advertising_set = await device.create_advertising_set(
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random_address=hci.Address(conf.random_address),
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advertising_parameters=bumble.device.AdvertisingParameters(
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advertising_event_properties=bumble.device.AdvertisingEventProperties(
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is_connectable=False
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),
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primary_advertising_interval_min=100,
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primary_advertising_interval_max=200,
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advertising_sid=i,
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primary_advertising_phy=hci.Phy.LE_1M, # 2m phy config throws error - because for primary advertising channels, 1mbit is only supported
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secondary_advertising_phy=hci.Phy.LE_2M, # this is the secondary advertising beeing send on non advertising channels (extendend advertising)
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#advertising_tx_power= # tx power in dbm (max 20)
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#secondary_advertising_max_skip=10,
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),
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advertising_data=(
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bigs[f'big{i}']['broadcast_audio_announcement'].get_advertising_data()
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+ bytes(
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core.AdvertisingData(
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[(core.AdvertisingData.BROADCAST_NAME, conf.name.encode())]
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)
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)
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+ advertising_manufacturer_data
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),
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periodic_advertising_parameters=bumble.device.PeriodicAdvertisingParameters(
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periodic_advertising_interval_min=80,
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periodic_advertising_interval_max=160,
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),
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periodic_advertising_data=bigs[f'big{i}']['basic_audio_announcement'].get_advertising_data(),
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auto_restart=True,
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auto_start=True,
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)
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bigs[f'big{i}']['advertising_set'] = advertising_set
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logging.info('Start Periodic Advertising')
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await advertising_set.start_periodic()
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logging.info('Setup BIG')
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if global_config.qos_config.iso_int_multiple_10ms == 1:
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frame_enable = 0
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else:
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frame_enable = 1
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big = await device.create_big(
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bigs[f'big{i}']['advertising_set'],
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parameters=bumble.device.BigParameters(
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num_bis=1,
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sdu_interval=global_config.qos_config.iso_int_multiple_10ms*10000, # Is the same as iso interval
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max_sdu=global_config.octets_per_frame,
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max_transport_latency=global_config.qos_config.max_transport_latency_ms,
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rtn=global_config.qos_config.number_of_retransmissions,
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broadcast_code=(
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bytes.fromhex(conf.code) if conf.code else None
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),
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framing=frame_enable # needed if iso interval is not frame interval of codedc
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),
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)
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bigs[f'big{i}']['big'] = big
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for bis_link in big.bis_links:
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await bis_link.setup_data_path(
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direction=bis_link.Direction.HOST_TO_CONTROLLER
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)
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iso_queue = bumble.device.IsoPacketStream(big.bis_links[0], conf.iso_que_len)
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logging.info('Setup ISO Data Path')
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bigs[f'big{i}']['iso_queue'] = iso_queue
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logging.debug(f'big{i} parameters are:')
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logging.debug('%s', pprint.pformat(vars(big)))
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logging.debug(f'Finished setup of big{i}.')
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await asyncio.sleep(i+1) # Wait for advertising to set up
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def on_flow():
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data_packet_queue = iso_queue.data_packet_queue
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print(
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f'\rPACKETS: pending={data_packet_queue.pending}, '
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f'queued={data_packet_queue.queued}, '
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f'completed={data_packet_queue.completed}',
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end='',
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)
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if global_config.debug:
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bigs[f'big{0}']['iso_queue'].data_packet_queue.on('flow', on_flow)
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return bigs
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class Streamer():
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"""
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Streamer class that supports multiple input formats. See bumble for streaming from wav or device
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Added functionallity on top of bumble:
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- loop parameter
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- if True the audio_source will be looped for ever
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- precode wav files
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- lc3 coded files
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- just use a .lc3 file as audio_source
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- lc3 coded from ram
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- use a bytestring b'' as audio_source
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"""
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def __init__(
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self,
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bigs,
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global_config : auracast_config.AuracastGlobalConfig,
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big_config: List[auracast_config.AuracastBigConfig]
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):
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self.task = None
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self.is_streaming = False
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self.bigs = bigs
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self.global_config = global_config
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self.big_config = big_config
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def start_streaming(self):
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if not self.is_streaming:
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self.task = asyncio.create_task(self.stream())
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else:
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logging.warning('Streamer is already running')
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async def stop_streaming(self):
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"""Gracefully stop streaming and release audio devices."""
