Add STT code

1 files changed, 581 insertions, 0 deletions

diff --git a/stt.py b/stt.py
new file mode 100644
index 0000000..34ef2e9
--- /dev/null
+++ b/stt.py
@@ -0,0 +1,581 @@
+from faster_whisper import WhisperModel
+import langcodes
+import numpy as np
+import os
+import pyaudio
+from pydub import AudioSegment
+from shared_thread_data import SharedThreadData
+import sys
+import time
+import typing
+import vad
+
+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):
+        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
+
+        print(f"Finding mic {which_mic}", file=sys.stderr)
+        self.dumpMicDevices()
+
+        got_match = False
+        device_index = -1
+        if which_mic == "index":
+            target_str = "Digital Audio Interface"
+        elif which_mic == "focusrite":
+            target_str = "Focusrite"
+        elif which_mic == "motu":
+            target_str = "In 1-2 (MOTU M Series)"
+        elif which_mic == "beyond":
+            target_str = "Microphone (Beyond)"
+        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=AudioStream.CHANNELS,
+                format=AudioStream.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 * AudioStream.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 = AudioStream.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) / AudioStream.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 * AudioStream.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) / (AudioStream.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
+
+    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()
+
+# Audio collector that enforces a minimum length on its audio data.
+class LengthEnforcingAudioCollector(AudioCollectorFilter):
+    def __init__(self, parent: AudioCollector, min_duration_s: float):
+        AudioCollectorFilter.__init__(self, parent)
+        self.min_duration_s = min_duration_s
+
+    def getAudio(self) -> bytes:
+        audio = self.parent.getAudio()
+        min_duration_frames = int(self.min_duration_s * AudioStream.FPS)
+        pad_len_frames = max(0, min_duration_frames - int(len(audio) /
+            AudioStream.FRAME_SZ))
+        pad = np.zeros(pad_len_frames, dtype=np.int16).tobytes()
+        return pad + audio
+
+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=AudioStream.FRAME_SZ,
+                frame_rate=AudioStream.FPS, channels=AudioStream.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=AudioStream.FRAME_SZ,
+                frame_rate=AudioStream.FPS, channels=AudioStream.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):
+        self.vad_options = vad.VadOptions(
+                min_silence_duration_ms=min_silence_ms,
+                max_speech_duration_s=max_speech_s)
+        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 *
+                AudioStream.FPS) / 1000.0)
+        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) / AudioStream.FRAME_SZ)
+                #print(f"now: {now}")
+                #print(f"min d: {min_delta_frames}")
+                delta_frames = now - s['end']
+                if delta_frames > min_delta_frames:
+                    cutoff = now - int(min_delta_frames / 2)
+
+        return (cutoff, len(segments) > 0)
+
+# A segment of transcribed audio. `start_ts` and `end_ts` are floating point
+# number of seconds since the beginning of audio data.
+class Segment:
+    def __init__(self,
+            transcript: str,
+            start_ts: float,
+            end_ts: float,
+            wall_ts: float,
+            avg_logprob: float,
+            no_speech_prob: float,
+            compression_ratio: float):
+        self.transcript = transcript
+        # start_ts, end_ts are timestamps in seconds relative to `wall_ts`.
+        self.start_ts = start_ts
+        self.end_ts = end_ts
+        # wall_ts is the time.time() at which the oldest audio sample leading
+        # to this transcript was collected.
+        self.wall_ts = wall_ts
+        self.avg_logprob = avg_logprob
+        self.no_speech_prob = no_speech_prob
+        self.compression_ratio = compression_ratio
+
+    def __str__(self):
+        ts = f"(ts: {self.start_ts}-{self.end_ts}) "
+
+        wall_ts_start = datetime.utcfromtimestamp(self.start_ts + self.wall_ts).strftime('%H:%M:%S')
+        wall_ts_end = datetime.utcfromtimestamp(self.end_ts + self.wall_ts).strftime('%H:%M:%S')
+        wall_ts = f"(wall ts: {wall_ts_start}-{wall_ts_end}) "
+
+        no_speech = f"(no_speech: {self.no_speech_prob}) "
+        avg_logprob = f"(avg_logprob: {self.avg_logprob}) "
+        return f"{self.transcript} " + ts + wall_ts + no_speech + avg_logprob
+
+class Whisper:
+    def __init__(self,
+            collector: AudioCollector,
+            cfg: typing.Dict):
+        self.collector = collector
+        self.model = None
+        self.cfg = cfg
+
+        abspath = os.path.abspath(__file__)
+        my_dir = os.path.dirname(abspath)
+        parent_dir = os.path.dirname(my_dir)
+
+        model_str = cfg["model"]
+        model_root = os.path.join(parent_dir, "Models",
+                os.path.normpath(model_str))
+        print(f"Model {cfg['model']} will be saved to {model_root}",
+                file=sys.stderr)
+
+        model_device = "cuda"
+        if cfg["use_cpu"]:
+            model_device = "cpu"
+
+        already_downloaded = os.path.exists(model_root)
+
+        self.model = WhisperModel(model_str,
+                device = model_device,
+                device_index = cfg["gpu_idx"],
+                compute_type = cfg["compute_type"],
+                download_root = model_root,
+                local_files_only = already_downloaded)
+
+    def transcribe(self, frames: bytes = None) -> typing.List[Segment]:
+        if frames is None:
+            frames = self.collector.getAudio()
+        # Convert from signed 16-bit int [-32768, 32767] to signed 32-bit float on
+        # [-1, 1].
