Further split between sound and core modules
This commit is contained in:
Fabio Manganiello
parent
65465e3a18
commit
3d2636b09c
2 changed files with 655 additions and 646 deletions
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@ -2,653 +2,8 @@
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.. moduleauthor:: Fabio Manganiello <blacklight86@gmail.com>
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"""
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import json
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import math
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import os
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import queue
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import tempfile
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import time
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from enum import Enum
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from threading import Thread, Event, RLock
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from .core import Sound
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from platypush.plugins import Plugin, action
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class PlaybackState(Enum):
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STOPPED='STOPPED',
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PLAYING='PLAYING',
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PAUSED='PAUSED'
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class RecordingState(Enum):
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STOPPED='STOPPED',
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RECORDING='RECORDING',
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PAUSED='PAUSED'
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class SoundPlugin(Plugin):
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"""
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Plugin to interact with a sound device.
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Requires:
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* **sounddevice** (``pip install sounddevice``)
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* **soundfile** (``pip install soundfile``)
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* **numpy** (``pip install numpy``)
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"""
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def __init__(self, input_device=None, output_device=None,
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input_blocksize=Sound._DEFAULT_BLOCKSIZE,
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output_blocksize=Sound._DEFAULT_BLOCKSIZE,
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playback_bufsize=Sound._DEFAULT_BUFSIZE, *args, **kwargs):
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"""
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:param input_device: Index or name of the default input device. Use :method:`platypush.plugins.sound.query_devices` to get the available devices. Default: system default
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:type input_device: int or str
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:param output_device: Index or name of the default output device. Use :method:`platypush.plugins.sound.query_devices` to get the available devices. Default: system default
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:type output_device: int or str
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:param input_blocksize: Blocksize to be applied to the input device. Try to increase this value if you get input overflow errors while recording. Default: 2048
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:type input_blocksize: int
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:param output_blocksize: Blocksize to be applied to the output device. Try to increase this value if you get output underflow errors while playing. Default: 2048
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:type output_blocksize: int
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:param playback_bufsize: Number of audio blocks that will be cached while playing (default: 20)
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:type playback_bufsize: int
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"""
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super().__init__(*args, **kwargs)
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self.input_device = input_device
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self.output_device = output_device
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self.input_blocksize = input_blocksize
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self.output_blocksize = output_blocksize
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self.playback_bufsize = playback_bufsize
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self.playback_state = PlaybackState.STOPPED
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self.playback_state_lock = RLock()
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self.playback_paused_changed = Event()
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self.recording_state = RecordingState.STOPPED
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self.recording_state_lock = RLock()
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self.recording_paused_changed = Event()
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self.active_streams = {}
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self.completed_callback_events = {}
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def _get_default_device(self, category):
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"""
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Query the default audio devices.
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:param category: Device category to query. Can be either input or output
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:type category: str
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"""
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import sounddevice as sd
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return sd.query_hostapis()[0].get('default_' + category.lower() + '_device')
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@action
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def query_devices(self, category=None):
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"""
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Query the available devices
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:param category: Device category to query. Can be either input or output. Default: None (query all devices)
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:type category: str
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:returns: A dictionary representing the available devices. Example::
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[
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{
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"name": "pulse",
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"hostapi": 0,
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"max_input_channels": 32,
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"max_output_channels": 32,
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"default_low_input_latency": 0.008684807256235827,
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"default_low_output_latency": 0.008684807256235827,
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"default_high_input_latency": 0.034807256235827665,
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"default_high_output_latency": 0.034807256235827665,
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"default_samplerate": 44100
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},
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{
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"name": "default",
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"hostapi": 0,
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"max_input_channels": 32,
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"max_output_channels": 32,
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"default_low_input_latency": 0.008684807256235827,
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"default_low_output_latency": 0.008684807256235827,
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"default_high_input_latency": 0.034807256235827665,
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"default_high_output_latency": 0.034807256235827665,
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"default_samplerate": 44100
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}
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]
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"""
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import sounddevice as sd
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devs = sd.query_devices()
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if category == 'input':
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devs = [d for d in devs if d.get('max_input_channels') > 0]
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elif category == 'output':
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devs = [d for d in devs if d.get('max_output_channels') > 0]
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return devs
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def _play_audio_callback(self, q, blocksize, streamtype):
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import sounddevice as sd
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is_raw_stream = streamtype == sd.RawOutputStream
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def audio_callback(outdata, frames, time, status):
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if self._get_playback_state() == PlaybackState.STOPPED:
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raise sd.CallbackAbort
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while self._get_playback_state() == PlaybackState.PAUSED:
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self.playback_paused_changed.wait()
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assert frames == blocksize
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if status.output_underflow:
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self.logger.warning('Output underflow: increase blocksize?')
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outdata = (b'\x00' if is_raw_stream else 0.) * len(outdata)
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return
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assert not status
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try:
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data = q.get_nowait()
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except queue.Empty:
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self.logger.warning('Buffer is empty: increase buffersize?')
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raise sd.CallbackAbort
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if len(data) < len(outdata):
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outdata[:len(data)] = data
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outdata[len(data):] = (b'\x00' if is_raw_stream else 0.) * \
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(len(outdata) - len(data))
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else:
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outdata[:] = data
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return audio_callback
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@action
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def play(self, file=None, sound=None, device=None, blocksize=None,
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bufsize=Sound._DEFAULT_BUFSIZE, samplerate=None, channels=None):
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"""
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Plays a sound file (support formats: wav, raw) or a synthetic sound.
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:param file: Sound file path. Specify this if you want to play a file
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:type file: str
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:param sound: Sound to play. Specify this if you want to play
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synthetic sounds. You can also create polyphonic sounds by just
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calling play multple times.