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if not self.is_streaming and self.task is None:
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return
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# Ask the streaming loop to finish
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self.is_streaming = False
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if self.task is not None:
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self.task.cancel()
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try:
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await self.task
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except asyncio.CancelledError:
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pass
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self.task = None
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# Close audio inputs (await to ensure ALSA devices are released)
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close_tasks = []
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for big in self.bigs.values():
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ai = big.get("audio_input")
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if ai and hasattr(ai, "close"):
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close_tasks.append(ai.close())
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# Remove reference so a fresh one is created next time
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big.pop("audio_input", None)
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if close_tasks:
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await asyncio.gather(*close_tasks, return_exceptions=True)
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async def stream(self):
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bigs = self.bigs
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big_config = self.big_config
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global_config = self.global_config
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for i, big in enumerate(bigs.values()):
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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['lc3_bytes_per_frame'] = global_config.octets_per_frame
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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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elif audio_source == 'webrtc':
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big['audio_input'] = WebRTCAudioInput()
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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=48000, # TODO: get samplerate from webrtc
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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['lc3_bytes_per_frame'] = global_config.octets_per_frame
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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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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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if big_config[i].loop:
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lc3_frames = itertools.cycle(lc3_frames)
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big['lc3_frames'] = lc3_frames
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# precoded lc3 file
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elif big_config[i].audio_source.endswith('.lc3'):
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big['precoded'] = True
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filename = big_config[i].audio_source.replace('file:', '')
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lc3_bytes = read_lc3_file(filename)
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lc3_frames = iter(lc3_bytes)
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if big_config[i].loop:
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lc3_frames = itertools.cycle(lc3_frames)
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big['lc3_frames'] = lc3_frames
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# use wav files and code them entirely before streaming
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elif big_config[i].precode_wav and big_config[i].audio_source.endswith('.wav'):
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big['precoded'] = True
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audio_input = await audio_io.create_audio_input(audio_source, input_format)
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audio_input.rewind = False
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pcm_format = await audio_input.open()
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if pcm_format.channels != 1:
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logging.error("Only 1 channels PCM configurations are supported")
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return
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if pcm_format.sample_type == audio_io.PcmFormat.SampleType.INT16:
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pcm_bit_depth = 16
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elif pcm_format.sample_type == audio_io.PcmFormat.SampleType.FLOAT32:
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pcm_bit_depth = None
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else:
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logging.error("Only INT16 and FLOAT32 sample types are supported")
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return
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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=pcm_format.sample_rate,
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)
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lc3_frame_samples = encoder.get_frame_samples() # number of the pcm samples per lc3 frame
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lc3_bytes = b''
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async for pcm_frame in audio_input.frames(lc3_frame_samples):
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lc3_bytes += encoder.encode(
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pcm_frame, num_bytes=global_config.octets_per_frame, bit_depth=pcm_bit_depth
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)
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lc3_frames = iter(lc3_bytes)
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# have a look at itertools.islice
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if big_config[i].loop:
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lc3_frames = itertools.cycle(lc3_frames)
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big['lc3_frames'] = lc3_frames
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# anything else, e.g. realtime stream from device (bumble)
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else:
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audio_input = await audio_io.create_audio_input(audio_source, input_format)
