Commit
·
372bf52
1
Parent(s):
b18cbba
keep a test script in base directory
Browse files- src/__init__.py +0 -0
- src/whisper_streaming/whisper_online.py +2 -173
- whisper_noserver_test.py +181 -0
src/__init__.py
ADDED
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File without changes
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src/whisper_streaming/whisper_online.py
CHANGED
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@@ -5,23 +5,12 @@ import librosa
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| 5 |
from functools import lru_cache
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import time
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import logging
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| 8 |
-
from backends import FasterWhisperASR, MLXWhisper, WhisperTimestampedASR, OpenaiApiASR
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| 9 |
-
from online_asr import OnlineASRProcessor, VACOnlineASRProcessor
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| 11 |
logger = logging.getLogger(__name__)
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-
@lru_cache(10**6)
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-
def load_audio(fname):
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a, _ = librosa.load(fname, sr=16000, dtype=np.float32)
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-
return a
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-
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-
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-
def load_audio_chunk(fname, beg, end):
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audio = load_audio(fname)
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beg_s = int(beg * 16000)
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end_s = int(end * 16000)
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return audio[beg_s:end_s]
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WHISPER_LANG_CODES = "af,am,ar,as,az,ba,be,bg,bn,bo,br,bs,ca,cs,cy,da,de,el,en,es,et,eu,fa,fi,fo,fr,gl,gu,ha,haw,he,hi,hr,ht,hu,hy,id,is,it,ja,jw,ka,kk,km,kn,ko,la,lb,ln,lo,lt,lv,mg,mi,mk,ml,mn,mr,ms,mt,my,ne,nl,nn,no,oc,pa,pl,ps,pt,ro,ru,sa,sd,si,sk,sl,sn,so,sq,sr,su,sv,sw,ta,te,tg,th,tk,tl,tr,tt,uk,ur,uz,vi,yi,yo,zh".split(
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","
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@@ -244,163 +233,3 @@ def set_logging(args, logger, others=[]):
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logging.getLogger(other).setLevel(args.log_level)
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| 246 |
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| 247 |
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# logging.getLogger("whisper_online_server").setLevel(args.log_level)
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-
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| 249 |
-
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| 250 |
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if __name__ == "__main__":
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-
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| 252 |
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import argparse
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| 254 |
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parser = argparse.ArgumentParser()
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parser.add_argument(
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"--audio_path",
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type=str,
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default='samples_jfk.wav',
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help="Filename of 16kHz mono channel wav, on which live streaming is simulated.",
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)
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add_shared_args(parser)
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parser.add_argument(
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"--start_at",
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type=float,
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default=0.0,
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help="Start processing audio at this time.",
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)
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parser.add_argument(
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"--offline", action="store_true", default=False, help="Offline mode."
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)
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parser.add_argument(
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"--comp_unaware",
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action="store_true",
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default=False,
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help="Computationally unaware simulation.",
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)
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-
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args = parser.parse_args()
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-
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# reset to store stderr to different file stream, e.g. open(os.devnull,"w")
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logfile = None # sys.stderr
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-
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| 283 |
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if args.offline and args.comp_unaware:
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logger.error(
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"No or one option from --offline and --comp_unaware are available, not both. Exiting."
