paraformer-speaker inference pipeline

shixian.shi

2023-10-10 8a0930d682fe3206e0b41c694fc03d7d10c7eed2

 funasr/bin/asr_inference_launch.py

@@ -5,6 +5,7 @@

import argparse
import logging
from optparse import Option
import os
import sys
import time
@@ -45,6 +46,15 @@
from funasr.utils.types import str2triple_str
from funasr.utils.types import str_or_none
from funasr.utils.vad_utils import slice_padding_fbank
from funasr.utils.speaker_utils import (check_audio_list, 
                                        sv_preprocess, 
                                        sv_chunk, 
                                        CAMPPlus, 
                                        extract_feature, 
                                        postprocess,
                                        distribute_spk)
from funasr.build_utils.build_model_from_file import build_model_from_file
from funasr.utils.cluster_backend import ClusterBackend
from tqdm import tqdm

def inference_asr(
@@ -739,6 +749,342 @@

            logging.info("decoding, utt: {}, predictions: {}".format(key, text_postprocessed_punc))
        torch.cuda.empty_cache()
        return asr_result_list

    return _forward


def inference_paraformer_vad_speaker(
        maxlenratio: float,
        minlenratio: float,
        batch_size: int,
        beam_size: int,
        ngpu: int,
        ctc_weight: float,
        lm_weight: float,
        penalty: float,
        log_level: Union[int, str],
        # data_path_and_name_and_type,
        asr_train_config: Optional[str],
        asr_model_file: Optional[str],
        cmvn_file: Optional[str] = None,
        lm_train_config: Optional[str] = None,
        lm_file: Optional[str] = None,
        token_type: Optional[str] = None,
        key_file: Optional[str] = None,
        word_lm_train_config: Optional[str] = None,
        bpemodel: Optional[str] = None,
        allow_variable_data_keys: bool = False,
        output_dir: Optional[str] = None,
        dtype: str = "float32",
        seed: int = 0,
        ngram_weight: float = 0.9,
        nbest: int = 1,
        num_workers: int = 1,
        vad_infer_config: Optional[str] = None,
        vad_model_file: Optional[str] = None,
        vad_cmvn_file: Optional[str] = None,
        time_stamp_writer: bool = True,
        punc_infer_config: Optional[str] = None,
        punc_model_file: Optional[str] = None,
        sv_model_file: Optional[str] = None,
        streaming: bool = False,
        embedding_node: str = "resnet1_dense",
        sv_threshold: float = 0.9465,
        outputs_dict: Optional[bool] = True,
        param_dict: dict = None,

        **kwargs,
):
    ncpu = kwargs.get("ncpu", 1)
    torch.set_num_threads(ncpu)

    if word_lm_train_config is not None:
        raise NotImplementedError("Word LM is not implemented")
    if ngpu > 1:
        raise NotImplementedError("only single GPU decoding is supported")

    logging.basicConfig(
        level=log_level,
        format="%(asctime)s (%(module)s:%(lineno)d) %(levelname)s: %(message)s",
    )

    if param_dict is not None:
        hotword_list_or_file = param_dict.get('hotword')
    else:
        hotword_list_or_file = None

    if ngpu >= 1 and torch.cuda.is_available():
        device = "cuda"
    else:
        device = "cpu"

    # 1. Set random-seed
    set_all_random_seed(seed)

    # 2. Build speech2vadsegment
    speech2vadsegment_kwargs = dict(
        vad_infer_config=vad_infer_config,
        vad_model_file=vad_model_file,
        vad_cmvn_file=vad_cmvn_file,
        device=device,
        dtype=dtype,
    )
    # logging.info("speech2vadsegment_kwargs: {}".format(speech2vadsegment_kwargs))
    speech2vadsegment = Speech2VadSegment(**speech2vadsegment_kwargs)