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:type sound: Sound. You can initialize it either from a list
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of `Sound` objects or from its JSON representation, e.g.:
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{
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"midi_note": 69, # 440 Hz A
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"gain": 1.0, # Maximum volume
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"duration": 1.0 # 1 second or until release/pause/stop
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}
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:param device: Output device (default: default configured device or
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system default audio output if not configured)
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:type device: int or str
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:param blocksize: Audio block size (default: configured
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`output_blocksize` or 2048)
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:type blocksize: int
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:param bufsize: Size of the audio buffer (default: 20)
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:type bufsize: int
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:param samplerate: Audio samplerate. Default: audio file samplerate if
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in file mode, 44100 Hz if in synth mode
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:type samplerate: int
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:param channels: Number of audio channels. Default: number of channels
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in the audio file in file mode, 1 if in synth mode
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:type channels: int
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"""
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if not file and not sound:
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raise RuntimeError('Please specify either a file to play or a ' +
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'list of sound objects')
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import sounddevice as sd
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if blocksize is None:
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blocksize = self.output_blocksize
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self.playback_paused_changed.clear()
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stream_index = None
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q = queue.Queue(maxsize=bufsize)
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f = None
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t = 0.
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if file:
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file = os.path.abspath(os.path.expanduser(file))
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if device is None:
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device = self.output_device
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if device is None:
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device = self._get_default_device('output')
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try:
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if file:
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import soundfile as sf
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f = sf.SoundFile(file)
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if not samplerate:
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samplerate = f.samplerate if f else Sound._DEFAULT_SAMPLERATE
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if not channels:
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channels = f.channels if f else 1
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self.logger.info('Starting playback of {} to device [{}]'.
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format(file or sound, device))
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if sound:
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sound = Sound.build(sound)
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# Audio queue pre-fill loop
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for _ in range(bufsize):
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if f:
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data = f.buffer_read(blocksize, dtype='float32')
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if not data:
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break
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else:
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blocktime = float(blocksize / samplerate)
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next_t = min(t+blocktime, sound.duration) \
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if sound.duration is not None else t+blocktime
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data = sound.get_wave(t_start=t, t_end=next_t,
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samplerate=samplerate)
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t = next_t
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if sound.duration is not None and t >= sound.duration:
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break
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while self._get_playback_state() == PlaybackState.PAUSED:
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self.playback_paused_changed.wait()
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q.put_nowait(data) # Pre-fill the audio queue
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streamtype = sd.RawOutputStream if file else sd.OutputStream
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completed_callback_event = Event()
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stream = streamtype(samplerate=samplerate, blocksize=blocksize,
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device=device, channels=channels,
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dtype='float32',
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callback=self._play_audio_callback(
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q=q, blocksize=blocksize,
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streamtype=streamtype),
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finished_callback=completed_callback_event.set)
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stream_index = self.start_playback(stream, completed_callback_event)
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with stream:
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# Timeout set until we expect all the buffered blocks to
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# be consumed
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timeout = blocksize * bufsize / samplerate
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while True:
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while self._get_playback_state() == PlaybackState.PAUSED:
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self.playback_paused_changed.wait()
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if f:
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data = f.buffer_read(blocksize, dtype='float32')
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if not data:
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break
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else:
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blocktime = float(blocksize / samplerate)
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next_t = min(t+blocktime, sound.duration) \
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if sound.duration is not None else t+blocktime
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data = sound.get_wave(t_start=t, t_end=next_t,
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samplerate=samplerate)
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t = next_t
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if sound.duration is not None and t >= sound.duration:
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break
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if self._get_playback_state() == PlaybackState.STOPPED:
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raise sd.CallbackAbort
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try:
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q.put(data, timeout=timeout)
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except queue.Full as e:
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if self._get_playback_state() != PlaybackState.PAUSED:
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raise e
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completed_callback_event.wait()
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except queue.Full as e:
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self.logger.warning('Playback timeout: audio callback failed?')
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finally:
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if f and not f.closed:
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f.close()
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f = None
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self.stop_playback(stream_index)
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@action
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def record(self, file=None, duration=None, device=None, sample_rate=None,
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blocksize=None, latency=0, channels=1, subtype='PCM_24'):
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"""
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Records audio to a sound file (support formats: wav, raw)
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:param file: Sound file (default: the method will create a temporary file with the recording)
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:type file: str
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:param duration: Recording duration in seconds (default: record until stop event)
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:type duration: float
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:param device: Input device (default: default configured device or system default audio input if not configured)
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:type device: int or str
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:param sample_rate: Recording sample rate (default: device default rate)
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:type sample_rate: int
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:param blocksize: Audio block size (default: configured `input_blocksize` or 2048)
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:type blocksize: int
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:param latency: Device latency in seconds (default: 0)
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:type latency: float
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:param channels: Number of channels (default: 1)
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:type channels: int
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:param subtype: Recording subtype - see `soundfile docs <https://pysoundfile.readthedocs.io/en/0.9.0/#soundfile.available_subtypes>`_ for a list of the available subtypes (default: PCM_24)
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:type subtype: str
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"""
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import sounddevice as sd
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self.recording_paused_changed.clear()
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if file:
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file = os.path.abspath(os.path.expanduser(file))
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else:
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file = tempfile.mktemp(prefix='platypush_recording_', suffix='.wav',
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dir='')
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if os.path.isfile(file):
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self.logger.info('Removing existing audio file {}'.format(file))
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os.unlink(file)
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if device is None:
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device = self.input_device
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if device is None:
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device = self._get_default_device('input')
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if sample_rate is None:
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dev_info = sd.query_devices(device, 'input')
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sample_rate = int(dev_info['default_samplerate'])
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if blocksize is None:
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blocksize = self.input_blocksize
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q = queue.Queue()
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def audio_callback(indata, frames, time, status):
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while self._get_recording_state() == RecordingState.PAUSED:
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self.recording_paused_changed.wait()
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if status:
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self.logger.warning('Recording callback status: {}'.format(
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str(status)))
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q.put(indata.copy())
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try:
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import soundfile as sf
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import numpy
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with sf.SoundFile(file, mode='x', samplerate=sample_rate,
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channels=channels, subtype=subtype) as f:
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with sd.InputStream(samplerate=sample_rate, device=device,
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channels=channels, callback=audio_callback,
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latency=latency, blocksize=blocksize):
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self.start_recording()
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self.logger.info('Started recording from device [{}] to [{}]'.
