예제 #1
0
def hyp(input_path, output_path, device, batch_size, html, ext, sample_rate,
        max_duration):

    os.makedirs(output_path, exist_ok=True)
    audio_source = ([
        (input_path, audio_name) for audio_name in os.listdir(input_path)
    ] if os.path.isdir(input_path) else [(os.path.dirname(input_path),
                                          os.path.basename(input_path))])
    model = PyannoteDiarizationModel(device=device, batch_size=batch_size)
    for i, (input_path, audio_name) in enumerate(audio_source):
        print(i, '/', len(audio_source), audio_name)
        audio_path = os.path.join(input_path, audio_name)
        noextname = audio_name[:-len(ext)]
        transcript_path = os.path.join(output_path, noextname + '.json')
        rttm_path = os.path.join(output_path, noextname + '.rttm')

        signal, sample_rate = audio.read_audio(audio_path,
                                               sample_rate=sample_rate,
                                               mono=True,
                                               dtype='float32',
                                               duration=max_duration)
        transcript = model(signal,
                           sample_rate=sample_rate,
                           extra=dict(audio_path=audio_path))
        transcripts.collect_speaker_names(transcript, set_speaker_data=True)

        transcripts.save(transcript_path, transcript)
        print(transcript_path)

        transcripts.save(rttm_path, transcript)
        print(rttm_path)

        if html:
            html_path = os.path.join(output_path, audio_name + '.html')
            vis.transcript(html_path,
                           sample_rate=sample_rate,
                           mono=True,
                           transcript=transcript,
                           duration=max_duration)
예제 #2
0
def main(args, ext_json=['.json', '.json.gz']):
    utils.enable_jit_fusion()

    assert args.output_json or args.output_html or args.output_txt or args.output_csv, \
     'at least one of the output formats must be provided'
    os.makedirs(args.output_path, exist_ok=True)

    audio_data_paths = set(
        p for f in args.input_path
        for p in ([os.path.join(f, g)
                   for g in os.listdir(f)] if os.path.isdir(f) else [f])
        if os.path.isfile(p) and any(map(p.endswith, args.ext)))
    json_data_paths = set(
        p for p in args.input_path if any(map(p.endswith, ext_json))
        and not utils.strip_suffixes(p, ext_json) in audio_data_paths)

    data_paths = list(audio_data_paths | json_data_paths)

    exclude = set([
        os.path.splitext(basename)[0]
        for basename in os.listdir(args.output_path)
        if basename.endswith('.json')
    ]) if args.skip_processed else None

    data_paths = [
        path for path in data_paths
        if exclude is None or os.path.basename(path) not in exclude
    ]

    text_pipeline, frontend, model, generator = setup(args)
    val_dataset = datasets.AudioTextDataset(
        data_paths, [text_pipeline],
        args.sample_rate,
        frontend=frontend if not args.frontend_in_model else None,
        mono=args.mono,
        time_padding_multiple=args.batch_time_padding_multiple,
        audio_backend=args.audio_backend,
        exclude=exclude,
        max_duration=args.transcribe_first_n_sec,
        mode='batched_channels'
        if args.join_transcript else 'batched_transcript',
        string_array_encoding=args.dataset_string_array_encoding,
        debug_short_long_records_features_from_whole_normalized_signal=args.
        debug_short_long_records_features_from_whole_normalized_signal)
    print('Examples count: ', len(val_dataset))
    val_meta = val_dataset.pop_meta()
    val_data_loader = torch.utils.data.DataLoader(
        val_dataset,
        batch_size=None,
        collate_fn=val_dataset.collate_fn,
        num_workers=args.num_workers)
    csv_sep = dict(tab='\t', comma=',')[args.csv_sep]
    csv_lines = []  # only used if args.output_csv is True

    oom_handler = utils.OomHandler(max_retries=args.oom_retries)
    for i, (meta, s, x, xlen, y, ylen) in enumerate(val_data_loader):
        print(f'Processing: {i}/{len(val_dataset)}')
        meta = [val_meta[t['example_id']] for t in meta]

        audio_path = meta[0]['audio_path']
        audio_name = transcripts.audio_name(audio_path)
        begin_end = [dict(begin=t['begin'], end=t['end']) for t in meta]
        begin = torch.tensor([t['begin'] for t in begin_end],
                             dtype=torch.float)
        end = torch.tensor([t['end'] for t in begin_end], dtype=torch.float)
        #TODO WARNING assumes frontend not in dataset
        if not args.frontend_in_model:
            print('\n' * 10 + 'WARNING\n' * 5)
            print(
                'transcribe.py assumes frontend in model, in other case time alignment was incorrect'
            )
            print('WARNING\n' * 5 + '\n')

