server playback
This commit is contained in:
@ -17,9 +17,9 @@
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"endTime": 30,
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"filename": "C:\\Users\\mickl\\Desktop\\cliptrim-ui\\ClipTrimApp\\audio-service\\recordings\\audio_capture_20260220_193822.wav",
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"name": "Pee pee\npoo poo",
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"playbackType": "playStop",
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"startTime": 27.587412587412587,
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"volume": 1
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"playbackType": "playOverlap",
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"startTime": 27.76674010920584,
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"volume": 0.25
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},
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{
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"endTime": 27.516843118383072,
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@ -8,10 +8,10 @@
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"save_path": "C:\\Users\\mickl\\Desktop\\cliptrim-ui\\ClipTrimApp\\audio-service\\recordings",
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"recording_length": 30,
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"output_device": {
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"channels": 2,
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"default_samplerate": 48000,
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"index": 40,
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"max_output_channels": 2,
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"name": "Speakers (Realtek(R) Audio)"
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"index": 45,
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"name": "VM to Discord (VB-Audio Voicemeeter VAIO)"
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},
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"http_port": 5010
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}
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BIN
audio-service/src/__pycache__/audio_clip.cpython-313.pyc
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audio-service/src/__pycache__/audio_clip.cpython-313.pyc
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audio-service/src/__pycache__/audio_io.cpython-313.pyc
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audio-service/src/__pycache__/audio_io.cpython-313.pyc
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64
audio-service/src/audio_clip.py
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64
audio-service/src/audio_clip.py
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@ -0,0 +1,64 @@
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import scipy.signal
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import scipy.io.wavfile as wavfile
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import numpy as np
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import os
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class AudioClip:
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def __init__(self, metadata, target_sample_rate=44100):
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"""
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metadata: dict with keys 'filename', 'start', 'end' (seconds)
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target_sample_rate: sample rate for playback
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"""
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self.metadata = metadata
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self.file_path = metadata['filename']
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self.start = metadata.get('startTime', 0)
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self.end = metadata.get('endTime', None)
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self.target_sample_rate = target_sample_rate
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self.volume = metadata.get('volume', 1.0)
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self.finished = False
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self.audio_data, self.sample_rate = self._load_and_process_audio()
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print(f"AudioClip created for {self.file_path} with start={self.start}s, end={self.end}s, sample_rate={self.sample_rate}Hz, length={len(self.audio_data)/self.sample_rate:.2f}s")
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self.position = 0 # sample index for playback
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def _load_and_process_audio(self):
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# Load audio file
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sample_rate, data = wavfile.read(self.file_path)
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# Convert to float32
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if data.dtype != np.float32:
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data = data.astype(np.float32) / np.max(np.abs(data))
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# Convert to mono if needed
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if len(data.shape) > 1:
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data = np.mean(data, axis=1)
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# Resample if needed
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if sample_rate != self.target_sample_rate:
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num_samples = int(len(data) * self.target_sample_rate / sample_rate)
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data = scipy.signal.resample(data, num_samples)
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sample_rate = self.target_sample_rate
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# Cache only the clip region
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start_sample = int(self.start * sample_rate)
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end_sample = int(self.end * sample_rate) if self.end else len(data)
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cached = data[start_sample:end_sample]
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cached *= self.volume # Apply volume
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return cached, sample_rate
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def get_samples(self, num_samples):
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# Return next chunk for playback
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if self.position >= len(self.audio_data):
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self.finished = True
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return np.zeros(num_samples, dtype=np.float32)
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end_pos = min(self.position + num_samples, len(self.audio_data))
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chunk = self.audio_data[self.position:end_pos]
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self.position = end_pos
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if self.position >= len(self.audio_data):
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self.finished = True
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# Pad if chunk is short
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if len(chunk) < num_samples:
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chunk = np.pad(chunk, (0, num_samples - len(chunk)), mode='constant')
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return chunk
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def is_finished(self):
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return self.finished
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def reset(self):
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self.position = 0
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self.finished = False
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166
audio-service/src/audio_io.py
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166
audio-service/src/audio_io.py
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@ -0,0 +1,166 @@
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import sounddevice as sd
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import numpy as np
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import os
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from datetime import datetime
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import scipy.io.wavfile as wavfile
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from metadata_manager import MetaDataManager
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from audio_clip import AudioClip
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# AudioClip class for clip playback
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class AudioIO:
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_instance = None
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def __new__(cls, *args, **kwargs):
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if cls._instance is None:
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# print("Creating new AudioRecorder instance")
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cls._instance = super().__new__(cls)
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cls._instance.init()
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return cls._instance
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def init(self):
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self.duration = 30
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self.channels = 2
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self.input_sample_rate = 44100
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self.output_sample_rate = 44100
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self.buffer = np.zeros((int(self.duration * self.input_sample_rate), self.channels), dtype=np.float32)
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self.recordings_dir = "recordings"
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sd.default.latency = 'low'