    # 3. Build speech2text
    speech2text_kwargs = dict(
        asr_train_config=asr_train_config,
        asr_model_file=asr_model_file,
        cmvn_file=cmvn_file,
        lm_train_config=lm_train_config,
        lm_file=lm_file,
        token_type=token_type,
        bpemodel=bpemodel,
        device=device,
        maxlenratio=maxlenratio,
        minlenratio=minlenratio,
        dtype=dtype,
        beam_size=beam_size,
        ctc_weight=ctc_weight,
        lm_weight=lm_weight,
        ngram_weight=ngram_weight,
        penalty=penalty,
        nbest=nbest,
        hotword_list_or_file=hotword_list_or_file,
    )
    speech2text = Speech2TextParaformer(**speech2text_kwargs)
    text2punc = None
    if punc_model_file is not None:
        text2punc = Text2Punc(punc_infer_config, punc_model_file, device=device, dtype=dtype)

    if output_dir is not None:
        writer = DatadirWriter(output_dir)
        ibest_writer = writer[f"1best_recog"]
        ibest_writer["token_list"][""] = " ".join(speech2text.asr_train_args.token_list)

    def _forward(data_path_and_name_and_type,
                 raw_inputs: Union[np.ndarray, torch.Tensor] = None,
                 output_dir_v2: Optional[str] = None,
                 fs: dict = None,
                 param_dict: dict = None,
                 **kwargs,
                 ):

        hotword_list_or_file = None
        if param_dict is not None:
            hotword_list_or_file = param_dict.get('hotword')

        if 'hotword' in kwargs:
            hotword_list_or_file = kwargs['hotword']

        speech2vadsegment.vad_model.vad_opts.max_single_segment_time = kwargs.get("max_single_segment_time", 60000)
        batch_size_token_threshold_s = kwargs.get("batch_size_token_threshold_s", int(speech2vadsegment.vad_model.vad_opts.max_single_segment_time*0.67/1000)) * 1000
        batch_size_token = kwargs.get("batch_size_token", 6000)
        print("batch_size_token: ", batch_size_token)

        if speech2text.hotword_list is None:
            speech2text.hotword_list = speech2text.generate_hotwords_list(hotword_list_or_file)

        # 3. Build data-iterator
        if data_path_and_name_and_type is None and raw_inputs is not None:
            if isinstance(raw_inputs, torch.Tensor):
                raw_inputs = raw_inputs.numpy()
            data_path_and_name_and_type = [raw_inputs, "speech", "waveform"]
        loader = build_streaming_iterator(
            task_name="asr",
            preprocess_args=None,
            data_path_and_name_and_type=data_path_and_name_and_type,
            dtype=dtype,
            fs=fs,
            batch_size=1,
            key_file=key_file,
            num_workers=num_workers,
        )

        if param_dict is not None:
            use_timestamp = param_dict.get('use_timestamp', True)
        else:
            use_timestamp = True

        finish_count = 0
        file_count = 1
        lfr_factor = 6
        # 7 .Start for-loop
        asr_result_list = []
        output_path = output_dir_v2 if output_dir_v2 is not None else output_dir
        writer = None
        if output_path is not None:
            writer = DatadirWriter(output_path)
            ibest_writer = writer[f"1best_recog"]

