Sound plugin moved back to __init__ for sake of better readthedocs generation

This commit is contained in:
Fabio Manganiello 2018-12-27 17:26:16 +01:00
parent 42236ac197
commit debf9a017b
2 changed files with 763 additions and 772 deletions

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@ -2,8 +2,769 @@
.. moduleauthor:: Fabio Manganiello <blacklight86@gmail.com>
"""
from .core import Sound
from .plugin import SoundPlugin
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, Mix
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, *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
"""
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_state = {}
self.playback_state_lock = RLock()
self.playback_paused_changed = {}
self.stream_mixes = {}
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, stream_index):
import sounddevice as sd
is_raw_stream = streamtype == sd.RawOutputStream
def audio_callback(outdata, frames, time, status):
if self._get_playback_state(stream_index) == PlaybackState.STOPPED:
raise sd.CallbackStop
while self._get_playback_state(stream_index) == PlaybackState.PAUSED:
self.playback_paused_changed[stream_index].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?')
return
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=None, samplerate=None, channels=None,
stream_index=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 frames for audio
files, 2 frames for synth sounds)
: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
:param stream_index: If specified, play to an already active stream
index (you can get them through
:method:`platypush.plugins.sound.query_streams`). Default:
creates a new audio stream through PortAudio.
"""
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
if bufsize is None:
if file:
bufsize = Sound._DEFAULT_FILE_BUFSIZE
else:
bufsize = Sound._DEFAULT_SYNTH_BUFSIZE
q = queue.Queue(maxsize=bufsize)
f = None
t = 0.
is_new_stream = stream_index is None or \
self.active_streams.get(stream_index) is None
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')
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
if is_new_stream:
stream_index = self._allocate_stream_index()
self.logger.info(('Starting playback of {} to sound device [{}] ' +
'on stream [{}]').format(
file or sound, device, stream_index))
if sound:
mix = self.stream_mixes[stream_index]
mix.add(sound)
if not is_new_stream:
return # Let the existing callback handle the new mix
try:
# Audio queue pre-fill loop
for _ in range(bufsize):
if f:
data = f.buffer_read(blocksize, dtype='float32')
if not data:
break
else:
duration = mix.duration()
blocktime = float(blocksize / samplerate)
next_t = min(t+blocktime, duration) \
if duration is not None else t+blocktime
data = mix.get_wave(t_start=t, t_end=next_t,
samplerate=samplerate)
t = next_t
if duration is not None and t >= duration:
break
q.put_nowait(data) # Pre-fill the audio queue
stream = self.active_streams[stream_index]
completed_callback_event = self.completed_callback_events[stream_index]
if stream is None:
streamtype = sd.RawOutputStream if file else sd.OutputStream
stream = streamtype(samplerate=samplerate, blocksize=blocksize,
device=device, channels=channels,
dtype='float32',
callback=self._play_audio_callback(
q=q, blocksize=blocksize,
streamtype=streamtype,
stream_index=stream_index),
finished_callback=completed_callback_event.set)
self._start_playback(stream_index=stream_index, stream=stream)
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(stream_index) == \
PlaybackState.PAUSED:
self.playback_paused_changed[stream_index].wait()
if f:
data = f.buffer_read(blocksize, dtype='float32')
if not data:
break
else:
duration = mix.duration()
blocktime = float(blocksize / samplerate)
next_t = min(t+blocktime, duration) \
if duration is not None else t+blocktime
data = mix.get_wave(t_start=t, t_end=next_t,
samplerate=samplerate)
t = next_t
if duration is not None and t >= duration:
break
if self._get_playback_state(stream_index) == \
PlaybackState.STOPPED:
break
try:
q.put(data, timeout=timeout)
except queue.Full as e:
if self._get_playback_state(stream_index) != \
PlaybackState.PAUSED:
raise e
completed_callback_event.wait()
except queue.Full as e:
if stream_index is None or \
self._get_playback_state(stream_index) != PlaybackState.STOPPED:
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.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_index = self._allocate_stream_index()
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()
self._start_playback(stream_index=stream_index,
stream=stream)
self.logger.info('Started recording pass-through from device ' +
'[{}] to sound 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 query_streams(self):
"""
:returns: A list of active audio streams
"""
streams = {
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()
}
for i, stream in streams.items():
stream['playback_state'] = self.playback_state[i].name
if i in self.stream_mixes:
stream['mix'] = { j: sound for j, sound in
enumerate(list(self.stream_mixes[i])) }
return streams
def _allocate_stream_index(self, completed_callback_event=None):
stream_index = None
with self.playback_state_lock:
for i in range(len(self.active_streams)+1):
if i not in self.active_streams:
stream_index = i
break
if stream_index is None:
raise RuntimeError('No stream index available')
self.active_streams[stream_index] = None
self.stream_mixes[stream_index] = Mix()
self.completed_callback_events[stream_index] = \
completed_callback_event if completed_callback_event else Event()
return stream_index
def _start_playback(self, stream_index, stream):
with self.playback_state_lock:
self.playback_state[stream_index] = PlaybackState.PLAYING
self.active_streams[stream_index] = stream
if isinstance(self.playback_paused_changed.get(stream_index), Event):
self.playback_paused_changed[stream_index].clear()
else:
self.playback_paused_changed[stream_index] = Event()
self.logger.info('Playback started on stream index {}'.