        duration = x.shape[-1] / args.sample_rate
        channel = [t['channel'] for t in meta]
        speaker = [t['speaker'] for t in meta]
        speaker_name = [t['speaker_name'] for t in meta]

        if x.numel() == 0:
            print(f'Skipping empty [{audio_path}].')
            continue

        try:
            tic = time.time()
            y, ylen = y.to(args.device), ylen.to(args.device)
            log_probs, logits, olen = model(
                x.squeeze(1).to(args.device), xlen.to(args.device))

            print('Input:', audio_name)
            print('Input time steps:', log_probs.shape[-1],
                  '| target time steps:', y.shape[-1])
            print(
                'Time: audio {audio:.02f} sec | processing {processing:.02f} sec'
                .format(audio=sum(map(transcripts.compute_duration, meta)),
                        processing=time.time() - tic))

            ts: shaping.Bt = duration * torch.linspace(
                0, 1, steps=log_probs.shape[-1],
                device=log_probs.device).unsqueeze(0).expand(x.shape[0], -1)

            ref_segments = [[
                dict(channel=channel[i],
                     begin=begin_end[i]['begin'],
                     end=begin_end[i]['end'],
                     ref=text_pipeline.postprocess(
                         text_pipeline.preprocess(meta[i]['ref'])))
            ] for i in range(len(meta))]
            hyp_segments = [
                alternatives[0] for alternatives in generator.generate(
                    tokenizer=text_pipeline.tokenizer,
                    log_probs=log_probs,
                    begin=begin,
                    end=end,
                    output_lengths=olen,
                    time_stamps=ts,
                    segment_text_key='hyp',
                    segment_extra_info=[
                        dict(speaker=s, speaker_name=sn, channel=c)
                        for s, sn, c in zip(speaker, speaker_name, channel)
                    ])
            ]
            hyp_segments = [
                transcripts.map_text(text_pipeline.postprocess, hyp=hyp)
                for hyp in hyp_segments
            ]
            hyp, ref = '\n'.join(
                transcripts.join(hyp=h)
                for h in hyp_segments).strip(), '\n'.join(
                    transcripts.join(ref=r) for r in ref_segments).strip()
            if args.verbose:
                print('HYP:', hyp)
            print('CER: {cer:.02%}'.format(cer=metrics.cer(hyp=hyp, ref=ref)))

            tic_alignment = time.time()
            if args.align and y.numel() > 0:
                alignment: shaping.BY = ctc.alignment(
                    log_probs.permute(2, 0, 1),
                    y[:, 0, :],  # assumed that 0 channel is char labels
                    olen,
                    ylen[:, 0],
                    blank=text_pipeline.tokenizer.eps_id,
                    pack_backpointers=args.pack_backpointers)
                aligned_ts: shaping.Bt = ts.gather(1, alignment)

                ref_segments = [
                    alternatives[0] for alternatives in generator.generate(
                        tokenizer=text_pipeline.tokenizer,
                        log_probs=torch.nn.functional.one_hot(
                            y[:,
                              0, :], num_classes=log_probs.shape[1]).permute(
                                  0, 2, 1),
                        begin=begin,
                        end=end,
                        output_lengths=ylen,
                        time_stamps=aligned_ts,
                        segment_text_key='ref',
                        segment_extra_info=[
                            dict(speaker=s, speaker_name=sn, channel=c)
                            for s, sn, c in zip(speaker, speaker_name, channel)
                        ])
                ]
                ref_segments = [
                    transcripts.map_text(text_pipeline.postprocess, hyp=ref)
                    for ref in ref_segments
                ]
            oom_handler.reset()
        except:
            if oom_handler.try_recover(model.parameters()):
                print(f'Skipping {i} / {len(val_dataset)}')
                continue
            else:
                raise

        print('Alignment time: {:.02f} sec'.format(time.time() -
                                                   tic_alignment))

        ref_transcript, hyp_transcript = [
            sorted(transcripts.flatten(segments), key=transcripts.sort_key)
            for segments in [ref_segments, hyp_segments]
        ]

        if args.max_segment_duration:
            if ref:
                ref_segments = list(
                    transcripts.segment_by_time(ref_transcript,
                                                args.max_segment_duration))
                hyp_segments = list(
                    transcripts.segment_by_ref(hyp_transcript, ref_segments))
            else:
                hyp_segments = list(
                    transcripts.segment_by_time(hyp_transcript,
                                                args.max_segment_duration))
                ref_segments = [[] for _ in hyp_segments]