        for keys, batch in loader:
            assert isinstance(batch, dict), type(batch)
            assert all(isinstance(s, str) for s in keys), keys
            _bs = len(next(iter(batch.values())))
            assert len(keys) == _bs, f"{len(keys)} != {_bs}"
            beg_vad = time.time()
            vad_results = speech2vadsegment(**batch)
            end_vad = time.time()
            print("time cost vad: ", end_vad - beg_vad)
            _, vadsegments = vad_results[0], vad_results[1][0]
            ##################################
            #####  speaker_verification  #####
            ##################################
            # load sv model
            sv_model_dict = torch.load(sv_model_file, map_location=torch.device('cpu'))
            sv_model = CAMPPlus()
            sv_model.load_state_dict(sv_model_dict)
            sv_model.eval()
            cb_model = ClusterBackend()
            vad_segments = []
            audio = batch['speech'].numpy().reshape(-1)
            for vadsegment in vadsegments:
                st = int(vadsegment[0]) / 1000
                ed = int(vadsegment[1]) / 1000
                vad_segments.append(
                    [st, ed, audio[int(st * 16000):int(ed * 16000)]])
            check_audio_list(vad_segments)
            # sv pipeline
            segments = sv_chunk(vad_segments)
            embeddings = []
            for s in segments:
                #_, embs = self.sv_pipeline([s[2]], output_emb=True)
                # embeddings.append(embs)
                wavs = sv_preprocess([s[2]])
                # embs = self.forward(wavs)
                embs = []
                for x in wavs:
                    x = extract_feature([x])
                    embs.append(sv_model(x))
                embs = torch.cat(embs)
                embeddings.append(embs.detach().numpy())
            embeddings = np.concatenate(embeddings)
            labels = cb_model(embeddings)
            sv_output = postprocess(segments, vad_segments, labels, embeddings)

            speech, speech_lengths = batch["speech"], batch["speech_lengths"]

            n = len(vadsegments)
            data_with_index = [(vadsegments[i], i) for i in range(n)]
            sorted_data = sorted(data_with_index, key=lambda x: x[0][1] - x[0][0])
            results_sorted = []
            
            if not len(sorted_data):
                key = keys[0]
                # no active segments after VAD
                if writer is not None:
                    # Write empty results
                    ibest_writer["token"][key] = ""
                    ibest_writer["token_int"][key] = ""
                    ibest_writer["vad"][key] = ""
                    ibest_writer["text"][key] = ""
                    ibest_writer["text_with_punc"][key] = ""
                    if use_timestamp:
                        ibest_writer["time_stamp"][key] = ""

                logging.info("decoding, utt: {}, empty speech".format(key))
                continue

            batch_size_token_ms = batch_size_token*60
            if speech2text.device == "cpu":
                batch_size_token_ms = 0
            if len(sorted_data) > 0 and len(sorted_data[0]) > 0:
                batch_size_token_ms = max(batch_size_token_ms, sorted_data[0][0][1] - sorted_data[0][0][0])
            
            batch_size_token_ms_cum = 0
            beg_idx = 0
            beg_asr_total = time.time()
            for j, _ in enumerate(tqdm(range(0, n))):
                batch_size_token_ms_cum += (sorted_data[j][0][1] - sorted_data[j][0][0])
                if j < n - 1 and (batch_size_token_ms_cum + sorted_data[j + 1][0][1] - sorted_data[j + 1][0][0]) < batch_size_token_ms and (sorted_data[j + 1][0][1] - sorted_data[j + 1][0][0]) < batch_size_token_threshold_s:
                    continue
                batch_size_token_ms_cum = 0
                end_idx = j + 1
                speech_j, speech_lengths_j = slice_padding_fbank(speech, speech_lengths, sorted_data[beg_idx:end_idx])
                beg_idx = end_idx
                batch = {"speech": speech_j, "speech_lengths": speech_lengths_j}
                batch = to_device(batch, device=device)
                # print("batch: ", speech_j.shape[0])
                beg_asr = time.time()
                results = speech2text(**batch)
                end_asr = time.time()
                # print("time cost asr: ", end_asr - beg_asr)

                if len(results) < 1:
                    results = [["", [], [], [], [], [], []]]
                results_sorted.extend(results)
            end_asr_total = time.time()
            print("total time cost asr: ", end_asr_total-beg_asr_total)
            restored_data = [0] * n
            for j in range(n):
                index = sorted_data[j][1]
                restored_data[index] = results_sorted[j]
            result = ["", [], [], [], [], [], []]
            for j in range(n):
                result[0] += restored_data[j][0]
                result[1] += restored_data[j][1]
                result[2] += restored_data[j][2]
                if len(restored_data[j][4]) > 0:
                    for t in restored_data[j][4]:
                        t[0] += vadsegments[j][0]
                        t[1] += vadsegments[j][0]
                    result[4] += restored_data[j][4]
                # result = [result[k]+restored_data[j][k] for k in range(len(result[:-2]))]