format(stream_index))
return stream_index
@action
def stop_playback(self, streams=None):
"""
:param streams: Streams to stop by index (default: all)
:type streams: list[int]
"""
with self.playback_state_lock:
streams = streams or self.active_streams.keys()
if not streams:
return
completed_callback_events = {}
for i in streams:
if i is None or not (i in self.active_streams):
continue
stream = self.active_streams[i]
if self.completed_callback_events[i]:
completed_callback_events[i] = self.completed_callback_events[i]
self.playback_state[i] = PlaybackState.STOPPED
for i, event in completed_callback_events.items():
event.wait()
del self.completed_callback_events[i]
del self.active_streams[i]
del self.stream_mixes[i]
self.logger.info('Playback stopped on streams [{}]'.format(
', '.join([str(stream) for stream in
completed_callback_events.keys()])))
@action
def pause_playback(self, streams=None):
"""
:param streams: Streams to pause by index (default: all)
:type streams: list[int]
"""
with self.playback_state_lock:
streams = streams or self.active_streams.keys()
if not streams:
return
for i in streams:
if i is None or not (i in self.active_streams):
continue
stream = self.active_streams[i]
if self.playback_state[i] == PlaybackState.PAUSED:
self.playback_state[i] = PlaybackState.PLAYING
elif self.playback_state[i] == PlaybackState.PLAYING:
self.playback_state[i] = PlaybackState.PAUSED
else:
continue
self.playback_paused_changed[i].set()
self.logger.info('Playback pause toggled on streams [{}]'.format(
', '.join([str(stream) for stream in streams])))
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()
@action
def release(self, stream_index=None,
sound_index=None, midi_note=None, frequency=None):
"""
Remove a sound from an active stream, either by sound index (use
:method:`platypush.sound.plugin.SoundPlugin.query_streams` to get
the sounds playing on the active streams), midi_note, frequency
or absolute file path.
:param stream_index: Stream index (default: sound removed from all the
active streams)
:type stream_index: int
:param sound_index: Sound index
:type sound_index: int
:param midi_note: MIDI note
:type midi_note: int
:param frequency: Sound frequency
:type frequency: float
"""
if sound_index is None and midi_note is None and frequency is None:
raise RuntimeError('Please specify either a sound index, ' +
'midi_note or frequency to release')
mixes = {
i: mix for i, mix in self.stream_mixes.items()
} if stream_index is None else {
stream_index: self.stream_mixes[stream_index]
}
streams_to_stop = []
for i, mix in mixes.items():
for j, sound in enumerate(mix):
if (sound_index is not None and j == sound_index) or \
(midi_note is not None
and sound.get('midi_note') == midi_note) or \
(frequency is not None
and sound.get('frequency') == frequency):
if len(list(mix)) == 1:
# Last sound in the mix
streams_to_stop.append(i)
else:
mix.remove(j)
if streams_to_stop:
self.stop_playback(streams_to_stop)
def _get_playback_state(self, stream_index):
with self.playback_state_lock:
return self.playback_state[stream_index]
def _get_recording_state(self):
with self.recording_state_lock:
return self.recording_state
# vim:sw=4:ts=4:et:

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@ -1,770 +0,0 @@
"""
.. 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, Mix
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, *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
"""
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_state = {}
self.playback_state_lock = RLock()
self.playback_paused_changed = {}
self.stream_mixes = {}
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, stream_index):
import sounddevice as sd
is_raw_stream = streamtype == sd.RawOutputStream
def audio_callback(outdata, frames, time, status):
if self._get_playback_state(stream_index) == PlaybackState.STOPPED:
raise sd.CallbackStop
while self._get_playback_state(stream_index) == PlaybackState.PAUSED:
self.playback_paused_changed[stream_index].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?')