        #### HACK for diarization
        elif args.ref_transcript_path and args.join_transcript:
            audio_name_hack = audio_name.split('.')[0]
            #TODO: normalize ref field
            ref_segments = [[t] for t in sorted(transcripts.load(
                os.path.join(args.ref_transcript_path, audio_name_hack +
                             '.json')),
                                                key=transcripts.sort_key)]
            hyp_segments = list(
                transcripts.segment_by_ref(hyp_transcript,
                                           ref_segments,
                                           set_speaker=True,
                                           soft=False))
        #### END OF HACK

        has_ref = bool(transcripts.join(ref=transcripts.flatten(ref_segments)))

        transcript = []
        for hyp_transcript, ref_transcript in zip(hyp_segments, ref_segments):
            hyp, ref = transcripts.join(hyp=hyp_transcript), transcripts.join(
                ref=ref_transcript)

            transcript.append(
                dict(audio_path=audio_path,
                     ref=ref,
                     hyp=hyp,
                     speaker_name=transcripts.speaker_name(ref=ref_transcript,
                                                           hyp=hyp_transcript),
                     words=metrics.align_words(
                         *metrics.align_strings(hyp=hyp, ref=ref))
                     if args.align_words else [],
                     words_ref=ref_transcript,
                     words_hyp=hyp_transcript,
                     **transcripts.summary(hyp_transcript),
                     **(dict(cer=metrics.cer(hyp=hyp, ref=ref))
                        if has_ref else {})))

        transcripts.collect_speaker_names(transcript,
                                          set_speaker_data=True,
                                          num_speakers=2)

        filtered_transcript = list(
            transcripts.prune(transcript,
                              align_boundary_words=args.align_boundary_words,
                              cer=args.prune_cer,
                              duration=args.prune_duration,
                              gap=args.prune_gap,
                              allowed_unk_count=args.prune_unk,
                              num_speakers=args.prune_num_speakers))

        print('Filtered segments:', len(filtered_transcript), 'out of',
              len(transcript))

        if args.output_json:
            transcript_path = os.path.join(args.output_path,
                                           audio_name + '.json')
            print(transcripts.save(transcript_path, filtered_transcript))

        if args.output_html:
            transcript_path = os.path.join(args.output_path,
                                           audio_name + '.html')
            print(
                vis.transcript(transcript_path, args.sample_rate, args.mono,
                               transcript, filtered_transcript))

        if args.output_txt:
            transcript_path = os.path.join(args.output_path,
                                           audio_name + '.txt')
            with open(transcript_path, 'w') as f:
                f.write(' '.join(t['hyp'].strip()
                                 for t in filtered_transcript))
            print(transcript_path)

        if args.output_csv:
            assert len({t['audio_path'] for t in filtered_transcript}) == 1
            audio_path = filtered_transcript[0]['audio_path']
            hyp = ' '.join(t['hyp'].strip() for t in filtered_transcript)
            begin = min(t['begin'] for t in filtered_transcript)
            end = max(t['end'] for t in filtered_transcript)
            csv_lines.append(
                csv_sep.join([audio_path, hyp,
                              str(begin),
                              str(end)]))

        if args.logits:
            logits_file_path = os.path.join(args.output_path,
                                            audio_name + '.pt')
            if args.logits_crop:
                begin_end = [
                    dict(
                        zip(['begin', 'end'], [
                            t['begin'] + c / float(o) * (t['end'] - t['begin'])
                            for c in args.logits_crop
                        ])) for o, t in zip(olen, begin_end)
                ]
                logits_crop = [slice(*args.logits_crop) for o in olen]
            else:
                logits_crop = [slice(int(o)) for o in olen]