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# Store early so stop_streaming can close even if open() fails
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big['audio_input'] = audio_input
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# SoundDeviceAudioInput (used for `mic:<device>` captures) has no `.rewind`.
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if hasattr(audio_input, "rewind"):
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audio_input.rewind = big_config[i].loop
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# Retry logic – ALSA sometimes keeps the device busy for a short time after the
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# previous stream has closed. Handle PortAudioError -9985 with back-off retries.
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import sounddevice as _sd
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max_attempts = 3
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for attempt in range(1, max_attempts + 1):
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try:
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pcm_format = await audio_input.open()
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break # success
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except _sd.PortAudioError as err:
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# -9985 == paDeviceUnavailable
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logging.error('Could not open audio device %s with error %s', audio_source, err)
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code = None
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if hasattr(err, 'errno'):
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code = err.errno
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elif len(err.args) > 1 and isinstance(err.args[1], int):
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code = err.args[1]
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if code == -9985 and attempt < max_attempts:
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backoff_ms = 200 * attempt
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logging.warning("PortAudio device busy (attempt %d/%d). Retrying in %.1f ms…", attempt, max_attempts, backoff_ms)
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# ensure device handle and PortAudio context are closed before retrying
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try:
|
||
if hasattr(audio_input, "aclose"):
|
||
await audio_input.aclose()
|
||
elif hasattr(audio_input, "close"):
|
||
audio_input.close()
|
||
except Exception:
|
||
pass
|
||
# Fully terminate PortAudio to drop lingering handles (sounddevice quirk)
|
||
if hasattr(_sd, "_terminate"):
|
||
try:
|
||
_sd._terminate()
|
||
except Exception:
|
||
pass
|
||
# Small pause then re-initialize PortAudio
|
||
await asyncio.sleep(0.1)
|
||
if hasattr(_sd, "_initialize"):
|
||
try:
|
||
_sd._initialize()
|
||
except Exception:
|
||
pass
|
||
|
||
# Back-off before next attempt
|
||
await asyncio.sleep(backoff_ms / 1000)
|
||
# Recreate audio_input fresh for next attempt
|
||
audio_input = await audio_io.create_audio_input(audio_source, input_format)
|
||
continue
|
||
# Other errors or final attempt – re-raise so caller can abort gracefully
|
||
raise
|
||
else:
|
||
# Loop exhausted without break
|
||
logging.error("Unable to open audio device after %d attempts – giving up", max_attempts)
|
||
return
|
||
|
||
if pcm_format.channels != 1:
|
||
logging.info("Input device provides %d channels – will down-mix to mono for LC3", pcm_format.channels)
|
||
if pcm_format.sample_type == audio_io.PcmFormat.SampleType.INT16:
|
||
pcm_bit_depth = 16
|
||
elif pcm_format.sample_type == audio_io.PcmFormat.SampleType.FLOAT32:
|
||
pcm_bit_depth = None
|
||
else:
|
||
logging.error("Only INT16 and FLOAT32 sample types are supported")
|
||
return
|
||
encoder = lc3.Encoder(
|
||
frame_duration_us=global_config.frame_duration_us,
|
||
sample_rate_hz=global_config.auracast_sampling_rate_hz,
|
||
num_channels=1,
|
||
input_sample_rate_hz=pcm_format.sample_rate,
|
||
)
|
||
lc3_frame_samples = encoder.get_frame_samples() # number of the pcm samples per lc3 frame
|
||
|
||
big['pcm_bit_depth'] = pcm_bit_depth
|
||
big['channels'] = pcm_format.channels
|
||
big['lc3_frame_samples'] = lc3_frame_samples
|
||
big['lc3_bytes_per_frame'] = global_config.octets_per_frame
|
||
big['audio_input'] = audio_input
|
||
big['encoder'] = encoder
|
||
big['precoded'] = False
|
||
|
||
|
||
logging.info("Streaming audio...")
|
||
bigs = self.bigs
|
||
self.is_streaming = True
|
||
# One streamer fits all
|
||
while self.is_streaming:
|
||
stream_finished = [False for _ in range(len(bigs))]
|
||
for i, big in enumerate(bigs.values()):
|
||
|
||
if big['precoded']:# everything was already lc3 coded beforehand
|
||
lc3_frame = bytes(
|
||
itertools.islice(big['lc3_frames'], big['lc3_bytes_per_frame'])
|
||
)
|
||
|
||
if lc3_frame == b'': # Not all streams may stop at the same time
|
||
stream_finished[i] = True
|
||
continue
|
||
else: # code lc3 on the fly
|
||
pcm_frame = await anext(big['audio_input'].frames(big['lc3_frame_samples']), None)
|
||
|
||
if pcm_frame is None: # Not all streams may stop at the same time
|
||
stream_finished[i] = True
|
||
continue
|
||
|
||
# Down-mix multi-channel PCM to mono for LC3 encoder if needed
|
||
if big.get('channels', 1) > 1:
|
||
if isinstance(pcm_frame, np.ndarray):
|
||
if pcm_frame.ndim > 1:
|
||
mono = pcm_frame.mean(axis=1).astype(pcm_frame.dtype)
|
||
pcm_frame = mono
|
||
else:
|
||
# Convert raw bytes to numpy, average channels, convert back
|
||
dtype = np.int16 if big['pcm_bit_depth'] == 16 else np.float32
|
||
samples = np.frombuffer(pcm_frame, dtype=dtype)
|
||
samples = samples.reshape(-1, big['channels']).mean(axis=1)
|
||
pcm_frame = samples.astype(dtype).tobytes()
|
||
|
||
lc3_frame = big['encoder'].encode(
|
||
pcm_frame, num_bytes=big['lc3_bytes_per_frame'], bit_depth=big['pcm_bit_depth']
|
||
)
|
||
|
||
await big['iso_queue'].write(lc3_frame)
|
||
|
||
if all(stream_finished): # Take into account that multiple files have different lengths
|
||
logging.info('All streams finished, stopping streamer')
|
||
self.is_streaming = False
|
||
break
|
||
|
||
|
||
# -----------------------------------------------------------------------------
|
||
# Main
|
||
# -----------------------------------------------------------------------------
|
||
|
||
async def broadcast(global_conf: auracast_config.AuracastGlobalConfig, big_conf: List[auracast_config.AuracastBigConfig]):
|
||
"""Start a broadcast."""
|
||
|
||
if global_conf.transport == 'auto':
|
||
devices = glob.glob('/dev/serial/by-id/*')
|
||
logging.info('Found serial devices: %s', devices)
|
||
for device in devices:
|
||
if 'usb-ZEPHYR_Zephyr_HCI_UART_sample' in device:
|
||
logging.info('Using: %s', device)
|
||
global_conf.transport = f'serial:{device},115200,rtscts'
|
||
break
|
||
|
||
# check again if transport is still auto
|
||
if global_conf.transport == 'auto':
|
||
raise AssertionError('No suitable transport found.')
