Merged sound.stream_recording and sound.record.

This commit is contained in:
Fabio Manganiello 2023-06-12 13:06:02 +02:00
parent a415c5b231
commit be794316a8
Signed by: blacklight
GPG key ID: D90FBA7F76362774
2 changed files with 20 additions and 184 deletions

View file

@ -32,7 +32,7 @@ class SoundRoute(StreamingRoute):
@contextmanager
def _audio_stream(self, **kwargs) -> Generator[None, None, None]:
response = send_request(
'sound.stream_recording',
'sound.record',
dtype='int16',
**kwargs,
)

View file

@ -1,8 +1,8 @@
import os
import queue
import stat
import tempfile
import time
import warnings
from enum import Enum
from threading import Thread, Event, RLock
@ -438,7 +438,20 @@ class SoundPlugin(Plugin):
self.stop_playback([stream_index])
@action
def stream_recording(
def stream_recording(self, *args, **kwargs):
"""
Deprecated alias for :meth:`.record`.
"""
warnings.warn(
'sound.stream_recording is deprecated, use sound.record instead',
DeprecationWarning,
stacklevel=1,
)
return self.record(*args, **kwargs)
@action
def record(
self,
device: Optional[str] = None,
fifo: Optional[str] = None,
@ -451,6 +464,7 @@ class SoundPlugin(Plugin):
channels: int = 1,
redis_queue: Optional[str] = None,
format: str = 'wav',
stream: bool = True,
):
"""
Return audio data from an audio source
@ -474,6 +488,8 @@ class SoundPlugin(Plugin):
:param redis_queue: If set, the audio chunks will also be published to
this Redis channel, so other consumers can process them downstream.
:param format: Audio format. Supported: wav, mp3, ogg, aac. Default: wav.
:param stream: If True (default), then the audio will be streamed to an
HTTP endpoint too (default: ``/sound/stream<.format>``).
"""
self.recording_paused_changed.clear()
@ -563,7 +579,7 @@ class SoundPlugin(Plugin):
continue
f.write(data)
if redis_queue:
if redis_queue and stream:
get_redis().publish(redis_queue, data)
except queue.Empty:
@ -574,186 +590,6 @@ class SoundPlugin(Plugin):
Thread(target=streaming_thread).start()
@action
def record(
self,
outfile=None,
duration=None,
device=None,
sample_rate=None,
format=None,
blocksize=None,
latency='high',
channels=1,
subtype='PCM_24',
):
"""
Records audio to a sound file (support formats: wav, raw)
:param outfile: Sound file (default: the method will create a temporary file with the recording)
:type outfile: 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 format: Audio format (default: WAV)
:type format: str
:param blocksize: Audio block size (default: configured `input_blocksize` or 2048)
:type blocksize: int
:param latency: Device latency in seconds (default: the device's default high latency)
:type latency: float
:param channels: Number of channels (default: 1)
:type channels: int
:param subtype: Recording subtype - see `Soundfile docs - Subtypes
<https://pysoundfile.readthedocs.io/en/0.9.0/#soundfile.available_subtypes>`_
for a list of the available subtypes (default: PCM_24)
:type subtype: str
"""
def recording_thread(
outfile,
duration,
device,
sample_rate,
format,
blocksize,
latency,
channels,
subtype,
):
self.recording_paused_changed.clear()
if outfile:
outfile = os.path.abspath(os.path.expanduser(outfile))
if os.path.isfile(outfile):
self.logger.info('Removing existing audio file %s', outfile)
os.unlink(outfile)
else:
outfile = tempfile.NamedTemporaryFile(
prefix='recording_',
suffix='.wav',
delete=False,
dir=tempfile.gettempdir(),
).name
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, duration, status):
while self._get_recording_state() == RecordingState.PAUSED:
self.recording_paused_changed.wait()
if status:
self.logger.warning('Recording callback status: %s', status)
q.put(
{
'timestamp': time.time(),
'frames': frames,
'time': duration,
'data': indata.copy(),
}
)
try:
with sf.SoundFile(
outfile,
mode='w',
samplerate=sample_rate,
format=format,
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()
get_bus().post(SoundRecordingStartedEvent(filename=outfile))
self.logger.info(
'Started recording from device [%s] to [%s]',
device,
outfile,
)
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)
if data and time.time() - data.get('timestamp') <= 1.0:
# Only write the block if the latency is still acceptable
f.write(data['data'])
f.flush()
except queue.Empty:
self.logger.warning('Recording timeout: audio callback failed?')
finally:
self.stop_recording()
get_bus().post(SoundRecordingStoppedEvent(filename=outfile))
Thread(
target=recording_thread,
args=(
outfile,
duration,
device,
sample_rate,
format,
blocksize,
latency,
channels,
subtype,
),
).start()
@action
def recordplay(
self,