src/ovtk_audiencekit/core/Audio.py

@@ -25,71 +25,59 @@ os.close(old_stderr)
 
 logger = logging.getLogger(__name__)
 
-
-def check_rate(index, channels, rate):
-    try:
-        return pyaudio.is_format_supported(rate,
-            output_channels=channels,
-            output_device=index,
-            output_format=pya.paFloat32)
-    except ValueError:
-        return False
-
-alt_rates = [44100, 48000]
 class Clip:
-    def __init__(self, path, output_index, buffer_length=2048, speed=1, force_stereo=True):
+    def __init__(self, path, samplerate=None, speed=1, force_stereo=True):
-        _raw, native_rate = librosa.load(path, sr=None, dtype='float32', mono=False)
+        self.path = path
-        self._channels = _raw.shape[0] if len(_raw.shape) == 2 else 1
+        raw, native_rate = librosa.load(self.path, sr=None, dtype='float32', mono=False)
-        if force_stereo and self._channels == 1:
-            _raw = np.resize(_raw, (2,*_raw.shape))
-            self._channels = 2
 
-        target_samplerate = native_rate
+        self.channels = raw.shape[0] if len(raw.shape) == 2 else 1
-        if not check_rate(output_index, self._channels , native_rate):
+        if force_stereo and self.channels == 1:
-            try:
+            raw = np.resize(raw, (2,*raw.shape))
-                target_samplerate = next((rate for rate in alt_rates if check_rate(output_index, self._channels , rate)))
+            self.channels = 2
-            except StopIteration:
-                logger.warn('Target audio device does not claim to support any sample rates! Attempting playback at native rate')
-        self._samplerate = target_samplerate
 
-        if native_rate != self._samplerate:
+        self.samplerate = samplerate or native_rate
-            _raw = librosa.resample(_raw, native_rate, self._samplerate, fix=True, scale=True)
+        if native_rate != self.samplerate:
+            raw = librosa.resample(raw, native_rate, self.samplerate, fix=True, scale=True)
 
-        self._raw = np.ascontiguousarray(self._stereo_transpose(_raw), dtype='float32')
+        self.raw = np.ascontiguousarray(self._stereo_transpose(raw), dtype='float32')
 
         if speed != 1:
             self.stretch(speed)
 
-        self._pos = 0
+    @property
-        self._playing = False
+    def length(self):
+        return self.raw.shape[0] / self.samplerate
+
+    def _stereo_transpose(self, ndata):
+        return ndata if self.channels == 1 else ndata.T
+
+    def stretch(self, speed):
+        stretched = tsm.wsola(self._stereo_transpose(self.raw), speed)
+        self.raw = np.ascontiguousarray(self._stereo_transpose(stretched), dtype='float32')
+
+    def save(self, filename):
+        soundfile.write(filename, self._stereo_transpose(self.raw), self.samplerate)
+
+
+class Stream:
+    def __init__(self, clip, output_index, buffer_length=4096):
+        self.clip = clip
+        self.pos = 0
+        self.playing = False
         self._end_event = AioEvent()
         self._stream = pyaudio.open(
             output_device_index=output_index,
             format=pya.paFloat32,
-            channels=self._channels,
+            channels=self.clip.channels,
-            rate=self._samplerate,
+            rate=self.clip.samplerate,
             frames_per_buffer=buffer_length,
             output=True,
             stream_callback=self._read_callback,
             start=False)
 
-    @property
-    def length(self):
-        return self._raw.shape[0] / self._samplerate
-
-    def _stereo_transpose(self, ndata):
-        return ndata if self._channels == 1 else ndata.T
-
-    def stretch(self, speed):
-        stretched = tsm.wsola(self._stereo_transpose(self._raw), speed)
-        self._raw = np.ascontiguousarray(self._stereo_transpose(stretched), dtype='float32')
-
-    def save(self, filename):
-        soundfile.write(filename, self._stereo_transpose(self._raw), self._samplerate)
-
     def _play(self):
-        self._playing = True
+        self.playing = True
-        self._pos = 0
+        self.pos = 0
 
         if not self._stream.is_active():
             self._stream.start_stream()
@@ -97,38 +85,48 @@ class Clip:
     def play(self):
         self._end_event.clear()
         self._play()
-        self._end_event.wait(timeout=self.length)
+        self._end_event.wait(timeout=self.clip.length)
 
     async def aplay(self):
         self._end_event.clear()
         self._play()
         try:
-            await self._end_event.coro_wait(timeout=self.length)
+            await self._end_event.coro_wait(timeout=self.clip.length)
         except asyncio.CancelledError:
-            self._playing = False
+            self.playing = False
             self._stream.stop_stream()
 
     def close(self):
         self._stream.close()
 
     def _read_callback(self, in_data, frame_count, time_info, status):
-        if self._channels > 1:
+        if self.clip.channels > 1:
-            buffer = np.zeros((frame_count, self._channels), dtype='float32')
+            buffer = np.zeros((frame_count, self.clip.channels), dtype='float32')
         else:
             buffer = np.zeros((frame_count,), dtype='float32')
 