+        audio = np.frombuffer(frames,
+                dtype=np.int16).flatten().astype(np.float32) / 32768.0
+
+        t0 = time.time()
+        segments, info = self.model.transcribe(
+                audio,
+                language = langcodes.find(self.cfg["language"]).language,
+                vad_filter = True,
+                temperature=0.0,
+                without_timestamps = False)
+        res = []
+        for s in segments:
+            # Manual touchup. I see a decent number of hallucinations sneaking
+            # in with high `no_speech_prob` and modest `avg_logprob`.
+            if s.no_speech_prob > 0.6 and s.avg_logprob < -0.5:
+                if self.cfg["enable_debug_mode"]:
+                    print(f"Drop probable hallucination (case 1) " +
+                            f"(text='{s.text}', " +
+                            f"no_speech_prob={s.no_speech_prob}, " +
+                            f"avg_logprob={s.avg_logprob})", file=sys.stderr)
+                continue
+            # Another touchup targeted at the vexatious "thanks for watching!"
+            # hallucination. This triggers a lot when listening to
+            # instrumental/electronic music.
+            if s.no_speech_prob > 0.15 and s.avg_logprob < -0.7:
+                if self.cfg["enable_debug_mode"]:
+                    print(f"Drop probable hallucination (case 2) " +
+                            f"(text='{s.text}', " +
+                            f"no_speech_prob={s.no_speech_prob}, " +
+                            f"avg_logprob={s.avg_logprob})", file=sys.stderr)
+                continue
+            if self.cfg["enable_debug_mode"]:
+                print(f"s get: {s}")
+            if s.avg_logprob < -1.0:
+                continue
+            if s.compression_ratio > 2.4:
+                continue
+            res.append(Segment(s.text, s.start, s.end,
+                self.collector.begin(),
+                s.avg_logprob, s.no_speech_prob, s.compression_ratio))
+        t1 = time.time()
+        if self.cfg["enable_debug_mode"]:
+            print(f"Transcription latency (s): {t1 - t0}")
+        return res
+
+class TranscriptCommit:
+    def __init__(self,
+            delta: str,
+            preview: str,
+            latency_s: float = None,
+            thresh_at_commit: int = None,
+            audio: bytes = None,
+            duration_s: float = None,
+            start_ts: float = None):
+        self.delta = delta
+        self.preview = preview
+        self.latency_s = latency_s
+        self.thresh_at_commit = thresh_at_commit
+        self.audio = audio
+        # Time at which the commit is generated
+        self.ts = time.time()
+        # Time corresponding to the start of the segment
+        self.start_ts = start_ts
+        # The duration of the audio segment, in seconds.