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format(device, file))
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recording_started_time = time.time()
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while self._get_recording_state() != RecordingState.STOPPED \
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and (duration is None or
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time.time() - recording_started_time < duration):
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while self._get_recording_state() == RecordingState.PAUSED:
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self.recording_paused_changed.wait()
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get_args = {
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'block': True,
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'timeout': max(0, duration - (time.time() -
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recording_started_time))
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} if duration is not None else {}
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data = q.get(**get_args)
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f.write(data)
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f.flush()
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except queue.Empty as e:
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self.logger.warning('Recording timeout: audio callback failed?')
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finally:
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self.stop_recording()
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@action
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def recordplay(self, duration=None, input_device=None, output_device=None,
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sample_rate=None, blocksize=None, latency=0, channels=1,
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dtype=None):
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"""
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Records audio and plays it on an output sound device (audio pass-through)
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:param duration: Recording duration in seconds (default: record until stop event)
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:type duration: float
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:param input_device: Input device (default: default configured device or system default audio input if not configured)
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:type input_device: int or str
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:param output_device: Output device (default: default configured device or system default audio output if not configured)
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:type output_device: int or str
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:param sample_rate: Recording sample rate (default: device default rate)
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:type sample_rate: int
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:param blocksize: Audio block size (default: configured `output_blocksize` or 2048)
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:type blocksize: int
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:param latency: Device latency in seconds (default: 0)
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:type latency: float
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:param channels: Number of channels (default: 1)
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:type channels: int
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:param dtype: Data type for the recording - see `soundfile docs <https://python-sounddevice.readthedocs.io/en/0.3.12/_modules/sounddevice.html#rec>`_ for available types (default: input device default)
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:type dtype: str
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"""
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import sounddevice as sd
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self.playback_paused_changed.clear()
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self.recording_paused_changed.clear()
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if input_device is None:
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input_device = self.input_device
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if input_device is None:
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input_device = self._get_default_device('input')
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if output_device is None:
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output_device = self.output_device
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if output_device is None:
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output_device = self._get_default_device('output')
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if sample_rate is None:
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dev_info = sd.query_devices(input_device, 'input')
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sample_rate = int(dev_info['default_samplerate'])
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if blocksize is None:
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blocksize = self.output_blocksize
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def audio_callback(indata, outdata, frames, time, status):
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while self._get_recording_state() == RecordingState.PAUSED:
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self.recording_paused_changed.wait()
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if status:
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self.logger.warning('Recording callback status: {}'.format(
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str(status)))
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||||
|
||||
outdata[:] = indata
|
||||
|
||||
|
||||
stream_index = None
|
||||
|
||||
try:
|
||||
import soundfile as sf
|
||||
import numpy
|
||||
|
||||
stream = sd.Stream(samplerate=sample_rate, channels=channels,
|
||||
blocksize=blocksize, latency=latency,
|
||||
device=(input_device, output_device),
|
||||
dtype=dtype, callback=audio_callback)
|
||||
self.start_recording()
|
||||
stream_index = self.start_playback(stream)
|
||||
|
||||
self.logger.info('Started recording pass-through from device ' +
|
||||
'[{}] to device [{}]'.
|
||||
format(input_device, output_device))
|
||||
|
||||
recording_started_time = time.time()
|
||||
|
||||
while self._get_recording_state() != RecordingState.STOPPED \
|
||||
and (duration is None or
|
||||
time.time() - recording_started_time < duration):
|
||||
while self._get_recording_state() == RecordingState.PAUSED:
|
||||
self.recording_paused_changed.wait()
|
||||
|
||||
time.sleep(0.1)
|
||||
|
||||
except queue.Empty as e:
|
||||
self.logger.warning('Recording timeout: audio callback failed?')
|
||||
finally:
|
||||
self.stop_playback(stream_index)
|
||||
self.stop_recording()
|
||||
|
||||
|
||||
@action
|
||||
def get_active_streams(self):
|
||||
"""
|
||||
:returns: A list of active players
|
||||
"""
|
||||
return {
|
||||
i: {
|
||||
attr: getattr(stream, attr)
|
||||
for attr in ['active', 'closed', 'stopped', 'blocksize',
|
||||
'channels', 'cpu_load', 'device', 'dtype',
|
||||
'latency', 'samplerate', 'samplesize']
|
||||
if hasattr(stream, attr)
|
||||
} for i, stream in self.active_streams.items()
|
||||
}
|
||||
|
||||
|
||||
def start_playback(self, stream, completed_callback_event=None):
|
||||
stream_index = None
|
||||
with self.playback_state_lock:
|
||||
self.playback_state = PlaybackState.PLAYING
|
||||
stream_index = None
|
||||
for i in range(len(self.active_streams)+1):
|
||||
if i not in self.active_streams:
|
||||
stream_index = i
|
||||
break
|
||||
|
||||
self.active_streams[stream_index] = stream
|
||||
self.completed_callback_events[stream_index] = \
|
||||
completed_callback_event if completed_callback_event else Event()
|
||||
|
||||
self.logger.info('Playback started on stream index {}'.