            # TODO: filter ref / hyp by channel?
            torch.save([
                dict(audio_path=audio_path,
                     logits=l[..., logits_crop[i]],
                     **begin_end[i],
                     ref=ref,
                     hyp=hyp) for i, l in enumerate(logits.cpu())
            ], logits_file_path)
            print(logits_file_path)

        print('Done: {:.02f} sec\n'.format(time.time() - tic))

    if args.output_csv:
        transcript_path = os.path.join(args.output_path, 'transcripts.csv')
        with open(transcript_path, 'w') as f:
            f.write('\n'.join(csv_lines))
        print(transcript_path)
예제 #3
0
def main(args):
    utils.enable_jit_fusion()

    assert args.output_json or args.output_html or args.output_txt or args.output_csv, \
     'at least one of the output formats must be provided'
    os.makedirs(args.output_path, exist_ok=True)
    data_paths = [
        p for f in args.input_path
        for p in ([os.path.join(f, g)
                   for g in os.listdir(f)] if os.path.isdir(f) else [f])
        if os.path.isfile(p) and any(map(p.endswith, args.ext))
    ] + [
        p
        for p in args.input_path if any(map(p.endswith, ['.json', '.json.gz']))
    ]
    exclude = set([
        os.path.splitext(basename)[0]
        for basename in os.listdir(args.output_path)
        if basename.endswith('.json')
    ] if args.skip_processed else [])
    data_paths = [
        path for path in data_paths if os.path.basename(path) not in exclude
    ]

    labels, frontend, model, decoder = setup(args)
    val_dataset = datasets.AudioTextDataset(
        data_paths, [labels],
        args.sample_rate,
        frontend=None,
        segmented=True,
        mono=args.mono,
        time_padding_multiple=args.batch_time_padding_multiple,
        audio_backend=args.audio_backend,
        exclude=exclude,
        max_duration=args.transcribe_first_n_sec,
        join_transcript=args.join_transcript,
        string_array_encoding=args.dataset_string_array_encoding)
    num_examples = len(val_dataset)
    print('Examples count: ', num_examples)
    val_data_loader = torch.utils.data.DataLoader(
        val_dataset,
        batch_size=None,
        collate_fn=val_dataset.collate_fn,
        num_workers=args.num_workers)
    csv_sep = dict(tab='\t', comma=',')[args.csv_sep]
    output_lines = []  # only used if args.output_csv is True

    oom_handler = utils.OomHandler(max_retries=args.oom_retries)
    for i, (meta, s, x, xlen, y, ylen) in enumerate(val_data_loader):
        print(f'Processing: {i}/{num_examples}')

        meta = [val_dataset.meta.get(m['example_id']) for m in meta]
        audio_path = meta[0]['audio_path']

        if x.numel() == 0:
            print(f'Skipping empty [{audio_path}].')
            continue

        begin = meta[0]['begin']
        end = meta[0]['end']
        audio_name = transcripts.audio_name(audio_path)

        try:
            tic = time.time()
            y, ylen = y.to(args.device), ylen.to(args.device)
            log_probs, olen = model(
                x.squeeze(1).to(args.device), xlen.to(args.device))

            #speech = vad.detect_speech(x.squeeze(1), args.sample_rate, args.window_size, aggressiveness = args.vad, window_size_dilate = args.window_size_dilate)
            #speech = vad.upsample(speech, log_probs)
            #log_probs.masked_fill_(models.silence_space_mask(log_probs, speech, space_idx = labels.space_idx, blank_idx = labels.blank_idx), float('-inf'))

            decoded = decoder.decode(log_probs, olen)

            print('Input:', audio_name)
            print('Input time steps:', log_probs.shape[-1],
                  '| target time steps:', y.shape[-1])
            print(
                'Time: audio {audio:.02f} sec | processing {processing:.02f} sec'
                .format(audio=sum(
                    transcripts.compute_duration(t) for t in meta),
                        processing=time.time() - tic))

            ts = (x.shape[-1] / args.sample_rate) * torch.linspace(
                0, 1,
                steps=log_probs.shape[-1]).unsqueeze(0) + torch.FloatTensor(
                    [t['begin'] for t in meta]).unsqueeze(1)
            channel = [t['channel'] for t in meta]
            speaker = [t['speaker'] for t in meta]
            ref_segments = [[
                dict(channel=channel[i],
                     begin=meta[i]['begin'],
                     end=meta[i]['end'],
                     ref=labels.decode(y[i, 0, :ylen[i]].tolist()))
            ] for i in range(len(decoded))]
            hyp_segments = [
                labels.decode(decoded[i],
                              ts[i],
                              channel=channel[i],
                              replace_blank=True,
                              replace_blank_series=args.replace_blank_series,
                              replace_repeat=True,
                              replace_space=False,
                              speaker=speaker[i] if isinstance(
                                  speaker[i], str) else None)
                for i in range(len(decoded))
            ]

            ref, hyp = '\n'.join(
                transcripts.join(ref=r)
                for r in ref_segments).strip(), '\n'.join(
                    transcripts.join(hyp=h) for h in hyp_segments).strip()
            if args.verbose:
                print('HYP:', hyp)
            print('CER: {cer:.02%}'.format(cer=metrics.cer(hyp=hyp, ref=ref)))