            key = keys[0]
            # result = result_segments[0]
            text, token, token_int = result[0], result[1], result[2]
            time_stamp = result[4] if len(result[4]) > 0 else None

            if use_timestamp and time_stamp is not None and len(time_stamp):
                postprocessed_result = postprocess_utils.sentence_postprocess(token, time_stamp)
            else:
                postprocessed_result = postprocess_utils.sentence_postprocess(token)
            text_postprocessed = ""
            time_stamp_postprocessed = ""
            text_postprocessed_punc = postprocessed_result
            if len(postprocessed_result) == 3:
                text_postprocessed, time_stamp_postprocessed, word_lists = postprocessed_result[0], \
                                                                           postprocessed_result[1], \
                                                                           postprocessed_result[2]
            else:
                text_postprocessed, word_lists = postprocessed_result[0], postprocessed_result[1]

            text_postprocessed_punc = text_postprocessed
            punc_id_list = []
            if len(word_lists) > 0 and text2punc is not None:
                beg_punc = time.time()
                text_postprocessed_punc, punc_id_list = text2punc(word_lists, 20)
                end_punc = time.time()
                print("time cost punc: ", end_punc - beg_punc)

            item = {'key': key, 'value': text_postprocessed_punc}
            if text_postprocessed != "":
                item['text_postprocessed'] = text_postprocessed
            if time_stamp_postprocessed != "":
                item['time_stamp'] = time_stamp_postprocessed

            item['sentences'] = time_stamp_sentence(punc_id_list, time_stamp_postprocessed, text_postprocessed)

            asr_result_list.append(item)
            finish_count += 1
            # asr_utils.print_progress(finish_count / file_count)
            if writer is not None:
                # Write the result to each file
                ibest_writer["token"][key] = " ".join(token)
                ibest_writer["token_int"][key] = " ".join(map(str, token_int))
                ibest_writer["vad"][key] = "{}".format(vadsegments)
                ibest_writer["text"][key] = " ".join(word_lists)
                ibest_writer["text_with_punc"][key] = text_postprocessed_punc
                if time_stamp_postprocessed is not None:
                    ibest_writer["time_stamp"][key] = "{}".format(time_stamp_postprocessed)

            logging.info("decoding, utt: {}, predictions: {}".format(key, text_postprocessed_punc))
        torch.cuda.empty_cache()
        distribute_spk(asr_result_list[0]['sentences'], sv_output)
        import pdb; pdb.set_trace()
        return asr_result_list

    return _forward
@@ -1684,6 +2030,8 @@
        return inference_paraformer(**kwargs)
    elif mode == "paraformer_streaming":
        return inference_paraformer_online(**kwargs)
    elif mode == "paraformer_vad_speaker":
        return inference_paraformer_vad_speaker(**kwargs)
    elif mode.startswith("paraformer_vad"):
        return inference_paraformer_vad_punc(**kwargs)
    elif mode == "mfcca":
@@ -1782,6 +2130,16 @@
        help="VAD model parameter file",
    )
    group.add_argument(
        "--punc_infer_config",
        type=str,
        help="PUNC infer configuration",
    )
    group.add_argument(
        "--punc_model_file",
        type=str,
        help="PUNC model parameter file",
    )
    group.add_argument(
        "--cmvn_file",
        type=str,
        help="Global CMVN file",
@@ -1797,6 +2155,11 @@
        help="ASR model parameter file",
    )
    group.add_argument(
        "--sv_model_file",
        type=str,
        help="SV model parameter file",
    )
    group.add_argument(
        "--lm_train_config",
        type=str,
        help="LM training configuration",