|
||||
format(stream_index))
|
||||
return stream_index
|
||||
|
||||
@action
|
||||
def stop_playback(self, *streams):
|
||||
with self.playback_state_lock:
|
||||
if not streams:
|
||||
streams = self.active_streams.keys()
|
||||
updated_n_players = len(self.active_streams)
|
||||
completed_callback_events = {}
|
||||
|
||||
for i in streams:
|
||||
if i is None or not (i in self.active_streams):
|
||||
continue
|
||||
|
||||
stream = self.active_streams[i]
|
||||
updated_n_players -= 1
|
||||
if self.completed_callback_events[i]:
|
||||
completed_callback_events[i] = self.completed_callback_events[i]
|
||||
|
||||
if not updated_n_players:
|
||||
self.playback_state = PlaybackState.STOPPED
|
||||
|
||||
for i, event in completed_callback_events.items():
|
||||
event.wait()
|
||||
del self.completed_callback_events[i]
|
||||
del self.active_streams[i]
|
||||
|
||||
self.logger.info('Playback stopped')
|
||||
|
||||
@action
|
||||
def pause_playback(self):
|
||||
with self.playback_state_lock:
|
||||
if self.playback_state == PlaybackState.PAUSED:
|
||||
self.playback_state = PlaybackState.PLAYING
|
||||
elif self.playback_state == PlaybackState.PLAYING:
|
||||
self.playback_state = PlaybackState.PAUSED
|
||||
else:
|
||||
return
|
||||
|
||||
self.playback_paused_changed.set()
|
||||
self.logger.info('Playback ' + ('paused' if self.playback_state ==
|
||||
PlaybackState.PAUSED else 'playing'))
|
||||
|
||||
def start_recording(self):
|
||||
with self.recording_state_lock:
|
||||
self.recording_state = RecordingState.RECORDING
|
||||
|
||||
@action
|
||||
def stop_recording(self):
|
||||
with self.recording_state_lock:
|
||||
self.recording_state = RecordingState.STOPPED
|
||||
self.logger.info('Recording stopped')
|
||||
|
||||
@action
|
||||
def pause_recording(self):
|
||||
with self.recording_state_lock:
|
||||
if self.recording_state == RecordingState.PAUSED:
|
||||
self.recording_state = RecordingState.RECORDING
|
||||
elif self.recording_state == RecordingState.RECORDING:
|
||||
self.recording_state = RecordingState.PAUSED
|
||||
else:
|
||||
return
|
||||
|
||||
self.logger.info('Recording paused state toggled')
|
||||
self.recording_paused_changed.set()
|
||||
|
||||
def _get_playback_state(self):
|
||||
with self.playback_state_lock:
|
||||
return self.playback_state
|
||||
|
||||
def _get_recording_state(self):
|
||||
with self.recording_state_lock:
|
||||
return self.recording_state
|
||||
|
||||
@action
|
||||
def get_state(self):
|
||||
return {
|
||||
'playback_state': self._get_playback_state().name,
|
||||
'recording_state': self._get_recording_state().name,
|
||||
}
|
||||
from .plugin import SoundPlugin
|
||||
|
||||
|
||||
# vim:sw=4:ts=4:et:
|
||||
|
|
|
|||
654
platypush/plugins/sound/plugin.py
Normal file
654
platypush/plugins/sound/plugin.py
Normal file
|
|
@ -0,0 +1,654 @@
|
|||
"""
|
||||
.. moduleauthor:: Fabio Manganiello <blacklight86@gmail.com>
|
||||
"""
|
||||
|
||||
import json
|
||||
import math
|
||||
import os
|
||||
import queue
|
||||
import tempfile
|
||||
import time
|
||||
|
||||
from enum import Enum
|
||||
from threading import Thread, Event, RLock
|
||||
|
||||
from .core import Sound
|
||||
from platypush.plugins import Plugin, action
|
||||
|
||||
|
||||
class PlaybackState(Enum):
|
||||
STOPPED='STOPPED',
|
||||
PLAYING='PLAYING',
|
||||
PAUSED='PAUSED'
|
||||
|
||||
|
||||
class RecordingState(Enum):
|
||||
STOPPED='STOPPED',
|
||||
RECORDING='RECORDING',
|
||||
PAUSED='PAUSED'
|
||||
|
||||
|
||||
class SoundPlugin(Plugin):
|
||||
"""
|
||||
Plugin to interact with a sound device.
|
||||
|
||||
Requires:
|
||||
|
||||
* **sounddevice** (``pip install sounddevice``)
|
||||
* **soundfile** (``pip install soundfile``)
|
||||
* **numpy** (``pip install numpy``)
|
||||
"""
|
||||
|
||||
def __init__(self, input_device=None, output_device=None,
|
||||
input_blocksize=Sound._DEFAULT_BLOCKSIZE,
|
||||
output_blocksize=Sound._DEFAULT_BLOCKSIZE,
|
||||
playback_bufsize=Sound._DEFAULT_BUFSIZE, *args, **kwargs):
|
||||
"""
|
||||
:param input_device: Index or name of the default input device. Use :method:`platypush.plugins.sound.query_devices` to get the available devices. Default: system default
|
||||
:type input_device: int or str
|
||||
|
||||
:param output_device: Index or name of the default output device. Use :method:`platypush.plugins.sound.query_devices` to get the available devices. Default: system default
|
||||
:type output_device: int or str
|
||||
|
||||
:param input_blocksize: Blocksize to be applied to the input device. Try to increase this value if you get input overflow errors while recording. Default: 2048
|
||||
:type input_blocksize: int
|
||||
|
||||
:param output_blocksize: Blocksize to be applied to the output device. Try to increase this value if you get output underflow errors while playing. Default: 2048
|
||||
:type output_blocksize: int
|
||||
|
||||
:param playback_bufsize: Number of audio blocks that will be cached while playing (default: 20)
|
||||
:type playback_bufsize: int
|
||||
"""
|
||||
|
||||
super().__init__(*args, **kwargs)
|
||||
|
||||
self.input_device = input_device
|
||||
self.output_device = output_device
|
||||
self.input_blocksize = input_blocksize
|
||||
self.output_blocksize = output_blocksize
|
||||
self.playback_bufsize = playback_bufsize
|
||||
|
||||
self.playback_state = PlaybackState.STOPPED
|
||||
self.playback_state_lock = RLock()
|
||||
self.playback_paused_changed = Event()
|
||||
self.recording_state = RecordingState.STOPPED
|
||||
self.recording_state_lock = RLock()
|
||||
self.recording_paused_changed = Event()
|
||||
self.active_streams = {}
|
||||
self.completed_callback_events = {}
|
||||
|
||||
def _get_default_device(self, category):
|
||||
"""
|
||||
Query the default audio devices.