            tic_alignment = time.time()
            if args.align and y.numel() > 0:
                #if ref_full:# and not ref:
                #	#assert len(set(t['channel'] for t in meta)) == 1 or all(t['type'] != 'channel' for t in meta)
                #	#TODO: add space at the end
                #	channel = torch.ByteTensor(channel).repeat_interleave(log_probs.shape[-1]).reshape(1, -1)
                #	ts = ts.reshape(1, -1)
                #	log_probs = log_probs.transpose(0, 1).unsqueeze(0).flatten(start_dim = -2)
                #	olen = torch.tensor([log_probs.shape[-1]], device = log_probs.device, dtype = torch.long)
                #	y = y_full[None, None, :].to(y.device)
                #	ylen = torch.tensor([[y.shape[-1]]], device = log_probs.device, dtype = torch.long)
                #	segments = [([], sum([h for r, h in segments], []))]

                alignment = ctc.alignment(
                    log_probs.permute(2, 0, 1),
                    y.squeeze(1),
                    olen,
                    ylen.squeeze(1),
                    blank=labels.blank_idx,
                    pack_backpointers=args.pack_backpointers)
                ref_segments = [
                    labels.decode(y[i, 0, :ylen[i]].tolist(),
                                  ts[i],
                                  alignment[i],
                                  channel=channel[i],
                                  speaker=speaker[i],
                                  key='ref',
                                  speakers=val_dataset.speakers)
                    for i in range(len(decoded))
                ]
            oom_handler.reset()
        except:
            if oom_handler.try_recover(model.parameters()):
                print(f'Skipping {i} / {num_examples}')
                continue
            else:
                raise

        print('Alignment time: {:.02f} sec'.format(time.time() -
                                                   tic_alignment))

        if args.max_segment_duration:
            ref_transcript, hyp_transcript = [
                list(sorted(sum(segments, []), key=transcripts.sort_key))
                for segments in [ref_segments, hyp_segments]
            ]
            if ref:
                ref_segments = list(
                    transcripts.segment(ref_transcript,
                                        args.max_segment_duration))
                hyp_segments = list(
                    transcripts.segment(hyp_transcript, ref_segments))
            else:
                hyp_segments = list(
                    transcripts.segment(hyp_transcript,
                                        args.max_segment_duration))
                ref_segments = [[] for _ in hyp_segments]

        transcript = [
            dict(audio_path=audio_path,
                 ref=ref,
                 hyp=hyp,
                 speaker=transcripts.speaker(ref=ref_transcript,
                                             hyp=hyp_transcript),
                 cer=metrics.cer(hyp=hyp, ref=ref),
                 words=metrics.align_words(hyp=hyp, ref=ref)[-1]
                 if args.align_words else [],
                 alignment=dict(ref=ref_transcript, hyp=hyp_transcript),
                 **transcripts.summary(hyp_transcript)) for ref_transcript,
            hyp_transcript in zip(ref_segments, hyp_segments)
            for ref, hyp in [(transcripts.join(ref=ref_transcript),
                              transcripts.join(hyp=hyp_transcript))]
        ]
        filtered_transcript = list(
            transcripts.prune(transcript,
                              align_boundary_words=args.align_boundary_words,
                              cer=args.cer,
                              duration=args.duration,
                              gap=args.gap,
                              unk=args.unk,
                              num_speakers=args.num_speakers))

        print('Filtered segments:', len(filtered_transcript), 'out of',
              len(transcript))

        if args.output_json:
            transcript_path = os.path.join(args.output_path,
                                           audio_name + '.json')
            print(transcript_path)
            with open(transcript_path, 'w') as f:
                json.dump(filtered_transcript,
                          f,
                          ensure_ascii=False,
                          sort_keys=True,
                          indent=2)

        if args.output_html:
            transcript_path = os.path.join(args.output_path,
                                           audio_name + '.html')
            print(transcript_path)
            vis.transcript(transcript_path, args.sample_rate, args.mono,
                           transcript, filtered_transcript)

        if args.output_txt:
            transcript_path = os.path.join(args.output_path,
                                           audio_name + '.txt')
            print(transcript_path)
            with open(transcript_path, 'w') as f:
                f.write(hyp)

        if args.output_csv:
            output_lines.append(
                csv_sep.join((audio_path, hyp, str(begin), str(end))) + '\n')

