diff options
Diffstat (limited to 'Scripts')
| -rw-r--r-- | Scripts/transcribe.py | 34 |
1 files changed, 25 insertions, 9 deletions
diff --git a/Scripts/transcribe.py b/Scripts/transcribe.py index d937cb6..4a438b7 100644 --- a/Scripts/transcribe.py +++ b/Scripts/transcribe.py @@ -4,7 +4,7 @@ from datetime import datetime from emotes_v2 import EmotesState from faster_whisper import WhisperModel from functools import partial -from math import ceil +from math import floor, ceil from playsound import playsound from profanity_filter import ProfanityFilter from sentence_splitter import split_text_into_sentences @@ -78,7 +78,7 @@ class AudioState: # Segment start time, end time, and text self.ranges_ls = [] self.frames = [] - self.drop_frames_till_i = -1 + self.drop_samples_till_i = -1 # Locks access to `text`. self.transcribe_lock = threading.Lock() @@ -153,10 +153,24 @@ def onAudioFramesAvailable( audio_state.CHUNK) if len(audio_state.frames) > max_frames: audio_state.frames = audio_state.frames[-1 * max_frames:] - if audio_state.drop_frames_till_i > 0: - audio_state.frames = audio_state.frames[audio_state.drop_frames_till_i:-1] - audio_state.drop_frames_till_i = -1 - + if audio_state.drop_samples_till_i > 0: + # Caller wants us to keep this many *whisper* samples, assuming that + # we're getting one full frame every (1024 / 16KHz) seconds. + # However we really get one full whisper frame a little slower, since + # mics usually have a higher sample rate than 16 KHz (see decimation + # code above). + # The ratio of (mic sample rate) / (16KHz) is simply `keep_every`. + # + # TODO we should be more careful and avoid dropping an entire chunk + # since phonemes can absolutely fit into that amount of time. However + # whisper can usually figure it out so it's not a huge deal. + n_frames_to_drop = audio_state.drop_samples_till_i / audio_state.CHUNK + n_frames_to_drop *= keep_every + n_frames_to_drop = int(ceil(n_frames_to_drop)) + if audio_state.enable_debug_mode: + print(f"Dropping {n_frames_to_drop} frames, buffer has {len(audio_state.frames)} frames total") + audio_state.frames = audio_state.frames[n_frames_to_drop:] + audio_state.drop_samples_till_i = -1 return (frames, pyaudio.paContinue) @@ -282,15 +296,15 @@ def transcribe(audio_state, model, frames, use_cpu: bool) -> typing.Tuple[str,st max_edit = audio_state.commit_fuzz_threshold if audio_state.enable_debug_mode: - print(f"c0: {c0}, c1: {c1}, c2: {c2}") + print(f"c0: {c0}, c1: {c1}, c2: {c2}, c3: {c3}") if c0_c1_d < max_edit and c1_c2_d < max_edit and c2_c3_d < max_edit: # For simplicity, completely reset saved audio ranges. audio_state.ranges_ls = [] committed_text = c0[2] if audio_state.enable_debug_mode: print(f"Dropping frames until {c0[1]}") - n_frames_to_drop = int(ceil(audio_state.RATE * c0[1])) - audio_state.drop_frames_till_i = n_frames_to_drop + n_samples_to_drop = int(ceil(audio_state.RATE * c0[1])) + audio_state.drop_samples_till_i = n_samples_to_drop preview_text = "" for seg in ranges: @@ -357,6 +371,8 @@ def transcribeAudio(audio_state, now - last_transcribe_time, audio_state.preview_text)) last_transcribe_time = now + print(f"Commit text: {text}") + print(f"Preview text: {preview_text}") # Translate if requested. translated = audio_state.preview_text |