|
||||
|
||||
:param category: Device category to query. Can be either input or output
|
||||
:type category: str
|
||||
"""
|
||||
|
||||
import sounddevice as sd
|
||||
return sd.query_hostapis()[0].get('default_' + category.lower() + '_device')
|
||||
|
||||
@action
|
||||
def query_devices(self, category=None):
|
||||
"""
|
||||
Query the available devices
|
||||
|
||||
:param category: Device category to query. Can be either input or output. Default: None (query all devices)
|
||||
:type category: str
|
||||
|
||||
:returns: A dictionary representing the available devices. Example::
|
||||
|
||||
[
|
||||
{
|
||||
"name": "pulse",
|
||||
"hostapi": 0,
|
||||
"max_input_channels": 32,
|
||||
"max_output_channels": 32,
|
||||
"default_low_input_latency": 0.008684807256235827,
|
||||
"default_low_output_latency": 0.008684807256235827,
|
||||
"default_high_input_latency": 0.034807256235827665,
|
||||
"default_high_output_latency": 0.034807256235827665,
|
||||
"default_samplerate": 44100
|
||||
},
|
||||
{
|
||||
"name": "default",
|
||||
"hostapi": 0,
|
||||
"max_input_channels": 32,
|
||||
"max_output_channels": 32,
|
||||
"default_low_input_latency": 0.008684807256235827,
|
||||
"default_low_output_latency": 0.008684807256235827,
|
||||
"default_high_input_latency": 0.034807256235827665,
|
||||
"default_high_output_latency": 0.034807256235827665,
|
||||
"default_samplerate": 44100
|
||||
}
|
||||
]
|
||||
|
||||
"""
|
||||
|
||||
import sounddevice as sd
|
||||
|
||||
devs = sd.query_devices()
|
||||
if category == 'input':
|
||||
devs = [d for d in devs if d.get('max_input_channels') > 0]
|
||||
elif category == 'output':
|
||||
devs = [d for d in devs if d.get('max_output_channels') > 0]
|
||||
|
||||
return devs
|
||||
|
||||
def _play_audio_callback(self, q, blocksize, streamtype):
|
||||
import sounddevice as sd
|
||||
|
||||
is_raw_stream = streamtype == sd.RawOutputStream
|
||||
def audio_callback(outdata, frames, time, status):
|
||||
if self._get_playback_state() == PlaybackState.STOPPED:
|
||||
raise sd.CallbackAbort
|
||||
|
||||
while self._get_playback_state() == PlaybackState.PAUSED:
|
||||
self.playback_paused_changed.wait()
|
||||
|
||||
assert frames == blocksize
|
||||
if status.output_underflow:
|
||||
self.logger.warning('Output underflow: increase blocksize?')
|
||||
outdata = (b'\x00' if is_raw_stream else 0.) * len(outdata)
|
||||
return
|
||||
|
||||
assert not status
|
||||
|
||||
try:
|
||||
data = q.get_nowait()
|
||||
except queue.Empty:
|
||||
self.logger.warning('Buffer is empty: increase buffersize?')
|
||||
raise sd.CallbackAbort
|
||||
|
||||
if len(data) < len(outdata):
|
||||
outdata[:len(data)] = data
|
||||
outdata[len(data):] = (b'\x00' if is_raw_stream else 0.) * \
|
||||
(len(outdata) - len(data))
|
||||
else:
|
||||
outdata[:] = data
|
||||
|
||||
return audio_callback
|
||||
|
||||
|
||||
@action
|
||||
def play(self, file=None, sound=None, device=None, blocksize=None,
|
||||
bufsize=Sound._DEFAULT_BUFSIZE, samplerate=None, channels=None):
|
||||
"""
|
||||
Plays a sound file (support formats: wav, raw) or a synthetic sound.
|
||||
|
||||
:param file: Sound file path. Specify this if you want to play a file
|
||||
:type file: str
|
||||
|
||||
:param sound: Sound to play. Specify this if you want to play
|
||||
synthetic sounds. You can also create polyphonic sounds by just
|
||||
calling play multple times.