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)
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sys.exit(1)
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-
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| 289 |
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# if args.log_level:
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# logging.basicConfig(format='whisper-%(levelname)s:%(name)s: %(message)s',
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# level=getattr(logging, args.log_level))
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-
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| 293 |
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set_logging(args, logger,others=["online_asr"])
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-
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| 295 |
-
audio_path = args.audio_path
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-
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| 297 |
-
SAMPLING_RATE = 16000
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| 298 |
-
duration = len(load_audio(audio_path)) / SAMPLING_RATE
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-
logger.info("Audio duration is: %2.2f seconds" % duration)
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-
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| 301 |
-
asr, online = asr_factory(args, logfile=logfile)
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-
if args.vac:
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min_chunk = args.vac_chunk_size
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-
else:
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min_chunk = args.min_chunk_size
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-
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# load the audio into the LRU cache before we start the timer
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a = load_audio_chunk(audio_path, 0, 1)
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-
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# warm up the ASR because the very first transcribe takes much more time than the other
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asr.transcribe(a)
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-
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beg = args.start_at
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start = time.time() - beg
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-
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def output_transcript(o, now=None):
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# output format in stdout is like:
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# 4186.3606 0 1720 Takhle to je
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# - the first three words are:
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# - emission time from beginning of processing, in milliseconds
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# - beg and end timestamp of the text segment, as estimated by Whisper model. The timestamps are not accurate, but they're useful anyway
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# - the next words: segment transcript
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if now is None:
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now = time.time() - start
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if o[0] is not None:
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log_string = f"{now*1000:1.0f}, {o[0]*1000:1.0f}-{o[1]*1000:1.0f} ({(now-o[1]):+1.0f}s): {o[2]}"
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-
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logger.debug(
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log_string
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)
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if logfile is not None:
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print(
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log_string,
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file=logfile,
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flush=True,
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)
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else:
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# No text, so no output
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pass
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-
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| 342 |
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if args.offline: ## offline mode processing (for testing/debugging)
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a = load_audio(audio_path)
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online.insert_audio_chunk(a)
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try:
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o = online.process_iter()
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except AssertionError as e:
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logger.error(f"assertion error: {repr(e)}")
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else:
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output_transcript(o)
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now = None
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elif args.comp_unaware: # computational unaware mode
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end = beg + min_chunk
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while True:
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a = load_audio_chunk(audio_path, beg, end)
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online.insert_audio_chunk(a)
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try:
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o = online.process_iter()
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except AssertionError as e:
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logger.error(f"assertion error: {repr(e)}")
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pass
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else:
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output_transcript(o, now=end)
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-
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logger.debug(f"## last processed {end:.2f}s")
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-
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if end >= duration:
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break
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-
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beg = end
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-
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if end + min_chunk > duration:
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end = duration
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else:
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end += min_chunk
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now = duration
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-
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else: # online = simultaneous mode
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end = 0
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| 380 |
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while True:
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now = time.time() - start
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if now < end + min_chunk:
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time.sleep(min_chunk + end - now)
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end = time.time() - start
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a = load_audio_chunk(audio_path, beg, end)
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beg = end
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online.insert_audio_chunk(a)
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-
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try:
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o = online.process_iter()
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except AssertionError as e:
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logger.error(f"assertion error: {e}")
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pass
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else:
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output_transcript(o)
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now = time.time() - start
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logger.debug(
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f"## last processed {end:.2f} s, now is {now:.2f}, the latency is {now-end:.2f}"
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)
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-
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if end >= duration:
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break
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now = None
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-
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o = online.finish()
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output_transcript(o, now=now)
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from functools import lru_cache
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import time
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import logging
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+
from .backends import FasterWhisperASR, MLXWhisper, WhisperTimestampedASR, OpenaiApiASR
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+
from .online_asr import OnlineASRProcessor, VACOnlineASRProcessor
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logger = logging.getLogger(__name__)
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WHISPER_LANG_CODES = "af,am,ar,as,az,ba,be,bg,bn,bo,br,bs,ca,cs,cy,da,de,el,en,es,et,eu,fa,fi,fo,fr,gl,gu,ha,haw,he,hi,hr,ht,hu,hy,id,is,it,ja,jw,ka,kk,km,kn,ko,la,lb,ln,lo,lt,lv,mg,mi,mk,ml,mn,mr,ms,mt,my,ne,nl,nn,no,oc,pa,pl,ps,pt,ro,ru,sa,sd,si,sk,sl,sn,so,sq,sr,su,sv,sw,ta,te,tg,th,tk,tl,tr,tt,uk,ur,uz,vi,yi,yo,zh".split(
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","
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logging.getLogger(other).setLevel(args.log_level)
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whisper_noserver_test.py
ADDED
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@@ -0,0 +1,181 @@
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|
| 1 |
+
#!/usr/bin/env python3
|
| 2 |
+
import sys
|
| 3 |
+
import numpy as np
|
| 4 |
+
import librosa
|
| 5 |
+
from functools import lru_cache
|
| 6 |
+
import time
|
| 7 |
+
import logging
|
| 8 |
+
|
| 9 |
+
logger = logging.getLogger(__name__)
|
| 10 |
+
|
| 11 |
+
from src.whisper_streaming.whisper_online import *
|
| 12 |
+
|
| 13 |
+
@lru_cache(10**6)
|
| 14 |
+
def load_audio(fname):
|
| 15 |
+
a, _ = librosa.load(fname, sr=16000, dtype=np.float32)
|
| 16 |
+
return a
|
| 17 |
+
|
| 18 |
+
|