|
||
|
||
async with create_device(global_conf) as device:
|
||
if not device.supports_le_periodic_advertising:
|
||
logger.error(color('Periodic advertising not supported', 'red'))
|
||
return
|
||
|
||
bigs = await init_broadcast( # the bigs dictionary contains all the global configurations
|
||
device,
|
||
global_conf,
|
||
big_conf
|
||
)
|
||
streamer = Streamer(bigs, global_conf, big_conf)
|
||
streamer.start_streaming()
|
||
|
||
await asyncio.wait([streamer.task])
|
||
|
||
|
||
# -----------------------------------------------------------------------------
|
||
if __name__ == "__main__":
|
||
import os
|
||
|
||
logging.basicConfig( #export LOG_LEVEL=INFO
|
||
level=os.environ.get('LOG_LEVEL', logging.DEBUG),
|
||
format='%(module)s.py:%(lineno)d %(levelname)s: %(message)s'
|
||
)
|
||
os.chdir(os.path.dirname(__file__))
|
||
|
||
|
||
config = auracast_config.AuracastConfigGroup(
|
||
bigs = [
|
||
auracast_config.AuracastBigConfigDeu(),
|
||
#auracast_config.AuracastBigConfigEng(),
|
||
#auracast_config.AuracastBigConfigFra(),
|
||
#auracast_config.AuracastBigConfigEs(),
|
||
#auracast_config.AuracastBigConfigIt(),
|
||
]
|
||
)
|
||
|
||
# TODO: How can we use other iso interval than 10ms ?(medium or low rel) ? - nrf53audio receiver repports I2S tx underrun
|
||
config.qos_config=auracast_config.AuracastQosHigh()
|
||
|
||
#config.transport='serial:/dev/serial/by-id/usb-ZEPHYR_Zephyr_HCI_UART_sample_81BD14B8D71B5662-if00,1000000,rtscts' # transport for nrf52 dongle
|
||
#config.transport='serial:/dev/serial/by-id/usb-SEGGER_J-Link_001050076061-if02,1000000,rtscts' # transport for nrf53dk
|
||
#config.transport='serial:/dev/serial/by-id/usb-SEGGER_J-Link_001057705357-if02,1000000,rtscts' # transport for nrf54l15dk
|
||
#config.transport='serial:/dev/serial/by-id/usb-ZEPHYR_Zephyr_HCI_UART_sample_95A087EADB030B24-if00,115200,rtscts' #nrf52dongle hci_uart usb cdc
|
||
#config.transport='usb:2fe3:000b' #nrf52dongle hci_usb # TODO: iso packet over usb not supported
|
||
config.transport= 'auto'
|
||
#config.transport='serial:/dev/ttyAMA2,1000000,rtscts' # transport for raspberry pi
|
||
|
||
|
||
for big in config.bigs: # TODO: encrypted streams are not working
|
||
#big.code = 'ff'*16 # returns hci/HCI_ENCRYPTION_MODE_NOT_ACCEPTABLE_ERROR
|
||
#big.code = '78 e5 dc f1 34 ab 42 bf c1 92 ef dd 3a fd 67 ae'
|
||
big.precode_wav = True
|
||
big.audio_source = big.audio_source.replace('.wav', '_10_16_32.lc3') #lc3 precoded files
|
||
big.audio_source = read_lc3_file(big.audio_source) # load files in advance
|
||
|
||
# --- Network_uncoded mode using NetworkAudioReceiver ---
|
||
#big.audio_source = NetworkAudioReceiverUncoded(port=50007, samplerate=16000, channels=1, chunk_size=1024)
|
||
|
||
# 16kHz works reliably with 3 streams
|
||
# 24kHz is only working with 2 streams - probably airtime constraint
|
||
# TODO: with more than three broadcasters (16kHz) no advertising (no primary channels is present anymore)
|
||
# TODO: find the bottleneck - probably airtime
|
||
# TODO: test encrypted streams
|
||
|
||
config.auracast_sampling_rate_hz = 16000
|
||
config.octets_per_frame = 40 # 32kbps@16kHz
|
||
#config.debug = True
|
||
|
||
run_async(
|
||
broadcast(
|
||
config,
|
||
config.bigs
|
||
)
|
||
)
|
||
|
||
# TODO: possible inputs:
|
||
# wav file locally
|
||
# precoded lc3 file locally
|
||
# realtime audio locally
|
||
# realtime audio network lc3 coded
|
||
# (realtime audio network uncoded)
|