|
||||
:type sound: Sound. You can initialize it either from a list
|
||||
of `Sound` objects or from its JSON representation, e.g.:
|
||||
|
||||
{
|
||||
"midi_note": 69, # 440 Hz A
|
||||
"gain": 1.0, # Maximum volume
|
||||
"duration": 1.0 # 1 second or until release/pause/stop
|
||||
}
|
||||
|
||||
:param device: Output device (default: default configured device or
|
||||
system default audio output if not configured)
|
||||
:type device: int or str
|
||||
|
||||
:param blocksize: Audio block size (default: configured
|
||||
`output_blocksize` or 2048)
|
||||
:type blocksize: int
|
||||
|
||||
:param bufsize: Size of the audio buffer (default: 20)
|
||||
:type bufsize: int
|
||||
|
||||
:param samplerate: Audio samplerate. Default: audio file samplerate if
|
||||
in file mode, 44100 Hz if in synth mode
|
||||
:type samplerate: int
|
||||
|
||||
:param channels: Number of audio channels. Default: number of channels
|
||||
in the audio file in file mode, 1 if in synth mode
|
||||
:type channels: int
|
||||
"""
|
||||
|
||||
if not file and not sound:
|
||||
raise RuntimeError('Please specify either a file to play or a ' +
|
||||
'list of sound objects')
|
||||
|
||||
import sounddevice as sd
|
||||
|
||||
if blocksize is None:
|
||||
blocksize = self.output_blocksize
|
||||
|
||||
self.playback_paused_changed.clear()
|
||||
|
||||
stream_index = None
|
||||
q = queue.Queue(maxsize=bufsize)
|
||||
f = None
|
||||
t = 0.
|
||||
|
||||
if file:
|
||||
file = os.path.abspath(os.path.expanduser(file))
|
||||
|
||||
if device is None:
|
||||
device = self.output_device
|
||||
if device is None:
|
||||
device = self._get_default_device('output')
|
||||
|
||||
try:
|
||||
if file:
|
||||
import soundfile as sf
|
||||
f = sf.SoundFile(file)
|
||||
if not samplerate:
|
||||
samplerate = f.samplerate if f else Sound._DEFAULT_SAMPLERATE
|
||||
if not channels:
|
||||
channels = f.channels if f else 1
|
||||
|
||||
self.logger.info('Starting playback of {} to device [{}]'.
|
||||
format(file or sound, device))
|
||||
|
||||
if sound:
|
||||
sound = Sound.build(sound)
|
||||
|
||||
# Audio queue pre-fill loop
|
||||
for _ in range(bufsize):
|
||||
if f:
|
||||
data = f.buffer_read(blocksize, dtype='float32')
|
||||
if not data:
|
||||
break
|
||||
else:
|
||||
blocktime = float(blocksize / samplerate)
|
||||
next_t = min(t+blocktime, sound.duration) \
|
||||
if sound.duration is not None else t+blocktime
|
||||
|
||||
data = sound.get_wave(t_start=t, t_end=next_t,
|
||||
samplerate=samplerate)
|
||||
t = next_t
|
||||
|
||||
if sound.duration is not None and t >= sound.duration:
|
||||
break
|
||||
|
||||
while self._get_playback_state() == PlaybackState.PAUSED:
|
||||
self.playback_paused_changed.wait()
|
||||
|
||||
q.put_nowait(data) # Pre-fill the audio queue
|
||||
|
||||
streamtype = sd.RawOutputStream if file else sd.OutputStream
|
||||
completed_callback_event = Event()
|
||||
stream = streamtype(samplerate=samplerate, blocksize=blocksize,
|
||||
device=device, channels=channels,
|
||||
dtype='float32',
|
||||
callback=self._play_audio_callback(
|
||||
q=q, blocksize=blocksize,
|
||||
streamtype=streamtype),
|
||||
finished_callback=completed_callback_event.set)
|
||||
|
||||
stream_index = self.start_playback(stream, completed_callback_event)
|
||||
|
||||
with stream:
|
||||
# Timeout set until we expect all the buffered blocks to
|
||||
# be consumed
|
||||
timeout = blocksize * bufsize / samplerate
|
||||
|
||||
while True:
|
||||
while self._get_playback_state() == PlaybackState.PAUSED:
|
||||
self.playback_paused_changed.wait()
|
||||
|
||||
if f:
|
||||
data = f.buffer_read(blocksize, dtype='float32')
|
||||
if not data:
|
||||
break
|
||||
else:
|
||||
blocktime = float(blocksize / samplerate)
|
||||
next_t = min(t+blocktime, sound.duration) \
|
||||
if sound.duration is not None else t+blocktime
|
||||
|
||||
data = sound.get_wave(t_start=t, t_end=next_t,
|
||||
samplerate=samplerate)
|
||||
t = next_t
|
||||
|
||||
if sound.duration is not None and t >= sound.duration:
|
||||
break
|
||||
|
||||
if self._get_playback_state() == PlaybackState.STOPPED:
|
||||
raise sd.CallbackAbort
|
||||
|
||||
try:
|
||||
q.put(data, timeout=timeout)
|
||||
except queue.Full as e:
|
||||
if self._get_playback_state() != PlaybackState.PAUSED:
|
||||
raise e
|
||||
|
||||
completed_callback_event.wait()
|
||||
except queue.Full as e:
|
||||
self.logger.warning('Playback timeout: audio callback failed?')