return
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=None, samplerate=None, channels=None,
stream_index=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 frames for audio
files, 2 frames for synth sounds)
: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
:param stream_index: If specified, play to an already active stream
index (you can get them through
:method:`platypush.plugins.sound.query_streams`). Default:
creates a new audio stream through PortAudio.
"""
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
if bufsize is None:
if file:
bufsize = Sound._DEFAULT_FILE_BUFSIZE
else:
bufsize = Sound._DEFAULT_SYNTH_BUFSIZE
q = queue.Queue(maxsize=bufsize)
f = None
t = 0.
is_new_stream = stream_index is None or \
self.active_streams.get(stream_index) is None
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')
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
if is_new_stream:
stream_index = self._allocate_stream_index()
self.logger.info(('Starting playback of {} to sound device [{}] ' +
'on stream [{}]').format(
file or sound, device, stream_index))
if sound:
mix = self.stream_mixes[stream_index]
mix.add(sound)
if not is_new_stream:
return # Let the existing callback handle the new mix
try:
# Audio queue pre-fill loop
for _ in range(bufsize):
if f:
data = f.buffer_read(blocksize, dtype='float32')
if not data:
break
else:
duration = mix.duration()
blocktime = float(blocksize / samplerate)
next_t = min(t+blocktime, duration) \
if duration is not None else t+blocktime
data = mix.get_wave(t_start=t, t_end=next_t,
samplerate=samplerate)
t = next_t
if duration is not None and t >= duration:
break
q.put_nowait(data) # Pre-fill the audio queue
stream = self.active_streams[stream_index]
completed_callback_event = self.completed_callback_events[stream_index]
if stream is None:
streamtype = sd.RawOutputStream if file else sd.OutputStream
stream = streamtype(samplerate=samplerate, blocksize=blocksize,
device=device, channels=channels,
dtype='float32',
callback=self._play_audio_callback(
q=q, blocksize=blocksize,
streamtype=streamtype,
stream_index=stream_index),
finished_callback=completed_callback_event.set)
self._start_playback(stream_index=stream_index, stream=stream)
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(stream_index) == \
PlaybackState.PAUSED:
self.playback_paused_changed[stream_index].wait()
if f:
data = f.buffer_read(blocksize, dtype='float32')
if not data:
break
else:
duration = mix.duration()
blocktime = float(blocksize / samplerate)
next_t = min(t+blocktime, duration) \
if duration is not None else t+blocktime
data = mix.get_wave(t_start=t, t_end=next_t,
samplerate=samplerate)
t = next_t
if duration is not None and t >= duration:
break
if self._get_playback_state(stream_index) == \
PlaybackState.STOPPED:
break
try:
q.put(data, timeout=timeout)
except queue.Full as e:
if self._get_playback_state(stream_index) != \
PlaybackState.PAUSED:
raise e
completed_callback_event.wait()
except queue.Full as e:
if stream_index is None or \
self._get_playback_state(stream_index) != PlaybackState.STOPPED:
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.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_index = self._allocate_stream_index()
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()
self._start_playback(stream_index=stream_index,
stream=stream)
self.logger.info('Started recording pass-through from device ' +
'[{}] to sound 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 query_streams(self):
"""
:returns: A list of active audio streams
"""
streams = {
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()
}
for i, stream in streams.items():
stream['playback_state'] = self.playback_state[i].name
if i in self.stream_mixes:
stream['mix'] = { j: sound for j, sound in
enumerate(list(self.stream_mixes[i])) }
return streams
def _allocate_stream_index(self, completed_callback_event=None):
stream_index = None
with self.playback_state_lock:
for i in range(len(self.active_streams)+1):
if i not in self.active_streams:
stream_index = i
break
if stream_index is None:
raise RuntimeError('No stream index available')
self.active_streams[stream_index] = None
self.stream_mixes[stream_index] = Mix()
self.completed_callback_events[stream_index] = \
completed_callback_event if completed_callback_event else Event()
return stream_index
def _start_playback(self, stream_index, stream):
with self.playback_state_lock:
self.playback_state[stream_index] = PlaybackState.PLAYING
self.active_streams[stream_index] = stream
if isinstance(self.playback_paused_changed.get(stream_index), Event):
self.playback_paused_changed[stream_index].clear()
else:
self.playback_paused_changed[stream_index] = Event()
self.logger.info('Playback started on stream index {}'.
