diff options
Diffstat (limited to 'app.py')
| -rw-r--r-- | app.py | 417 |
1 files changed, 417 insertions, 0 deletions
@@ -0,0 +1,417 @@ +from datetime import datetime +from pydub import AudioSegment + +import math +import numpy as np +import os +import pyaudio +import sys +import time +import typing +import vad +import wave + +class AudioStream(): + FORMAT = pyaudio.paInt16 + # Size of each frame (audio sample), in bytes. If you change FORMAT, make + # sure this stays up to date! + FRAME_SZ = 2 + # Frames per second. + FPS = 16000 + CHANNELS = 1 + def __init__(self): + pass + + def getSamples(self) -> bytes: + raise NotImplementedError("getSamples is not implemented!") + +class MicStream(AudioStream): + CHUNK_SZ = 1024 + + def __init__(self, which_mic: str, fps: int = AudioStream.FPS): + self.p = pyaudio.PyAudio() + self.stream = None + self.sample_rate = None + # Each time pyaudio gives us audio data, it's in the form of a chunk of + # samples. We keep these in a list to keep the audio callback as light + # as possible. Whenever downstream layers want data, we collapse the + # list into a single array of data (a bytes object). + self.chunks = [] + # If set, incoming frames are simply discarded. + self.paused = False + self.fps = fps + + print(f"Finding mic {which_mic}", file=sys.stderr) + self.dumpMicDevices() + + got_match = False + device_index = -1 + focusrite_str = "Focusrite" + index_str = "Digital Audio Interface" + if which_mic == "index": + target_str = index_str + elif which_mic == "focusrite": + target_str = focusrite_str + else: + print(f"Mic {which_mic} requested, treating it as a numerical " + + "device ID", file=sys.stderr) + device_index = int(which_mic) + got_match = True + if not got_match: + info = self.p.get_host_api_info_by_index(0) + numdevices = info.get('deviceCount') + for i in range(0, numdevices): + if (self.p.get_device_info_by_host_api_device_index(0, i).get('maxInputChannels')) > 0: + device_name = self.p.get_device_info_by_host_api_device_index(0, i).get('name') + if target_str in device_name: + print(f"Got matching mic: {device_name}", + file=sys.stderr) + device_index = i + got_match = True + break + if not got_match: + raise KeyError(f"Mic {which_mic} not found") + + info = self.p.get_device_info_by_host_api_device_index(0, device_index) + print(f"Found mic {which_mic}: {info['name']}", file=sys.stderr) + self.sample_rate = int(info['defaultSampleRate']) + print(f"Mic sample rate: {self.sample_rate}", file=sys.stderr) + + self.stream = self.p.open( + rate=self.sample_rate, + channels=self.CHANNELS, + format=self.FORMAT, + input=True, + frames_per_buffer=MicStream.CHUNK_SZ, + input_device_index=device_index, + stream_callback=self.onAudioFramesAvailable) + + self.stream.start_stream() + + AudioStream.__init__(self) + + def pause(self, state: bool = True): + self.paused = state + + def dumpMicDevices(self): + info = self.p.get_host_api_info_by_index(0) + numdevices = info.get('deviceCount') + + for i in range(0, numdevices): + if (self.p.get_device_info_by_host_api_device_index(0, i).get('maxInputChannels')) > 0: + device_name = self.p.get_device_info_by_host_api_device_index(0, i).get('name') + print("Input Device id ", i, " - ", device_name) + + def onAudioFramesAvailable(self, + frames, + frame_count, + time_info, + status_flags): + if self.paused: + # Don't literally pause, just start returning silence. This allows + # the `min_segment_age_s` check to work while paused. + n_frames = int(frame_count * self.fps / + float(self.sample_rate)) + self.chunks.append(np.zeros(n_frames, + dtype=np.int16).tobytes()) + return (frames, pyaudio.paContinue) + + decimated = b'' + # In pyaudio, a `frame` is a single sample of audio data. + frame_len = self.FRAME_SZ + next_frame = 0.0 + # The mic probably has a higher sample rate than Whisper wants, so + # decrease the sample rate by dropping samples. Note that this + # algorithm only works if the mic's rate is higher than whisper's + # expected rate. + keep_every = float(self.sample_rate) / self.fps + for i in range(frame_count): + if i >= next_frame: + decimated += frames[i*frame_len:(i+1)*frame_len] + next_frame += keep_every + self.chunks.append(decimated) + + return (frames, pyaudio.paContinue) + + # Get audio data and the corresponding timestamp. + def getSamples(self) -> bytes: + chunks = self.chunks + self.chunks = [] + result = b''.join(chunks) + return result + +class AudioCollector: + def __init__(self, stream: AudioStream): + self.stream = stream + self.frames = b'' + # Note: by design, this is the only spot where we anchor our timestamps + # against the real world. This is done to make it possible to profile + # test cases which read from disk (at much faster than real speed) in + # the same way that we profile real-time data. + self.wall_ts = time.time() + + def getAudio(self) -> bytes: + frames = self.stream.getSamples() + if frames: + self.frames += frames + return self.frames + + def dropAudioPrefix(self, dur_s: float) -> bytes: + n_bytes = int(dur_s * self.stream.fps) * self.stream.FRAME_SZ + n_bytes = min(n_bytes, len(self.frames)) + cut_portion = self.frames[:n_bytes] + self.frames = self.frames[n_bytes:] + self.wall_ts += float(n_bytes / self.stream.FRAME_SZ) / self.stream.fps + return cut_portion + + def dropAudioPrefixByFrames(self, dur_frames: int) -> bytes: + n_bytes = dur_frames * self.stream.FRAME_SZ + n_bytes = min(n_bytes, len(self.frames)) + cut_portion = self.frames[:n_bytes] + self.frames = self.frames[n_bytes:] + self.wall_ts += float(n_bytes / self.stream.FRAME_SZ) / self.stream.fps + return cut_portion + + def keepLast(self, dur_s: float) -> bytes: + drop_len = max(0, self.duration() - dur_s) + return self.dropAudioPrefix(drop_len) + + def dropAudio(self): + self.wall_ts += self.duration() + cut_portion = self.frames + self.frames = b'' + return cut_portion + + def duration(self): + return len(self.frames) / (self.stream.fps * self.stream.FRAME_SZ) + + def begin(self): + return self.wall_ts + + def now(self): + return self.begin() + self.duration() + +class AudioCollectorFilter: + def __init__(self, parent: AudioCollector): + self.parent = parent + self.stream = self.parent.stream + + def getAudio(self) -> bytes: + return self.parent.getAudio() + def dropAudioPrefix(self, dur_s: float): + return self.parent.dropAudioPrefix(dur_s) + def dropAudioPrefixByFrames(self, dur_frames: int): + return self.parent.dropAudioPrefixByFrames(dur_frames) + def keepLast(self, dur_s): + return self.parent.keepLast(dur_s) + def dropAudio(self): + return self.parent.dropAudio() + def duration(self): + return self.parent.duration() + def begin(self): + return self.parent.begin() + def now(self): + return self.parent.now() + +class NormalizingAudioCollector(AudioCollectorFilter): + def __init__(self, parent: AudioCollector): + AudioCollectorFilter.__init__(self, parent) + + def getAudio(self) -> bytes: + audio = self.parent.getAudio() + + audio = AudioSegment(audio, sample_width=self.stream.FRAME_SZ, + frame_rate=self.stream.fps, channels=self.stream.CHANNELS) + audio = audio.normalize() + + frames = np.array(audio.get_array_of_samples()) + frames = np.int16(frames).tobytes() + + return frames + +class CompressingAudioCollector(AudioCollectorFilter): + def __init__(self, parent: AudioCollector): + AudioCollectorFilter.__init__(self, parent) + + def getAudio(self) -> bytes: + audio = self.parent.getAudio() + + audio = AudioSegment(audio, + sample_width=self.stream.FRAME_SZ, + frame_rate=self.stream.fps, + channels=self.stream.CHANNELS) + # subtle compression has a slight positive effect on my benchmark + audio = audio.compress_dynamic_range(threshold=-10, ratio=2.0) + + frames = np.array(audio.get_array_of_samples()) + frames = np.int16(frames).tobytes() + + return frames + +class AudioSegmenter: + def __init__(self, + min_silence_ms=250, + max_speech_s=5, + stream: AudioStream = None): + self.vad_options = vad.VadOptions( + min_silence_duration_ms=min_silence_ms, + max_speech_duration_s=max_speech_s) + self.stream = stream + pass + + def segmentAudio(self, audio: bytes): + audio = np.frombuffer(audio, + dtype=np.int16).flatten().astype(np.float32) / 32768.0 + return vad.get_speech_timestamps(audio, vad_options=self.vad_options) + + # Returns the stable cutoff (if any) and whether there are any segments. + def getStableCutoff(self, audio: bytes) -> typing.Tuple[int, bool]: + min_delta_frames = int((self.vad_options.min_silence_duration_ms * + self.stream.fps) / 1000) + cutoff = None + + last_end = None + segments = self.segmentAudio(audio) + + for i in range(len(segments)): + s = segments[i] + #print(f"s: {s}") + #print(f"last_end: {last_end}") + + if last_end: + delta_frames = s['start'] - last_end + #print(f"delta frames: {delta_frames}") + if delta_frames > min_delta_frames: + cutoff = s['start'] + else: + last_end = s['end'] + + if i == len(segments) - 1: + now = int(len(audio) / self.stream.FRAME_SZ) + delta_frames = now - s['end'] + if delta_frames > min_delta_frames: + cutoff = now - int(min_delta_frames / 2) + + return (cutoff, len(segments) > 0) + +def install_in_venv(pkgs: typing.List[str]) -> bool: + pkgs_str = " ".join(pkgs) + print(f"Installing {pkgs_str}") + pip_proc = subprocess.Popen( + f"Resources/Python/python.exe -m pip install {pkgs_str} --no-warn-script-location".split(), + stdout=subprocess.PIPE, + stderr=subprocess.PIPE) + pip_stdout, pip_stderr = pip_proc.communicate() + pip_stdout = pip_stdout.decode("utf-8") + pip_stderr = pip_stderr.decode("utf-8") + print(pip_stdout, file=sys.stderr) + print(pip_stderr, file=sys.stderr) + if pip_proc.returncode != 0: + print(f"`pip install {pkgs_str}` exited with {pip_proc.returncode}", + file=sys.stderr) + return False + return True + +def saveAudio(audio: bytes, path: str, stream: AudioStream): + with wave.open(path, 'wb') as wf: + print(f"Saving audio to {path}", file=sys.stderr) + wf.setnchannels(stream.CHANNELS) + wf.setsampwidth(stream.FRAME_SZ) + wf.setframerate(stream.fps) + wf.writeframes(audio) + +def concatenate_wav_files(output_path): + # List all .wav files in the CWD + wav_files = [f for f in os.listdir('.') if f.endswith('.wav')] + + # Initialize parameters for wave file + params = None + + # Open the output file + with wave.open(output_path, 'wb') as output_wav: + for wav_file in wav_files: + print(f"Processing {wav_file}") + with wave.open(wav_file, 'rb') as input_wav: + # Check if parameters are the same for each file + if params is None: + params = input_wav.getparams() + output_wav.setparams(params) + + # Read and write frames + frames = input_wav.readframes(input_wav.getnframes()) + output_wav.writeframes(frames) + +if __name__ == "__main__": + abspath = os.path.abspath(__file__) + dname = os.path.dirname(abspath) + os.chdir(dname) + print(f"Set cwd to {os.getcwd()}", file=sys.stderr) + + concatenate_wav_files("concatenated.wav") + sys.exit(0) + + stream = MicStream("index") + stream_hd = MicStream("index", fps=44100) + + collector = AudioCollector(stream) + #collector = NormalizingAudioCollector(collector) + collector = CompressingAudioCollector(collector) + + collector_hd = AudioCollector(stream_hd) + #collector_hd = NormalizingAudioCollector(collector_hd) + collector_hd = CompressingAudioCollector(collector_hd) + + min_silence_ms = 1000 + max_speech_s = 30 + segmenter = AudioSegmenter( + min_silence_ms=min_silence_ms, + max_speech_s=max_speech_s, + stream=stream) + + while True: + audio = collector.getAudio() + collector_hd.getAudio() + stable_cutoff, has_audio = segmenter.getStableCutoff(audio) + + #print(f"has audio: {has_audio}") + #print(f"stable cutoff: {stable_cutoff}") + + if has_audio and stable_cutoff: + commit_audio = collector.dropAudioPrefixByFrames(stable_cutoff) + print(f"stable cutoff: {stable_cutoff}") + hd_cutoff = int(math.floor(stable_cutoff * stream_hd.fps / + stream.fps)) + print(f"hd cutoff: {hd_cutoff}") + commit_audio_hd = collector_hd.dropAudioPrefixByFrames(hd_cutoff) + print(f"hd audio len: {len(commit_audio_hd)}") + + # Calculate naive measure of volume + audio_v = AudioSegment(commit_audio_hd, + sample_width=stream_hd.FRAME_SZ, + frame_rate=stream_hd.fps, + channels=stream_hd.CHANNELS) + audio_v = np.array(audio_v.get_array_of_samples()) + audio_v = np.int16(audio_v) + audio_v = np.sqrt(np.mean(np.square(audio_v))) + audio_v /= np.sqrt(len(commit_audio_hd) / stream_hd.FRAME_SZ) + audio_v = math.log(audio_v, 10) + print(f"volume: {audio_v}") + # cutoff is a fine-tuned value based on volumes seen while in vr + # (index mic) + if audio_v < -1.3 or audio_v > -0.8: + # Discard sample + print("Discarding too-quiet/too-loud segment") + collector.keepLast(1.0) + collector_hd.keepLast(1.0) + continue + + + ts = datetime.fromtimestamp(time.time()) + filename = str(ts.strftime('%Y_%m_%d__%H-%M-%S')) + ".wav" + saveAudio(commit_audio_hd, filename, stream_hd) + + if not has_audio: + #print("VAD detects no audio, skip transcription", file=sys.stderr) + collector.keepLast(1.0) + collector_hd.keepLast(1.0) + |