1. refactored wav chunk processing method

2. renamed streamlit to validation_ui
Malar Kannan 2020-05-28 11:18:39 +05:30
parent d87369c8fe
commit de21952349
3 changed files with 47 additions and 31 deletions

View File

@ -34,7 +34,8 @@ def extract_data(
events = ExtendedPath(meta_path).read_json() events = ExtendedPath(meta_path).read_json()
yield call_wav, wav_path, events yield call_wav, wav_path, events
else: else:
typer.echo(f"missing json corresponding to {wav_path}") if verbose:
typer.echo(f"missing json corresponding to {wav_path}")
def contains_asr(x): def contains_asr(x):
return "AsrResult" in x return "AsrResult" in x
@ -55,6 +56,19 @@ def extract_data(
- datetime.datetime(1900, 1, 1) - datetime.datetime(1900, 1, 1)
).total_seconds() * 1000 ).total_seconds() * 1000
def process_utterance_chunk(wav_seg, start_time, end_time, monologue):
# offset by 1sec left side to include vad? discarded audio
full_tscript_wav_seg = wav_seg[
time_to_msecs(start_time) - 1000 : time_to_msecs(end_time) # + 1000
]
tscript_wav_seg = strip_silence(full_tscript_wav_seg)
tscript_wav_fb = BytesIO()
tscript_wav_seg.export(tscript_wav_fb, format="wav")
tscript_wav = tscript_wav_fb.getvalue()
text = "".join(lens["elements"].Each()["value"].collect()(monologue))
text_clean = re.sub(r"\[.*\]", "", text)
return tscript_wav, tscript_wav_seg.duration_seconds, text_clean
def dual_asr_data_generator(wav_seg, wav_path, meta): def dual_asr_data_generator(wav_seg, wav_path, meta):
left_audio, right_audio = wav_seg.split_to_mono() left_audio, right_audio = wav_seg.split_to_mono()
channel_map = {"Agent": right_audio, "Client": left_audio} channel_map = {"Agent": right_audio, "Client": left_audio}
@ -64,7 +78,9 @@ def extract_data(
speaker_channel = channel_map.get(monologue["speaker_name"]) speaker_channel = channel_map.get(monologue["speaker_name"])
if not speaker_channel: if not speaker_channel:
if verbose: if verbose:
print(f'unknown speaker tag {monologue["speaker_name"]} in wav:{wav_path} skipping.') print(
f'unknown speaker tag {monologue["speaker_name"]} in wav:{wav_path} skipping.'
)
continue continue
try: try:
start_time = ( start_time = (
@ -81,23 +97,20 @@ def extract_data(
) )
except IndexError: except IndexError:
if verbose: if verbose:
print(f'error when loading timestamp events in wav:{wav_path} skipping.') print(
f"error when loading timestamp events in wav:{wav_path} skipping."
)
continue continue
tscript_wav, seg_dur, text_clean = process_utterance_chunk(
# offset by 500 msec to include first vad? discarded audio speaker_channel, start_time, end_time, monologue
full_tscript_wav_seg = speaker_channel[time_to_msecs(start_time) - 500 : time_to_msecs(end_time)] )
tscript_wav_seg = strip_silence(full_tscript_wav_seg) if seg_dur < 0.5:
tscript_wav_fb = BytesIO()
tscript_wav_seg.export(tscript_wav_fb, format="wav")
tscript_wav = tscript_wav_fb.getvalue()
text = "".join(lens["elements"].Each()["value"].collect()(monologue))
text_clean = re.sub(r"\[.*\]", "", text)
# only if some reasonable audio data is present yield it
if tscript_wav_seg.duration_seconds < 0.5:
if verbose: if verbose:
print(f'transcript chunk "{text_clean}" contains no audio in {wav_path} skipping.') print(
f'transcript chunk "{text_clean}" contains no audio in {wav_path} skipping.'
)
continue continue
yield text_clean, tscript_wav_seg.duration_seconds, tscript_wav yield text_clean, seg_dur, tscript_wav
def mono_asr_data_generator(wav_seg, wav_path, meta): def mono_asr_data_generator(wav_seg, wav_path, meta):
monologues = lens["monologues"].Each().collect()(meta) monologues = lens["monologues"].Each().collect()(meta)
@ -117,30 +130,33 @@ def extract_data(
) )
except IndexError: except IndexError:
if verbose: if verbose:
print(f'error when loading timestamp events in wav:{wav_path} skipping.') print(
f"error when loading timestamp events in wav:{wav_path} skipping."
)
continue continue
# offset by 500 msec to include first vad? discarded audio tscript_wav, seg_dur, text_clean = process_utterance_chunk(
full_tscript_wav_seg = wav_seg[time_to_msecs(start_time) - 500 : time_to_msecs(end_time)] wav_seg, start_time, end_time, monologue
tscript_wav_seg = strip_silence(full_tscript_wav_seg) )
tscript_wav_fb = BytesIO() if seg_dur < 0.5:
tscript_wav_seg.export(tscript_wav_fb, format="wav")
tscript_wav = tscript_wav_fb.getvalue()
text = "".join(lens["elements"].Each()["value"].collect()(monologue))
text_clean = re.sub(r"\[.*\]", "", text)
if tscript_wav_seg.duration_seconds < 0.5:
if verbose: if verbose:
print(f'transcript chunk "{text_clean}" contains no audio in {wav_path} skipping.') print(
f'transcript chunk "{text_clean}" contains no audio in {wav_path} skipping.'
)
continue continue
yield text_clean, tscript_wav_seg.duration_seconds, tscript_wav yield text_clean, seg_dur, tscript_wav
def generate_rev_asr_data(): def generate_rev_asr_data():
full_asr_data = [] full_asr_data = []
total_duration = 0 total_duration = 0
for wav, wav_path, ev in wav_event_generator(call_audio_dir): for wav, wav_path, ev in wav_event_generator(call_audio_dir):
if wav.channels > 2: if wav.channels > 2:
print(f'skipping many channel audio {wav_path}') print(f"skipping many channel audio {wav_path}")
asr_data_generator = mono_asr_data_generator if wav.channels == 1 else dual_asr_data_generator asr_data_generator = (
mono_asr_data_generator
if wav.channels == 1
else dual_asr_data_generator
)
asr_data = asr_data_generator(wav, wav_path, ev) asr_data = asr_data_generator(wav, wav_path, ev)
total_duration += wav.duration_seconds total_duration += wav.duration_seconds
full_asr_data.append(asr_data) full_asr_data.append(asr_data)

View File

@ -122,7 +122,7 @@ def dump_validation_ui_data(
), ),
) )
if annotation_only: if annotation_only:
result = pnr_data result = list(pnr_data)
else: else:
wer_key = "domain_wer" if use_domain_asr else "pretrained_wer" wer_key = "domain_wer" if use_domain_asr else "pretrained_wer"
result = sorted(pnr_data, key=lambda x: x[wer_key], reverse=True) result = sorted(pnr_data, key=lambda x: x[wer_key], reverse=True)