format(stream_index))
return stream_index
@action
def stop_playback(self, streams=None):
"""
:param streams: Streams to stop by index (default: all)
:type streams: list[int]
"""
with self.playback_state_lock:
streams = streams or self.active_streams.keys()
if not streams:
return
completed_callback_events = {}
for i in streams:
if i is None or not (i in self.active_streams):
continue
stream = self.active_streams[i]
if self.completed_callback_events[i]:
completed_callback_events[i] = self.completed_callback_events[i]
self.playback_state[i] = PlaybackState.STOPPED
for i, event in completed_callback_events.items():
event.wait()
del self.completed_callback_events[i]
del self.active_streams[i]
del self.stream_mixes[i]
self.logger.info('Playback stopped on streams [{}]'.format(
', '.join([str(stream) for stream in
completed_callback_events.keys()])))
@action
def pause_playback(self, streams=None):
"""
:param streams: Streams to pause by index (default: all)
:type streams: list[int]
"""
with self.playback_state_lock:
streams = streams or self.active_streams.keys()
if not streams:
return
for i in streams:
if i is None or not (i in self.active_streams):
continue
stream = self.active_streams[i]
if self.playback_state[i] == PlaybackState.PAUSED:
self.playback_state[i] = PlaybackState.PLAYING
elif self.playback_state[i] == PlaybackState.PLAYING:
self.playback_state[i] = PlaybackState.PAUSED
else:
continue
self.playback_paused_changed[i].set()
self.logger.info('Playback pause toggled on streams [{}]'.format(
', '.join([str(stream) for stream in streams])))
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()
@action
def release(self, stream_index=None,
sound_index=None, midi_note=None, frequency=None):
"""
Remove a sound from an active stream, either by sound index (use
:method:`platypush.sound.plugin.SoundPlugin.query_streams` to get
the sounds playing on the active streams), midi_note, frequency
or absolute file path.
:param stream_index: Stream index (default: sound removed from all the
active streams)
:type stream_index: int
:param sound_index: Sound index
:type sound_index: int
:param midi_note: MIDI note
:type midi_note: int
:param frequency: Sound frequency
:type frequency: float
"""
if sound_index is None and midi_note is None and frequency is None:
raise RuntimeError('Please specify either a sound index, ' +
'midi_note or frequency to release')
mixes = {
i: mix for i, mix in self.stream_mixes.items()
} if stream_index is None else {
stream_index: self.stream_mixes[stream_index]
}
streams_to_stop = []
for i, mix in mixes.items():
for j, sound in enumerate(mix):
if (sound_index is not None and j == sound_index) or \
(midi_note is not None
and sound.get('midi_note') == midi_note) or \
(frequency is not None
and sound.get('frequency') == frequency):
if len(list(mix)) == 1:
# Last sound in the mix
streams_to_stop.append(i)
else:
mix.remove(j)
if streams_to_stop:
self.stop_playback(streams_to_stop)
def _get_playback_state(self, stream_index):
with self.playback_state_lock:
return self.playback_state[stream_index]
def _get_recording_state(self):
with self.recording_state_lock:
return self.recording_state
# vim:sw=4:ts=4:et: