jasper-asr/jasper/data/rev_recycler.py

183 lines
7.2 KiB
Python

import typer
from itertools import chain
from io import BytesIO
from pathlib import Path
import re
app = typer.Typer()
@app.command()
def extract_data(
call_audio_dir: Path = typer.Option(Path("/dataset/rev/wavs"), show_default=True),
call_meta_dir: Path = typer.Option(Path("/dataset/rev/jsons"), show_default=True),
output_dir: Path = typer.Option(Path("./data"), show_default=True),
dataset_name: str = typer.Option("rev_transribed", show_default=True),
verbose: bool = False,
):
from pydub import AudioSegment
from .utils import ExtendedPath, asr_data_writer, strip_silence
from lenses import lens
import datetime
call_asr_data: Path = output_dir / Path("asr_data")
call_asr_data.mkdir(exist_ok=True, parents=True)
def wav_event_generator(call_audio_dir):
for wav_path in call_audio_dir.glob("**/*.wav"):
if verbose:
typer.echo(f"loading events for file {wav_path}")
call_wav = AudioSegment.from_file_using_temporary_files(wav_path)
rel_meta_path = wav_path.with_suffix(".json").relative_to(call_audio_dir)
meta_path = call_meta_dir / rel_meta_path
if meta_path.exists():
events = ExtendedPath(meta_path).read_json()
yield call_wav, wav_path, events
else:
typer.echo(f"missing json corresponding to {wav_path}")
def contains_asr(x):
return "AsrResult" in x
def channel(n):
def filter_func(ev):
return (
ev["AsrResult"]["Channel"] == n
if "Channel" in ev["AsrResult"]
else n == 0
)
return filter_func
# def compute_endtime(call_wav, state):
# for (i, st) in enumerate(state):
# start_time = st["AsrResult"]["Alternatives"][0].get("StartTime", 0)
# transcript = st["AsrResult"]["Alternatives"][0]["Transcript"]
# if i + 1 < len(state):
# end_time = state[i + 1]["AsrResult"]["Alternatives"][0]["StartTime"]
# else:
# end_time = call_wav.duration_seconds
# full_code_seg = call_wav[start_time * 1000 : end_time * 1000]
# code_seg = strip_silence(full_code_seg)
# code_fb = BytesIO()
# code_seg.export(code_fb, format="wav")
# code_wav = code_fb.getvalue()
# # only starting 1 min audio has reliable alignment
# if start_time > 60:
# break
# # only of some audio data is present yield it
# if code_seg.duration_seconds >= 0.5:
# yield transcript, code_seg.duration_seconds, code_wav
# def generate_call_asr_data():
# full_asr_data = []
# total_duration = 0
# for wav, wav_path, ev in wav_event_generator(call_audio_dir):
# asr_data = asr_data_generator(wav, wav_path, ev)
# total_duration += wav.duration_seconds
# full_asr_data.append(asr_data)
# typer.echo(f"loaded {len(full_asr_data)} calls of duration {total_duration}s")
# n_dps = asr_data_writer(call_asr_data, dataset_name, chain(*full_asr_data))
# typer.echo(f"written {n_dps} data points")
# generate_call_asr_data()
def time_to_msecs(time_str):
return (
datetime.datetime.strptime(time_str, "%H:%M:%S,%f")
- datetime.datetime(1900, 1, 1)
).total_seconds() * 1000
def asr_data_generator(wav_seg, wav_path, meta):
left_audio, right_audio = wav_seg.split_to_mono()
channel_map = {"Agent": right_audio, "Client": left_audio}
monologues = lens["monologues"].Each().collect()(meta)
for monologue in monologues:
# print(monologue["speaker_name"])
speaker_channel = channel_map.get(monologue["speaker_name"])
if not speaker_channel:
print(f'unknown speaker tag {monologue["speaker_name"]} in wav:{wav_path} skipping.')
continue
try:
start_time = (
lens["elements"]
.Each()
.Filter(lambda x: "timestamp" in x)["timestamp"]
.collect()(monologue)[0]
)
end_time = (
lens["elements"]
.Each()
.Filter(lambda x: "end_timestamp" in x)["end_timestamp"]
.collect()(monologue)[-1]
)
except IndexError:
print(f'error when loading timestamp events in wav:{wav_path} skipping.')
# offset by 500 msec to include first vad? discarded audio
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)
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)
yield text_clean, tscript_wav_seg.duration_seconds, tscript_wav
def generate_rev_asr_data():
full_asr_data = []
total_duration = 0
for wav, wav_path, ev in wav_event_generator(call_audio_dir):
asr_data = asr_data_generator(wav, wav_path, ev)
total_duration += wav.duration_seconds
full_asr_data.append(asr_data)
typer.echo(f"loaded {len(full_asr_data)} calls of duration {total_duration}s")
n_dps = asr_data_writer(call_asr_data, dataset_name, chain(*full_asr_data))
typer.echo(f"written {n_dps} data points")
generate_rev_asr_data()
# DEBUG
# data = list(wav_event_generator(call_audio_dir))
# wav_seg, wav_path, meta = data[0]
# left_audio, right_audio = wav_seg.split_to_mono()
# channel_map = {"Agent": right_audio, "Client": left_audio}
# # data[0][2]['speakers']
# # data[0][1]
# monologues = lens["monologues"].Each().collect()(meta)
# for monologue in monologues:
# # print(monologue["speaker_name"])
# speaker_channel = channel_map.get(monologue["speaker_name"])
# # monologue = monologues[0]
# # monologue
# start_time = (
# lens["elements"]
# .Each()
# .Filter(lambda x: "timestamp" in x)["timestamp"]
# .collect()(monologue)[0]
# )
# end_time = (
# lens["elements"]
# .Each()
# .Filter(lambda x: "end_timestamp" in x)["end_timestamp"]
# .collect()(monologue)[-1]
# )
# start_time, end_time
#
# # offset by 500 msec to include first vad? discarded audio
# speaker_channel[time_to_msecs(start_time) - 500 : time_to_msecs(end_time)]
#
# # start_time = lens["elements"][0].get()(monologue)['timestamp']
# # end_time = lens["elements"][-1].get()(monologue)['timestamp']
# text = "".join(lens["elements"].Each()["value"].collect()(monologue))
# text_clean = re.sub(r"\[.*\]", "", text)
# # print(text)
# # print(text_clean)
def main():
app()
if __name__ == "__main__":
main()