| 19 |
+
def load_audio_chunk(fname, beg, end):
|
| 20 |
+
audio = load_audio(fname)
|
| 21 |
+
beg_s = int(beg * 16000)
|
| 22 |
+
end_s = int(end * 16000)
|
| 23 |
+
return audio[beg_s:end_s]
|
| 24 |
+
|
| 25 |
+
if __name__ == "__main__":
|
| 26 |
+
|
| 27 |
+
import argparse
|
| 28 |
+
|
| 29 |
+
parser = argparse.ArgumentParser()
|
| 30 |
+
parser.add_argument(
|
| 31 |
+
"--audio_path",
|
| 32 |
+
type=str,
|
| 33 |
+
default='samples_jfk.wav',
|
| 34 |
+
help="Filename of 16kHz mono channel wav, on which live streaming is simulated.",
|
| 35 |
+
)
|
| 36 |
+
add_shared_args(parser)
|
| 37 |
+
parser.add_argument(
|
| 38 |
+
"--start_at",
|
| 39 |
+
type=float,
|
| 40 |
+
default=0.0,
|
| 41 |
+
help="Start processing audio at this time.",
|
| 42 |
+
)
|
| 43 |
+
parser.add_argument(
|
| 44 |
+
"--offline", action="store_true", default=False, help="Offline mode."
|
| 45 |
+
)
|
| 46 |
+
parser.add_argument(
|
| 47 |
+
"--comp_unaware",
|
| 48 |
+
action="store_true",
|
| 49 |
+
default=False,
|
| 50 |
+
help="Computationally unaware simulation.",
|
| 51 |
+
)
|
| 52 |
+
|
| 53 |
+
args = parser.parse_args()
|
| 54 |
+
|
| 55 |
+
# reset to store stderr to different file stream, e.g. open(os.devnull,"w")
|
| 56 |
+
logfile = None # sys.stderr
|
| 57 |
+
|
| 58 |
+
if args.offline and args.comp_unaware:
|
| 59 |
+
logger.error(
|
| 60 |
+
"No or one option from --offline and --comp_unaware are available, not both. Exiting."
|
| 61 |
+
)
|
| 62 |
+
sys.exit(1)
|
| 63 |
+
|
| 64 |
+
# if args.log_level:
|
| 65 |
+
# logging.basicConfig(format='whisper-%(levelname)s:%(name)s: %(message)s',
|
| 66 |
+
# level=getattr(logging, args.log_level))
|
| 67 |
+
|
| 68 |
+
set_logging(args, logger,others=["src.whisper_streaming.online_asr"])
|
| 69 |
+
|
| 70 |
+
audio_path = args.audio_path
|
| 71 |
+
|
| 72 |
+
SAMPLING_RATE = 16000
|
| 73 |
+
duration = len(load_audio(audio_path)) / SAMPLING_RATE
|
| 74 |
+
logger.info("Audio duration is: %2.2f seconds" % duration)
|
| 75 |
+
|
| 76 |
+
asr, online = asr_factory(args, logfile=logfile)
|
| 77 |
+
if args.vac:
|
| 78 |
+
min_chunk = args.vac_chunk_size
|
| 79 |
+
else:
|
| 80 |
+
min_chunk = args.min_chunk_size
|
| 81 |
+
|
| 82 |
+
# load the audio into the LRU cache before we start the timer
|
| 83 |
+
a = load_audio_chunk(audio_path, 0, 1)
|
| 84 |
+
|
| 85 |
+
# warm up the ASR because the very first transcribe takes much more time than the other
|
| 86 |
+
asr.transcribe(a)
|
| 87 |
+
|
| 88 |
+
beg = args.start_at
|
| 89 |
+
start = time.time() - beg
|
| 90 |
+
|
| 91 |
+
def output_transcript(o, now=None):
|
| 92 |
+
# output format in stdout is like:
|
| 93 |
+
# 4186.3606 0 1720 Takhle to je
|
| 94 |
+
# - the first three words are:
|
| 95 |
+
# - emission time from beginning of processing, in milliseconds
|
| 96 |
+
# - beg and end timestamp of the text segment, as estimated by Whisper model. The timestamps are not accurate, but they're useful anyway
|
| 97 |
+
# - the next words: segment transcript
|
| 98 |
+
if now is None:
|
| 99 |
+
now = time.time() - start
|
| 100 |
+
if o[0] is not None:
|
| 101 |
+
log_string = f"{now*1000:1.0f}, {o[0]*1000:1.0f}-{o[1]*1000:1.0f} ({(now-o[1]):+1.0f}s): {o[2]}"
|
| 102 |
+
|
| 103 |
+
logger.debug(
|
| 104 |
+
log_string
|
| 105 |
+
)
|
| 106 |
+
|
| 107 |
+
if logfile is not None:
|
| 108 |
+
print(
|
| 109 |
+
log_string,
|
| 110 |
+
file=logfile,
|
| 111 |
+
flush=True,
|
| 112 |
+
)
|
| 113 |
+
else:
|
| 114 |
+
# No text, so no output
|
| 115 |
+
pass
|
| 116 |
+
|
| 117 |
+
if args.offline: ## offline mode processing (for testing/debugging)
|
| 118 |
+
a = load_audio(audio_path)
|
| 119 |
+
online.insert_audio_chunk(a)
|
| 120 |
+
try:
|
| 121 |
+
o = online.process_iter()
|
| 122 |
+
except AssertionError as e:
|
| 123 |
+
logger.error(f"assertion error: {repr(e)}")
|
| 124 |
+
else:
|
| 125 |
+
output_transcript(o)
|
| 126 |
+
now = None
|
| 127 |
+
elif args.comp_unaware: # computational unaware mode
|
| 128 |
+
end = beg + min_chunk
|
| 129 |
+
while True:
|
| 130 |
+
a = load_audio_chunk(audio_path, beg, end)
|
| 131 |
+
online.insert_audio_chunk(a)
|
| 132 |
+
try:
|
| 133 |
+
o = online.process_iter()
|
| 134 |
+
except AssertionError as e:
|
| 135 |
+
logger.error(f"assertion error: {repr(e)}")
|
| 136 |
+
pass
|
| 137 |
+
else:
|
| 138 |
+
output_transcript(o, now=end)
|
| 139 |
+
|
| 140 |
+
logger.debug(f"## last processed {end:.2f}s")
|
| 141 |
+
|
| 142 |
+
if end >= duration:
|
| 143 |
+
break
|
| 144 |
+
|
| 145 |
+
beg = end
|
| 146 |
+
|
| 147 |
+
if end + min_chunk > duration:
|
| 148 |
+
end = duration
|
| 149 |
+
else:
|
| 150 |
+
end += min_chunk
|
| 151 |
+
now = duration
|
| 152 |
+
|
| 153 |
+
else: # online = simultaneous mode
|
| 154 |
+
end = 0
|
| 155 |
+
while True:
|
| 156 |
+
now = time.time() - start
|
| 157 |
+
if now < end + min_chunk:
|
| 158 |
+
time.sleep(min_chunk + end - now)
|
| 159 |
+
end = time.time() - start
|
| 160 |
+
a = load_audio_chunk(audio_path, beg, end)
|
| 161 |
+
beg = end
|
| 162 |
+
online.insert_audio_chunk(a)
|
| 163 |
+
|
| 164 |
+
try:
|
| 165 |
+
o = online.process_iter()
|
| 166 |
+
except AssertionError as e:
|
| 167 |
+
logger.error(f"assertion error: {e}")
|
| 168 |
+
pass
|
| 169 |
+
else:
|
| 170 |
+
output_transcript(o)
|
| 171 |
+
now = time.time() - start
|
| 172 |
+
logger.debug(
|
| 173 |
+
f"## last processed {end:.2f} s, now is {now:.2f}, the latency is {now-end:.2f}"
|
| 174 |
+
)
|
| 175 |
+
|
| 176 |
+
if end >= duration:
|
| 177 |
+
break
|
| 178 |
+
now = None
|
| 179 |
+
|
| 180 |
+
o = online.finish()
|
| 181 |
+
output_transcript(o, now=now)
|