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self.in_stream = sd.InputStream(
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callback=self.record_callback
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)
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self.out_stream = sd.OutputStream(
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callback=self.playback_callback,
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latency=3
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)
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self.clip_map = {}
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def refresh_streams(self):
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was_active = self.in_stream.active
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if was_active:
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self.in_stream.stop()
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self.out_stream.stop()
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self.buffer = np.zeros((int(self.duration * self.input_sample_rate), self.channels), dtype=np.float32)
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# print(f"AudioRecorder initialized with duration={self.duration}s, sample_rate={self.sample_rate}Hz, channels={self.channels}")
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self.in_stream = sd.InputStream(
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callback=self.record_callback
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)
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self.out_stream = sd.OutputStream(
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callback=self.playback_callback
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)
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if was_active:
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self.in_stream.start()
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self.out_stream.start()
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def record_callback(self, indata, frames, time, status):
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if status:
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# print(f"Recording status: {status}")
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pass
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# Circular buffer implementation
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self.buffer = np.roll(self.buffer, -frames, axis=0)
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self.buffer[-frames:] = indata
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def playback_callback(self, outdata, frames, time, status):
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if status:
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# print(f"Playback status: {status}")
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pass
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outdata.fill(0)
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# Iterate over a copy of the items to avoid modifying the dictionary during iteration
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for clip_id, clip_list in list(self.clip_map.items()):
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for clip in clip_list[:]: # Iterate over a copy of the list
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if not clip.is_finished():
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samples = clip.get_samples(frames)
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outdata[:] += samples.reshape(-1, 1) # Mix into output
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if clip.is_finished():
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self.clip_map[clip_id].remove(clip)
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if len(self.clip_map[clip_id]) == 0:
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del self.clip_map[clip_id]
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break # Exit inner loop since the key is deleted
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def save_last_n_seconds(self):
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# Create output directory if it doesn't exist
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os.makedirs(self.recordings_dir, exist_ok=True)
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# Generate filename with timestamp
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timestamp = datetime.now().strftime("%Y%m%d_%H%M%S")
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filename = os.path.join(self.recordings_dir, f"audio_capture_{timestamp}.wav")
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# Normalize audio to prevent clipping
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audio_data = self.buffer / np.max(np.abs(self.buffer)) * .5
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# Convert float32 to int16 for WAV file
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audio_data_int16 = (audio_data * 32767).astype(np.int16)
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# Write buffer to file
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wavfile.write(filename, int(self.input_sample_rate), audio_data_int16)
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meta = MetaDataManager()
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clip_metadata = {
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"filename": filename,
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"name": f"Clip {timestamp}",
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"playbackType":"playStop",
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"volume": 1.0,
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}
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meta.add_clip_to_collection("Uncategorized", clip_metadata )
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return clip_metadata
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def set_buffer_duration(self, duration):
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self.duration = duration
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self.buffer = np.zeros((int(duration * self.input_sample_rate), self.channels), dtype=np.float32)
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def set_recording_directory(self, directory):
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self.recordings_dir = directory
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def start_recording(self):
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if(self.in_stream.active):
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# print("Already recording")
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return
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# print('number of channels', self.channels)
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self.in_stream.start()
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self.out_stream.start()
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self.output_sample_rate = self.out_stream.samplerate
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self.input_sample_rate = self.in_stream.samplerate
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def stop_recording(self):
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if(not self.in_stream.active):
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# print("Already stopped")
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return
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self.in_stream.stop()
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self.out_stream.stop()
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def is_recording(self):
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return self.in_stream.active
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def play_clip(self, clip_metadata):
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print(f"Playing clip: {clip_metadata}")
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clip_id = clip_metadata.get("filename")
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if clip_metadata.get("playbackType") == "playStop":
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if clip_id in self.clip_map:
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del self.clip_map[clip_id]
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return
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else:
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self.clip_map[clip_id] = []
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if clip_id not in self.clip_map:
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self.clip_map[clip_id] = []
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self.clip_map[clip_id].append(AudioClip(clip_metadata, target_sample_rate=self.output_sample_rate))
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@ -1,154 +0,0 @@
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import sounddevice as sd
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import numpy as np
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import os
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from datetime import datetime
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import scipy.io.wavfile as wavfile
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from metadata_manager import MetaDataManager
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|
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class AudioRecorder:
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_instance = None
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def __new__(cls, *args, **kwargs):
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if cls._instance is None:
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# print("Creating new AudioRecorder instance")
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cls._instance = super().__new__(cls)
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cls._instance.init()
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return cls._instance
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def init(self):
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"""
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Initialize audio recorder with configurable parameters.