|
||||
finally:
|
||||
if f and not f.closed:
|
||||
f.close()
|
||||
f = None
|
||||
|
||||
self.stop_playback(stream_index)
|
||||
|
||||
|
||||
@action
|
||||
def record(self, file=None, duration=None, device=None, sample_rate=None,
|
||||
blocksize=None, latency=0, channels=1, subtype='PCM_24'):
|
||||
"""
|
||||
Records audio to a sound file (support formats: wav, raw)
|
||||
|
||||
:param file: Sound file (default: the method will create a temporary file with the recording)
|
||||
:type file: str
|
||||
|
||||
:param duration: Recording duration in seconds (default: record until stop event)
|
||||
:type duration: float
|
||||
|
||||
:param device: Input device (default: default configured device or system default audio input if not configured)
|
||||
:type device: int or str
|
||||
|
||||
:param sample_rate: Recording sample rate (default: device default rate)
|
||||
:type sample_rate: int
|
||||
|
||||
:param blocksize: Audio block size (default: configured `input_blocksize` or 2048)
|
||||
:type blocksize: int
|
||||
|
||||
:param latency: Device latency in seconds (default: 0)
|
||||
:type latency: float
|
||||
|
||||
:param channels: Number of channels (default: 1)
|
||||
:type channels: int
|
||||
|
||||
:param subtype: Recording subtype - see `soundfile docs <https://pysoundfile.readthedocs.io/en/0.9.0/#soundfile.available_subtypes>`_ for a list of the available subtypes (default: PCM_24)
|
||||
:type subtype: str
|
||||
"""
|
||||
|
||||
import sounddevice as sd
|
||||
|
||||
self.recording_paused_changed.clear()
|
||||
|
||||
if file:
|
||||
file = os.path.abspath(os.path.expanduser(file))
|
||||
else:
|
||||
file = tempfile.mktemp(prefix='platypush_recording_', suffix='.wav',
|
||||
dir='')
|
||||
|
||||
if os.path.isfile(file):
|
||||
self.logger.info('Removing existing audio file {}'.format(file))
|
||||
os.unlink(file)
|
||||
|
||||
if device is None:
|
||||
device = self.input_device
|
||||
if device is None:
|
||||
device = self._get_default_device('input')
|
||||
|
||||
if sample_rate is None:
|
||||
dev_info = sd.query_devices(device, 'input')
|
||||
sample_rate = int(dev_info['default_samplerate'])
|
||||
|
||||
if blocksize is None:
|
||||
blocksize = self.input_blocksize
|
||||
|
||||
q = queue.Queue()
|
||||
|
||||
def audio_callback(indata, frames, time, status):
|
||||
while self._get_recording_state() == RecordingState.PAUSED:
|
||||
self.recording_paused_changed.wait()
|
||||
|
||||
if status:
|
||||
self.logger.warning('Recording callback status: {}'.format(
|
||||
str(status)))
|
||||
|
||||
q.put(indata.copy())
|
||||
|
||||
|
||||
try:
|
||||
import soundfile as sf
|
||||
import numpy
|
||||
|
||||
with sf.SoundFile(file, mode='x', samplerate=sample_rate,
|
||||
channels=channels, subtype=subtype) as f:
|
||||
with sd.InputStream(samplerate=sample_rate, device=device,
|
||||
channels=channels, callback=audio_callback,
|
||||
latency=latency, blocksize=blocksize):
|
||||
self.start_recording()
|
||||
self.logger.info('Started recording from device [{}] to [{}]'.
|
||||
format(device, file))
|
||||
|
||||
recording_started_time = time.time()
|
||||
|
||||
while self._get_recording_state() != RecordingState.STOPPED \
|
||||
and (duration is None or
|
||||
time.time() - recording_started_time < duration):
|
||||
while self._get_recording_state() == RecordingState.PAUSED:
|
||||
self.recording_paused_changed.wait()
|
||||
|
||||
get_args = {
|
||||
'block': True,
|
||||
'timeout': max(0, duration - (time.time() -
|
||||
recording_started_time))
|
||||
} if duration is not None else {}
|
||||
|
||||
data = q.get(**get_args)
|
||||
f.write(data)
|
||||
|
||||
f.flush()
|
||||
|
||||
except queue.Empty as e:
|
||||
self.logger.warning('Recording timeout: audio callback failed?')
|
||||
finally:
|
||||
self.stop_recording()
|
||||
|
||||
|
||||
@action
|
||||
def recordplay(self, duration=None, input_device=None, output_device=None,
|
||||
sample_rate=None, blocksize=None, latency=0, channels=1,
|
||||
dtype=None):
|
||||
"""
|
||||
Records audio and plays it on an output sound device (audio pass-through)
|
||||
|
||||
:param duration: Recording duration in seconds (default: record until stop event)
|
||||
:type duration: float
|
||||
|
||||
:param input_device: Input device (default: default configured device or system default audio input if not configured)
|
||||
:type input_device: int or str
|
||||
|
||||
:param output_device: Output device (default: default configured device or system default audio output if not configured)
|
||||
:type output_device: int or str
|
||||
|
||||
:param sample_rate: Recording sample rate (default: device default rate)
|
||||
:type sample_rate: int
|
||||
|
||||
:param blocksize: Audio block size (default: configured `output_blocksize` or 2048)
|
||||
:type blocksize: int
|
||||
|
||||
:param latency: Device latency in seconds (default: 0)
|
||||
:type latency: float
|
||||
|
||||
:param channels: Number of channels (default: 1)
|
||||
:type channels: int
|
||||
|
||||
:param dtype: Data type for the recording - see `soundfile docs <https://python-sounddevice.readthedocs.io/en/0.3.12/_modules/sounddevice.html#rec>`_ for available types (default: input device default)
|
||||
:type dtype: str
|
||||
"""
|
||||
|
||||
import sounddevice as sd
|
||||
|
||||
self.playback_paused_changed.clear()
|
||||
self.recording_paused_changed.clear()
|
||||
|
||||
if input_device is None:
|
||||
input_device = self.input_device
|
||||
if input_device is None:
|
||||
input_device = self._get_default_device('input')
|
||||
|
||||
if output_device is None:
|
||||
output_device = self.output_device
|
||||
if output_device is None:
|
||||
output_device = self._get_default_device('output')
|
||||
|
||||
if sample_rate is None:
|
||||
dev_info = sd.query_devices(input_device, 'input')
|
||||
sample_rate = int(dev_info['default_samplerate'])
|
||||
|
||||
if blocksize is None:
|
||||
blocksize = self.output_blocksize
|
||||
|
||||
def audio_callback(indata, outdata, frames, time, status):
|
||||
while self._get_recording_state() == RecordingState.PAUSED:
|
||||
self.recording_paused_changed.wait()
|
||||
|
||||
if status:
|
||||
self.logger.warning('Recording callback status: {}'.format(
|
||||
str(status)))
|
||||
|
||||
outdata[:] = indata
|
||||
|
||||
|
||||
stream_index = None
|
||||
|
||||
try:
|
||||
import soundfile as sf
|
||||
import numpy
|
||||
|
||||
stream = sd.Stream(samplerate=sample_rate, channels=channels,
|
||||
blocksize=blocksize, latency=latency,
|
||||
device=(input_device, output_device),
|
||||
dtype=dtype, callback=audio_callback)
|
||||
self.start_recording()
|
||||
stream_index = self.start_playback(stream)
|
||||
|
||||
self.logger.info('Started recording pass-through from device ' +
|
||||
'[{}] to device [{}]'.