+        self.duration_s = duration_s
+
+
+class VadCommitter:
+    def __init__(self,
+            cfg: typing.Dict,
+            collector: AudioCollector,
+            whisper: Whisper,
+            segmenter: AudioSegmenter):
+        self.cfg = cfg
+        self.collector = collector
+        self.whisper = whisper
+        self.segmenter = segmenter
+
+    def getDelta(self) -> TranscriptCommit:
+        audio = self.collector.getAudio()
+        stable_cutoff, has_audio = self.segmenter.getStableCutoff(audio)
+
+        delta = ""
+        commit_audio = None
+        latency_s = None
+        duration_s = self.collector.duration()
+        start_ts = self.collector.begin()
+
+        if has_audio and stable_cutoff:
+            #print(f"stable cutoff get: {stable_cutoff}", file=sys.stderr)
+            latency_s = self.collector.now() - self.collector.begin()
+            duration_s = stable_cutoff / AudioStream.FPS
+            start_ts = self.collector.begin()
+            commit_audio = self.collector.dropAudioPrefixByFrames(stable_cutoff)
+
+            segments = self.whisper.transcribe(commit_audio)
+            delta = ''.join(s.transcript for s in segments)
+            audio = self.collector.getAudio()
+            if self.cfg["enable_debug_mode"]:
+                for s in segments:
+                    print(f"commit segment: {s}", file=sys.stderr)
+                print(f"delta get: {delta}", file=sys.stderr)
+
+            if False:
+                ts = datetime.fromtimestamp(self.collector.now() - latency_s)
+                filename = str(ts.strftime('%Y_%m_%d__%H-%M-%S')) + ".wav"
+                saveAudio(commit_audio, filename)
+
+        preview = ""
+        if self.cfg["enable_previews"] and has_audio:
+            segments = self.whisper.transcribe(audio)
+            preview = "".join(s.transcript for s in segments)
+
+        if not has_audio:
+            #print("VAD detects no audio, skip transcription", file=sys.stderr)
+            self.collector.keepLast(1.0)
+
+        return TranscriptCommit(
+                delta.strip(),
+                preview.strip(),
+                latency_s,
+                audio=audio,
+                duration_s=duration_s,
+                start_ts=start_ts)
+
+def transcriptionThread(shared_data: SharedThreadData):
+    last_stable_commit = None
+
+    stream = MicStream(shared_data.cfg["microphone"])
+    collector = AudioCollector(stream)
+    collector = NormalizingAudioCollector(collector)
+    collector = CompressingAudioCollector(collector)
+    whisper = Whisper(collector, shared_data.cfg)
+    segmenter = AudioSegmenter(min_silence_ms=shared_data.cfg["min_silence_duration_ms"],
+            max_speech_s=shared_data.cfg["max_speech_duration_s"])
+    committer = VadCommitter(shared_data.cfg, collector, whisper, segmenter)
+
+    transcript = ""
+    preview = ""
+
+    while not shared_data.exit_event.is_set():
+        time.sleep(shared_data.cfg["transcription_loop_delay_ms"] / 1000.0);
+
+        op = None
+
+        commit = committer.getDelta()
+
+        if len(commit.delta) > 0 or len(commit.preview) > 0:
+            # Avoid re-sending text after long pauses. User controls the length
+            # of the pause in the UI.
+            if shared_data.cfg["reset_after_silence_s"] > 0:
+                silence_duration = 0
+                if last_stable_commit:
+                    last_commit_end_ts = \
+                            last_stable_commit.start_ts + \
+                            last_stable_commit.duration_s
+                    silence_duration = commit.start_ts - last_commit_end_ts
+                if silence_duration > shared_data.cfg["reset_after_silence_s"]:
+                    print(f"Resetting transcript after {silence_duration}-second "
+                            "silence", file=sys.stderr)
+                    transcript = ""
+                    preview = ""
+                if commit.delta:
+                    last_stable_commit = commit
+
+            # Hard-cap displayed transcript length at 4k characters to prevent
+            # runaway memory use in UI. Keep the full transcript to avoid
+            # breaking OSC pager.
+            transcript = transcript[-4096:]
+            def join_segments(a, b):
+                if len(a) > 0 and a[-1] != ' ':
+                    return a + ' ' + b
+                else:
+                    return a + b
+            transcript = join_segments(transcript, commit.delta)
+            preview = commit.preview
+
+            try:
+                print(f"Transcript: {transcript}")
+            except UnicodeEncodeError:
+                print("Failed to encode transcript - discarding delta",
+                        file=sys.stderr)
+                continue
+            try:
+                print(f"Preview: {preview}")
+            except UnicodeEncodeError:
+                print("Failed to encode preview - discarding", file=sys.stderr)
+
+            with shared_data.word_lock:
+                shared_data.word = join_segments(transcript, preview)
+
+            if shared_data.cfg["enable_debug_mode"]:
+                print(f"commit latency: {commit.latency_s}", file=sys.stderr)
+                print(f"commit thresh: {commit.thresh_at_commit}",
+                        file=sys.stderr)
+
+        if len(transcript) > 0 and \
+                (not transcript.endswith(' ')) and \
+                (not commit.delta.startswith(' ')):
+            commit.delta = ' ' + commit.delta
+        if len(commit.delta) > 0 and \
+                (not commit.delta.endswith(' ')) and \
+                (not commit.preview.startswith(' ')):
+            commit.preview = ' ' + commit.preview
+