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:param duration: Length of audio buffer in seconds
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:param sample_rate: Audio sample rate (if None, use default device sample rate)
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:param channels: Number of audio channels
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"""
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# print(f"Initializing AudioRecorder")
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self.duration = 30
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self.sample_rate = 44100
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self.channels = 2
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self.buffer = np.zeros((int(self.duration * self.sample_rate), self.channels), dtype=np.float32)
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self.recordings_dir = "recordings"
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self.stream = sd.InputStream(
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callback=self.record_callback
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)
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def refresh_stream(self):
|
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"""
|
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Refresh the audio stream with updated parameters.
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"""
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||||
was_active = self.stream.active
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if was_active:
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self.stream.stop()
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|
||||
self.buffer = np.zeros((int(self.duration * self.sample_rate), self.channels), dtype=np.float32)
|
||||
# print(f"AudioRecorder initialized with duration={self.duration}s, sample_rate={self.sample_rate}Hz, channels={self.channels}")
|
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self.stream = sd.InputStream(
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callback=self.record_callback
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||||
)
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||||
|
||||
if was_active:
|
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self.stream.start()
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||||
|
||||
|
||||
|
||||
|
||||
def record_callback(self, indata, frames, time, status):
|
||||
"""
|
||||
Circular buffer callback for continuous recording.
|
||||
|
||||
:param indata: Input audio data
|
||||
:param frames: Number of frames
|
||||
:param time: Timestamp
|
||||
:param status: Recording status
|
||||
"""
|
||||
if status:
|
||||
# print(f"Recording status: {status}")
|
||||
pass
|
||||
|
||||
# Circular buffer implementation
|
||||
self.buffer = np.roll(self.buffer, -frames, axis=0)
|
||||
self.buffer[-frames:] = indata
|
||||
|
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def save_last_n_seconds(self):
|
||||
"""
|
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Save the last n seconds of audio to a file.
|
||||
|
||||
:param output_dir: Directory to save recordings
|
||||
:return: Path to saved audio file
|
||||
"""
|
||||
# Create output directory if it doesn't exist
|
||||
os.makedirs(self.recordings_dir, exist_ok=True)
|
||||
|
||||
# Generate filename with timestamp
|
||||
timestamp = datetime.now().strftime("%Y%m%d_%H%M%S")
|
||||
filename = os.path.join(self.recordings_dir, f"audio_capture_{timestamp}.wav")
|
||||
|
||||
# Normalize audio to prevent clipping
|
||||
audio_data = self.buffer / np.max(np.abs(self.buffer)) * .5
|
||||
|
||||
# Convert float32 to int16 for WAV file
|
||||
audio_data_int16 = (audio_data * 32767).astype(np.int16)
|
||||
|
||||
# Write buffer to file
|
||||
wavfile.write(filename, int(self.sample_rate), audio_data_int16)
|
||||
|
||||
meta = MetaDataManager()
|
||||
|
||||
clip_metadata = {
|
||||
"filename": filename,
|
||||
"name": f"Clip {timestamp}",
|
||||
"playbackType":"playStop",
|
||||
"volume": 1.0,
|
||||
}
|
||||
|
||||
meta.add_clip_to_collection("Uncategorized", clip_metadata )
|
||||
|
||||
|
||||
return clip_metadata
|
||||
|
||||
def set_buffer_duration(self, duration):
|
||||
"""
|
||||
Set the duration of the audio buffer.
|
||||
|
||||
:param duration: New buffer duration in seconds
|
||||
"""
|
||||
self.duration = duration
|
||||
self.buffer = np.zeros((int(duration * self.sample_rate), self.channels), dtype=np.float32)
|
||||
|
||||
def set_recording_directory(self, directory):
|
||||
"""
|
||||
Set the directory where recordings will be saved.
|
||||
|
||||
:param directory: Path to the recordings directory
|
||||
"""
|
||||
self.recordings_dir = directory
|
||||
|
||||
def start_recording(self):
|
||||
"""
|
||||
Start continuous audio recording with circular buffer.
|
||||
"""
|
||||
if(self.stream.active):
|
||||
# print("Already recording")
|
||||
return
|
||||
# print('number of channels', self.channels)
|
||||
|
||||
self.stream.start()
|
||||
|
||||
def stop_recording(self):
|
||||
"""
|
||||
Stop continuous audio recording with circular buffer.