|
||||
format(input_device, output_device))
|
||||
|
||||
recording_started_time = time.time()
|
||||
|
||||
while self._get_recording_state() != RecordingState.STOPPED \
|
||||
and (duration is None or
|
||||
time.time() - recording_started_time < duration):
|
||||
while self._get_recording_state() == RecordingState.PAUSED:
|
||||
self.recording_paused_changed.wait()
|
||||
|
||||
time.sleep(0.1)
|
||||
|
||||
except queue.Empty as e:
|
||||
self.logger.warning('Recording timeout: audio callback failed?')
|
||||
finally:
|
||||
self.stop_playback(stream_index)
|
||||
self.stop_recording()
|
||||
|
||||
|
||||
@action
|
||||
def get_active_streams(self):
|
||||
"""
|
||||
:returns: A list of active players
|
||||
"""
|
||||
return {
|
||||
i: {
|
||||
attr: getattr(stream, attr)
|
||||
for attr in ['active', 'closed', 'stopped', 'blocksize',
|
||||
'channels', 'cpu_load', 'device', 'dtype',
|
||||
'latency', 'samplerate', 'samplesize']
|
||||
if hasattr(stream, attr)
|
||||
} for i, stream in self.active_streams.items()
|
||||
}
|
||||
|
||||
|
||||
def start_playback(self, stream, completed_callback_event=None):
|
||||
stream_index = None
|
||||
with self.playback_state_lock:
|
||||
self.playback_state = PlaybackState.PLAYING
|
||||
stream_index = None
|
||||
for i in range(len(self.active_streams)+1):
|
||||
if i not in self.active_streams:
|
||||
stream_index = i
|
||||
break
|
||||
|
||||
self.active_streams[stream_index] = stream
|
||||
self.completed_callback_events[stream_index] = \
|
||||
completed_callback_event if completed_callback_event else Event()
|
||||
|
||||
self.logger.info('Playback started on stream index {}'.
|
||||
format(stream_index))
|
||||
return stream_index
|
||||
|
||||
@action
|
||||
def stop_playback(self, *streams):
|
||||
with self.playback_state_lock:
|
||||
if not streams:
|
||||
streams = self.active_streams.keys()
|
||||
updated_n_players = len(self.active_streams)
|
||||
completed_callback_events = {}
|
||||
|
||||
for i in streams:
|
||||
if i is None or not (i in self.active_streams):
|
||||
continue
|
||||
|
||||
stream = self.active_streams[i]
|
||||
updated_n_players -= 1
|
||||
if self.completed_callback_events[i]:
|
||||
completed_callback_events[i] = self.completed_callback_events[i]
|
||||
|
||||
if not updated_n_players:
|
||||
self.playback_state = PlaybackState.STOPPED
|
||||
|
||||
for i, event in completed_callback_events.items():
|
||||
event.wait()
|
||||
del self.completed_callback_events[i]
|
||||
del self.active_streams[i]
|
||||
|
||||
self.logger.info('Playback stopped')
|
||||
|
||||
@action
|
||||
def pause_playback(self):
|
||||
with self.playback_state_lock:
|
||||
if self.playback_state == PlaybackState.PAUSED:
|
||||
self.playback_state = PlaybackState.PLAYING
|
||||
elif self.playback_state == PlaybackState.PLAYING:
|
||||
self.playback_state = PlaybackState.PAUSED
|
||||
else:
|
||||
return
|
||||
|
||||
self.playback_paused_changed.set()
|
||||
self.logger.info('Playback ' + ('paused' if self.playback_state ==
|
||||
PlaybackState.PAUSED else 'playing'))
|
||||
|
||||
def start_recording(self):
|
||||
with self.recording_state_lock:
|
||||
self.recording_state = RecordingState.RECORDING
|
||||
|
||||
@action
|
||||
def stop_recording(self):
|
||||
with self.recording_state_lock:
|
||||
self.recording_state = RecordingState.STOPPED
|
||||
self.logger.info('Recording stopped')
|
||||
|
||||
@action
|
||||
def pause_recording(self):
|
||||
with self.recording_state_lock:
|
||||
if self.recording_state == RecordingState.PAUSED:
|
||||
self.recording_state = RecordingState.RECORDING
|
||||
elif self.recording_state == RecordingState.RECORDING:
|
||||
self.recording_state = RecordingState.PAUSED
|
||||
else:
|
||||
return
|
||||
|
||||
self.logger.info('Recording paused state toggled')
|
||||
self.recording_paused_changed.set()
|
||||
|
||||
def _get_playback_state(self):
|
||||
with self.playback_state_lock:
|
||||
return self.playback_state
|
||||
|
||||
def _get_recording_state(self):
|
||||
with self.recording_state_lock:
|
||||
return self.recording_state
|
||||
|
||||
@action
|
||||
def get_state(self):
|
||||
return {
|
||||
'playback_state': self._get_playback_state().name,
|
||||
'recording_state': self._get_recording_state().name,
|
||||
}
|
||||
|
||||
|
||||
# vim:sw=4:ts=4:et:
|
||||
Loading…
Reference in a new issue