        print('Done: {:.02f} sec\n'.format(time.time() - tic))

    if args.output_csv:
        with open(os.path.join(args.output_path, 'transcripts.csv'), 'w') as f:
            f.writelines(output_lines)
예제 #4
0
def ref(input_path, output_path, sample_rate, window_size, device,
        max_duration, debug_audio, html, ext):
    os.makedirs(output_path, exist_ok=True)
    audio_source = ([
        (input_path, audio_name) for audio_name in os.listdir(input_path)
    ] if os.path.isdir(input_path) else [(os.path.dirname(input_path),
                                          os.path.basename(input_path))])
    for i, (input_path, audio_name) in enumerate(audio_source):
        print(i, '/', len(audio_source), audio_name)
        audio_path = os.path.join(input_path, audio_name)
        noextname = audio_name[:-len(ext)]
        transcript_path = os.path.join(output_path, noextname + '.json')
        rttm_path = os.path.join(output_path, noextname + '.rttm')

        signal, sample_rate = audio.read_audio(audio_path,
                                               sample_rate=sample_rate,
                                               mono=False,
                                               dtype='float32',
                                               duration=max_duration)

        speaker_id_ref, speaker_id_ref_ = select_speaker(
            signal.to(device),
            silence_absolute_threshold=0.05,
            silence_relative_threshold=0.2,
            kernel_size_smooth_signal=128,
            kernel_size_smooth_speaker=4096,
            kernel_size_smooth_silence=4096)

        transcript = [
            dict(audio_path=audio_path,
                 begin=float(begin) / sample_rate,
                 end=(float(begin) + float(duration)) / sample_rate,
                 speaker=speaker,
                 speaker_name=transcripts.default_speaker_names[speaker])
            for speaker in range(1, len(speaker_id_ref_))
            for begin, duration, mask in zip(
                *models.rle1d(speaker_id_ref_[speaker])) if mask == 1
        ]

        #transcript = [dict(audio_path = audio_path, begin = float(begin) / sample_rate, end = (float(begin) + float(duration)) / sample_rate, speaker_name = str(int(speaker)), speaker = int(speaker)) for begin, duration, speaker in zip(*models.rle1d(speaker_id_ref.cpu()))]

        transcript_without_speaker_missing = [
            t for t in transcript
            if t['speaker'] != transcripts.speaker_missing
        ]
        transcripts.save(transcript_path, transcript_without_speaker_missing)
        print(transcript_path)

        transcripts.save(rttm_path, transcript_without_speaker_missing)
        print(rttm_path)

        if debug_audio:
            audio.write_audio(
                transcript_path + '.wav',
                torch.cat([
                    signal[..., :speaker_id_ref.shape[-1]],
                    convert_speaker_id(speaker_id_ref[..., :signal.shape[-1]],
                                       to_bipole=True).unsqueeze(0).cpu() *
                    0.5, speaker_id_ref_[..., :signal.shape[-1]].cpu() * 0.5
                ]),
                sample_rate,
                mono=False)
            print(transcript_path + '.wav')

        if html:
            html_path = os.path.join(output_path, audio_name + '.html')
            vis.transcript(html_path,
                           sample_rate=sample_rate,
                           mono=True,
                           transcript=transcript,
                           duration=max_duration)
예제 #5
0
audio_source = ([
    (args.input_path, audio_name) for audio_name in os.listdir(args.input_path)
] if os.path.isdir(args.input_path) else [(os.path.dirname(args.input_path),
                                           os.path.basename(args.input_path))])

for i, (input_path, audio_name) in enumerate(audio_source):
    print(i, '/', len(audio_source), audio_name)

    audio_path = os.path.join(input_path, audio_name)
    transcript_path = os.path.join(args.output_path, audio_name + '.json')
    html_path = os.path.join(args.output_path, audio_name + '.html')

    diarization = pipeline(dict(audio=audio_path))

    transcript = [
        dict(audio_path=audio_path,
             begin=turn.start,
             end=turn.end,
             speaker_name=speaker)
        for turn, _, speaker in diarization.itertracks(yield_label=True)
    ]
    speaker_names = [None] + list(set(t['speaker_name'] for t in transcript))
    for t in transcript:
        t['speaker'] = speaker_names.index(t['speaker_name'])

    json.dump(transcript, open(transcript_path, 'w'), indent=2, sort_keys=True)
    vis.transcript(html_path,
                   sample_rate=args.sample_rate,
                   mono=True,
                   transcript=transcript_path)