-        if self._playing:
+        if self.playing:
-            newpos = self._pos + frame_count
+            newpos = self.pos + frame_count
-            clip_chunk = self._raw[self._pos:newpos]
+            clip_chunk = self.clip.raw[self.pos:newpos]
-            self._pos = newpos
+            self.pos = newpos
             buffer[0:clip_chunk.shape[0]] = clip_chunk
 
-            if self._pos >= self._raw.shape[0]:
+            if self.pos >= self.clip.raw.shape[0]:
-                self._playing = False
+                self.playing = False
                 self._end_event.set()
 
         return buffer, pya.paContinue
 
+    @staticmethod
+    def check_rate(index, channels, rate):
+        try:
+            return pyaudio.is_format_supported(rate,
+                output_channels=channels,
+                output_device=index,
+                output_format=pya.paFloat32)
+        except ValueError:
+            return False
+
     @staticmethod
     def find_output_index(output):
         if output is None:

src/ovtk_audiencekit/core/__init__.py

@@ -1,3 +1,3 @@
 from .WebsocketServerProcess import WebsocketServerProcess
 from .MainProcess import MainProcess
-from .Clip import Clip
+from .Audio import Clip, Stream

src/ovtk_audiencekit/plugins/AudioAlert/AudioAlert.py

@@ -1,47 +1,75 @@
 import asyncio
 from collections import deque
 
+import maya
+
 from ovtk_audiencekit.plugins import PluginBase
-from ovtk_audiencekit.core import Clip
+from ovtk_audiencekit.core import Clip, Stream
 
 class AudioAlert(PluginBase):
-    def setup(self, output=None, buffer_length=2048, cutoff_prevention_buffers=None):
+    def setup(self, output=None, timeout_min=1, sample_rate=None, buffer_length=4096, force_stereo=True):
-        if cutoff_prevention_buffers:
+        self.force_stereo = force_stereo
-            self.logger.info('`cutoff_prevention_buffers` are depricated')
+        self.timeout_min = timeout_min
+        self.clips = {}
+        self.streams = {}
+        self.buffer_length = int(buffer_length)
+        self.output_index = Stream.find_output_index(output)
+        if sample_rate is None:
+            try:
+                sample_rate = next((rate for rate in [44100, 48000] if Stream.check_rate(self.output_index, 1, rate)))
+            except StopIteration:
+                self.logger.warn('Target audio device does not claim to support common sample rates! Attempting playback at native rate of audio')
 
-        self.sounds = {}
+        self._cleanup_task = asyncio.create_task(self._cleanup())
-        self._buffer_length = int(buffer_length)
-        self._output_index = Clip.find_output_index(output)
 
     def run(self, path, speed=1, immediate=True, poly=1, **kwargs):
-        sound = None
-
-        if poly != 1:
         poly = int(poly)
-            sound_dq = self.sounds.get(path)
+        key = f'{path}@{speed}x'
-            if sound_dq is None or type(sound_dq) != deque or sound_dq.maxlen != poly:
+        clip = self.clips.get(key, [None, None])[0]
-                sound_dq = deque(maxlen=poly)
+
-                self.sounds[path] = sound_dq
+        if clip is None:
-            if len(sound_dq) != poly:
+            clip = Clip(path, speed=speed, force_stereo=self.force_stereo)
-                self.logger.debug("filling", len(sound_dq), poly, sound_dq)
+            self.clips[key] = [clip, maya.now()]
-                sound = Clip(path,
-                            self._output_index,
-                            buffer_length=self._buffer_length,
-                            speed=speed)
-                sound_dq.append(sound)
         else:
-                self.logger.debug("rotate", len(sound_dq), poly, sound_dq)
+            self.clips[key][1] = maya.now()
-                sound_dq.rotate(1)
+
-                sound = sound_dq[0]
+        stream_dq, refs = self.streams.get(path, (None, set()))
+        if stream_dq is None:
+            stream_dq = deque(maxlen=poly)
+            self.streams[path] = (stream_dq, refs)
+        refs.add(key)
+
+        if stream_dq.maxlen != poly:
+            self.logger.warn('Cannot change poly while streams are active!')
+
+        if len(stream_dq) != poly:
+            stream = Stream(clip, self.output_index,
+                            buffer_length=self.buffer_length)
+            stream_dq.append(stream)
         else:
-            if self.sounds.get(path) is None:
+            stream_dq.rotate(1)
-                self.sounds[path] = Clip(path,
+            stream = stream_dq[0]
-                                         self._output_index,
-                                         buffer_length=self._buffer_length,
-                                         speed=speed)
-            sound = self.sounds.get(path)
 