|
||||
"""
|
||||
if(not self.stream.active):
|
||||
# print("Already stopped")
|
||||
return
|
||||
|
||||
self.stream.stop()
|
||||
|
||||
def is_recording(self):
|
||||
"""
|
||||
Check if the audio stream is currently active.
|
||||
|
||||
:return: True if recording, False otherwise
|
||||
"""
|
||||
return self.stream.active
|
||||
@ -1,7 +1,7 @@
|
||||
import argparse
|
||||
import os
|
||||
import sys
|
||||
from audio_recorder import AudioRecorder
|
||||
from audio_io import AudioIO
|
||||
from windows_audio import WindowsAudioManager
|
||||
import sounddevice as sd
|
||||
from metadata_manager import MetaDataManager
|
||||
@ -41,6 +41,8 @@ def main():
|
||||
os.makedirs(settings.get_settings('save_path'), exist_ok=True)
|
||||
|
||||
|
||||
io = AudioIO()
|
||||
io.start_recording()
|
||||
# Register blueprints
|
||||
app.register_blueprint(recording_bp)
|
||||
app.register_blueprint(device_bp)
|
||||
|
||||
Binary file not shown.
Binary file not shown.
@ -1,11 +1,11 @@
|
||||
from flask import Blueprint, request, jsonify
|
||||
from windows_audio import WindowsAudioManager
|
||||
from audio_recorder import AudioRecorder
|
||||
from audio_io import AudioIO
|
||||
|
||||
device_bp = Blueprint('device', __name__)
|
||||
|
||||
audio_manager = WindowsAudioManager()
|
||||
recorder = AudioRecorder()
|
||||
recorder = AudioIO()
|
||||
|
||||
# @device_bp.route('/device/set', methods=['POST'])
|
||||
# def set_audio_device():
|
||||
|
||||
@ -1,27 +1,27 @@
|
||||
|
||||
from flask import Blueprint, request, jsonify
|
||||
from audio_recorder import AudioRecorder
|
||||
from audio_io import AudioIO
|
||||
import os
|
||||
|
||||
recording_bp = Blueprint('recording', __name__)
|
||||
|
||||
@recording_bp.route('/record/start', methods=['POST'])
|
||||
def start_recording():
|
||||
recorder = AudioRecorder()
|
||||
recorder = AudioIO()
|
||||
print('HTTP: Starting audio recording')
|
||||
recorder.start_recording()
|
||||
return jsonify({'status': 'recording started'})
|
||||
|
||||
@recording_bp.route('/record/stop', methods=['POST'])
|
||||
def stop_recording():
|
||||
recorder = AudioRecorder()
|
||||
recorder = AudioIO()
|
||||
# print('HTTP: Stopping audio recording')
|
||||
recorder.stop_recording()
|
||||
return jsonify({'status': 'recording stopped'})
|
||||
|
||||
@recording_bp.route('/record/save', methods=['POST'])
|
||||
def save_recording():
|
||||
recorder = AudioRecorder()
|
||||
recorder = AudioIO()
|
||||
# print('HTTP: Saving audio recording')
|
||||
saved_file = recorder.save_last_n_seconds()
|
||||
return jsonify({'status': 'recording saved', 'file': saved_file})
|
||||
@ -29,7 +29,7 @@ def save_recording():
|
||||
|
||||
@recording_bp.route('/record/status', methods=['GET'])
|
||||
def recording_status():
|
||||
recorder = AudioRecorder()
|
||||
recorder = AudioIO()
|
||||
# print('HTTP: Checking recording status')
|
||||
status = 'recording' if recorder.is_recording() else 'stopped'
|
||||
return jsonify({'status': status})
|
||||
@ -45,8 +45,12 @@ def recording_delete():
|
||||
|
||||
@recording_bp.route('/playback/start', methods=['POST'])
|
||||
def playback_start():
|
||||
print(f"Playing clip")
|
||||
# print('HTTP: Starting audio playback')
|
||||
clip = request.json
|
||||
try:
|
||||
io = AudioIO()
|
||||
io.play_clip(clip)
|
||||
# os.remove(filename)
|
||||
return jsonify({'status': 'success'})
|
||||
except Exception as e:
|
||||
|
||||
@ -1,6 +1,6 @@
|
||||
import os
|
||||
import json
|
||||
from audio_recorder import AudioRecorder
|
||||
from audio_io import AudioIO
|
||||
from windows_audio import WindowsAudioManager
|
||||
|
||||
class SettingsManager:
|
||||
@ -44,7 +44,6 @@ class SettingsManager:
|
||||
input = self.settings["input_device"]
|
||||
if("output_device" in self.settings):
|
||||