         if immediate:
-            asyncio.create_task(sound.aplay())
+            asyncio.create_task(stream.aplay())
         else:
-            sound.play()
+            stream.play()
+
+    async def _cleanup(self):
+        while True:
+            await asyncio.sleep(60)
+            now = maya.now()
+            for key, [clip, last_used] in list(self.clips.items()):
+                if now >= last_used.add(minutes=self.timeout_min, seconds=clip.length):
+                    del self.clips[key]
+                    self.logger.debug(f'Dropping {key}')
+
+                    streams, refs = self.streams.get(clip.path, (None, None))
+                    if refs:
+                        refs.remove(key)
+                        self.logger.debug(f'Stream {clip.path} now refs {refs}')
+                        if len(refs) == 0:
+                            self.logger.debug('Closing streams...')
+                            for stream in streams:
+                                stream.close()
+                            del self.streams[clip.path]

src/ovtk_audiencekit/plugins/TTS/TTS.py

@@ -8,7 +8,7 @@ from TTS.config import load_config
 
 from ovtk_audiencekit.plugins import PluginBase
 from ovtk_audiencekit.events import Message, SysMessage
-from ovtk_audiencekit.core import Clip
+from ovtk_audiencekit.core import Clip, Stream
 from ovtk_audiencekit.core.Data import CACHE_DIR
 
 
@@ -18,12 +18,12 @@ class TextToSpeechPlugin(PluginBase):
 
         self.speaker_wav = speaker_wav
 
-        self._output_index = Clip.find_output_index(output)
+        self.output_index = Stream.find_output_index(output)
 
         conf_overrides = {k[2:]: v for k, v in kwargs.items() if k.startswith('o_')}
 
-        self._cache = os.path.join(CACHE_DIR, 'tts')
+        self.cache_dir = os.path.join(CACHE_DIR, 'tts')
-        os.makedirs(os.path.dirname(self._cache), exist_ok=True)
+        os.makedirs(os.path.dirname(self.cache_dir), exist_ok=True)
 
         self.cuda = cuda
 
@@ -36,7 +36,7 @@ class TextToSpeechPlugin(PluginBase):
             vocoder_path, vocoder_config_path = None, None
 
         if conf_overrides:
-            override_conf_path = os.path.join(self._cache, f'{self._name}_override.json')
+            override_conf_path = os.path.join(self.cache_dir, f'{self._name}_override.json')
 
             config = load_config(config_path)
             for key, value in conf_overrides.items():
@@ -55,7 +55,7 @@ class TextToSpeechPlugin(PluginBase):
 
     def make_tts_wav(self, text, filename=None):
         if filename is None:
-            filename = os.path.join(self._cache, f'{uuid.uuid1()}.wav')
+            filename = os.path.join(self.cache_dir, f'{uuid.uuid1()}.wav')
 
         if self.speaker_wav:
             wav = self.synthesizer.tts(text, None, 'en', self.speaker_wav)
@@ -72,17 +72,20 @@ class TextToSpeechPlugin(PluginBase):
                 text += '.'
             filename = self.make_tts_wav(text)
             # TODO: Play direct from memory
-            clip = Clip(filename, self._output_index, force_stereo=False)
+            clip = Clip(filename, force_stereo=True)
+            stream = Stream(clip, self.output_index)
             if wait:
                 async def play():
-                    await clip.aplay()
+                    await stream.aplay()
-                    clip.close()
+                    stream.close()
+                    os.remove(os.path.join(self.cache_dir, filename))
                 asyncio.create_task(play())
             else:
-                clip.play()
+                stream.play()
-                clip.close()
+                stream.close()
+                os.remove(os.path.join(self.cache_dir, filename))
         except Exception as e:
-            print(e)
+            self.logger.error(f"Failed to make speech from input: {e}")
             if source_event := _ctx.get('event'):
                 msg = SysMessage(self._name, 'Failed to make speech from input!!')