output = self.settings["output_device"]
|
||||
|
||||
#see if input device is in "devices", if not set to the first index
|
||||
if input is not None and any(d['name'] == input["name"] for d in input_devices):
|
||||
# print(f"Using saved input device index: {input}")
|
||||
@ -64,6 +63,7 @@ class SettingsManager:
|
||||
if not "http_port" in self.settings:
|
||||
self.settings["http_port"] = 5010
|
||||
|
||||
|
||||
self.save_settings()
|
||||
|
||||
|
||||
@ -88,13 +88,14 @@ class SettingsManager:
|
||||
json.dump(self.settings, f, indent=4)
|
||||
|
||||
def refresh_settings(self):
|
||||
recorder = AudioRecorder()
|
||||
recorder = AudioIO()
|
||||
# Update recorder parameters based on new setting
|
||||
recorder.set_buffer_duration(self.get_settings('recording_length'))
|
||||
recorder.recordings_dir = self.get_settings('save_path')
|
||||
|
||||
audio_manager = WindowsAudioManager()
|
||||
audio_manager.set_default_input_device(self.get_settings('input_device')['index'])
|
||||
audio_manager.set_default_output_device(self.get_settings('output_device')['index'])
|
||||
|
||||
recorder.refresh_stream()
|
||||
recorder.refresh_streams()
|
||||
|
||||
|
||||
@ -81,7 +81,7 @@ class WindowsAudioManager:
|
||||
|
||||
def set_default_input_device(self, device_index):
|
||||
if(device_index is None):
|
||||
return self.get_current_input_device_sample_rate()
|
||||
return 0
|
||||
"""
|
||||
Set the default input audio device.
|
||||
|
||||
@ -95,41 +95,18 @@ class WindowsAudioManager:
|
||||
device_info = sd.query_devices(device_index)
|
||||
return device_info['default_samplerate']
|
||||
|
||||
def get_current_input_device_sample_rate(self):
|
||||
def set_default_output_device(self, device_index):
|
||||
if(device_index is None):
|
||||
return self.get_current_output_device_sample_rate()
|
||||
"""
|
||||
Get the sample rate of the current input device.
|
||||
Set the default output audio device.
|
||||
|
||||
:return: Sample rate of the current input device
|
||||
:param device_index: Index of the audio device
|
||||
:return: Sample rate of the selected device
|
||||
"""
|
||||
device_info = sd.query_devices(self.default_input)
|
||||
sd.default.device[1] = device_index
|
||||
self.default_output = device_index
|
||||
|
||||
# Get the sample rate of the selected device
|
||||
device_info = sd.query_devices(device_index)
|
||||
return device_info['default_samplerate']
|
||||
|
||||
def get_system_volume(self):
|
||||
"""
|
||||
Get the system master volume.
|
||||
|
||||
:return: Current system volume (0.0 to 1.0)
|
||||
"""
|
||||
devices = AudioUtilities.GetSpeakers()
|
||||
interface = devices.Activate(
|
||||
IAudioEndpointVolume._iid_,
|
||||
CLSCTX_ALL,
|
||||
None
|
||||
)
|
||||
volume = interface.QueryInterface(IAudioEndpointVolume)
|
||||
return volume.GetMasterVolumeLevelScalar()
|
||||
|
||||
def set_system_volume(self, volume_level):
|
||||
"""
|
||||
Set the system master volume.
|
||||
|
||||
:param volume_level: Volume level (0.0 to 1.0)
|
||||
"""
|
||||
devices = AudioUtilities.GetSpeakers()
|
||||
interface = devices.Activate(
|
||||
IAudioEndpointVolume._iid_,
|
||||
CLSCTX_ALL,
|
||||
None
|
||||
)
|
||||
volume = interface.QueryInterface(IAudioEndpointVolume)
|
||||
volume.SetMasterVolumeLevelScalar(volume_level, None)
|
||||
@ -43,7 +43,7 @@ export default class PythonSubprocessManager {
|
||||
console.log(`Python stdout: ${data.toString()}`);
|
||||
});
|
||||
this.process.stderr.on('data', (data: Buffer) => {
|
||||
console.error(`Python stderr: ${data.toString()}`);
|
||||
// console.error(`Python stderr: ${data.toString()}`);
|
||||
const lines = data.toString().split('\n');
|
||||
// eslint-disable-next-line no-restricted-syntax
|
||||
for (const line of lines) {
|
||||
|
||||
Reference in New Issue
Block a user