diff --git a/dataset/rir_noise/rir_noise.py b/dataset/rir_noise/rir_noise.py
index e7b12289043..009175e5bcc 100644
--- a/dataset/rir_noise/rir_noise.py
+++ b/dataset/rir_noise/rir_noise.py
@@ -34,14 +34,14 @@
 
 DATA_HOME = os.path.expanduser('~/.cache/paddle/dataset/speech')
 
-URL_ROOT = 'http://www.openslr.org/resources/28'
+URL_ROOT = '--no-check-certificate http://www.openslr.org/resources/28'
 DATA_URL = URL_ROOT + '/rirs_noises.zip'
 MD5_DATA = 'e6f48e257286e05de56413b4779d8ffb'
 
 parser = argparse.ArgumentParser(description=__doc__)
 parser.add_argument(
     "--target_dir",
-    default=DATA_HOME + "/Aishell",
+    default=DATA_HOME + "/rirs_noise",
     type=str,
     help="Directory to save the dataset. (default: %(default)s)")
 parser.add_argument(
@@ -81,6 +81,10 @@ def create_manifest(data_dir, manifest_path_prefix):
                         },
                         ensure_ascii=False))
         manifest_path = manifest_path_prefix + '.' + dtype
+
+        if not os.path.exists(os.path.dirname(manifest_path)):
+            os.makedirs(os.path.dirname(manifest_path))
+
         with codecs.open(manifest_path, 'w', 'utf-8') as fout:
             for line in json_lines:
                 fout.write(line + '\n')
diff --git a/dataset/voxceleb/voxceleb1.py b/dataset/voxceleb/voxceleb1.py
index 9058620083a..95827f708c0 100644
--- a/dataset/voxceleb/voxceleb1.py
+++ b/dataset/voxceleb/voxceleb1.py
@@ -149,7 +149,7 @@ def prepare_dataset(base_url, data_list, target_dir, manifest_path,
     # we will download the voxceleb1 data to ${target_dir}/vox1/dev/ or ${target_dir}/vox1/test directory 
     if not os.path.exists(os.path.join(target_dir, "wav")):
         # download all dataset part
-        print("start to download the vox1 dev zip package")
+        print(f"start to download the vox1 zip package to {target_dir}")
         for zip_part in data_list.keys():
             download_url = " --no-check-certificate " + base_url + "/" + zip_part
             download(
diff --git a/dataset/voxceleb/voxceleb2.py b/dataset/voxceleb/voxceleb2.py
index 22a2e2ffe21..fe9e8b9c834 100644
--- a/dataset/voxceleb/voxceleb2.py
+++ b/dataset/voxceleb/voxceleb2.py
@@ -22,10 +22,12 @@
 import glob
 import json
 import os
+import subprocess
 from pathlib import Path
 
 import soundfile
 
+from utils.utility import check_md5sum
 from utils.utility import download
 from utils.utility import unzip
 
@@ -35,12 +37,22 @@
 BASE_URL = "--no-check-certificate https://www.robots.ox.ac.uk/~vgg/data/voxceleb/data/"
 
 # dev data
-DEV_DATA_URL = BASE_URL + '/vox2_aac.zip'
-DEV_MD5SUM = "bbc063c46078a602ca71605645c2a402"
+DEV_LIST = {
+    "vox2_dev_aac_partaa": "da070494c573e5c0564b1d11c3b20577",
+    "vox2_dev_aac_partab": "17fe6dab2b32b48abaf1676429cdd06f",
+    "vox2_dev_aac_partac": "1de58e086c5edf63625af1cb6d831528",
+    "vox2_dev_aac_partad": "5a043eb03e15c5a918ee6a52aad477f9",
+    "vox2_dev_aac_partae": "cea401b624983e2d0b2a87fb5d59aa60",
+    "vox2_dev_aac_partaf": "fc886d9ba90ab88e7880ee98effd6ae9",
+    "vox2_dev_aac_partag": "d160ecc3f6ee3eed54d55349531cb42e",
+    "vox2_dev_aac_partah": "6b84a81b9af72a9d9eecbb3b1f602e65",
+}
+
+DEV_TARGET_DATA = "vox2_dev_aac_parta* vox2_dev_aac.zip bbc063c46078a602ca71605645c2a402"
 
 # test data
-TEST_DATA_URL = BASE_URL + '/vox2_test_aac.zip'
-TEST_MD5SUM = "0d2b3ea430a821c33263b5ea37ede312"
+TEST_LIST = {"vox2_test_aac.zip": "0d2b3ea430a821c33263b5ea37ede312"}
+TEST_TARGET_DATA = "vox2_test_aac.zip vox2_test_aac.zip 0d2b3ea430a821c33263b5ea37ede312"
 
 parser = argparse.ArgumentParser(description=__doc__)
 parser.add_argument(
@@ -68,6 +80,14 @@
 
 
 def create_manifest(data_dir, manifest_path_prefix):
+    """Generate the voxceleb2 dataset manifest file.
+    We will create the ${manifest_path_prefix}.vox2 as the final manifest file 
+    The dev and test wav info will be put in one manifest file.
+
+    Args:
+        data_dir (str): voxceleb2 wav directory, which include dev and test subdataset
+        manifest_path_prefix (str): manifest file prefix
+    """
     print("Creating manifest %s ..." % manifest_path_prefix)
     json_lines = []
     data_path = os.path.join(data_dir, "**", "*.wav")
@@ -119,7 +139,19 @@ def create_manifest(data_dir, manifest_path_prefix):
         print(f"{total_sec / total_num} sec/utt", file=f)
 
 
-def download_dataset(url, md5sum, target_dir, dataset):
+def download_dataset(base_url, data_list, target_data, target_dir, dataset):
+    """Download the voxceleb2 zip package
+
+    Args:
+        base_url (str): the voxceleb2 dataset download baseline url
+        data_list (dict): the dataset part zip package and the md5 value
+        target_data (str): the final dataset zip info
+        target_dir (str): the dataset stored directory
+        dataset (str): the dataset name, dev or test
+
+    Raises:
+        RuntimeError: the md5sum occurs error
+    """
     if not os.path.exists(target_dir):
         os.makedirs(target_dir)
 
@@ -129,9 +161,34 @@ def download_dataset(url, md5sum, target_dir, dataset):
     # but the test dataset will unzip to aac
     # so, wo create the ${target_dir}/test and unzip the m4a to test dir
     if not os.path.exists(os.path.join(target_dir, dataset)):
-        filepath = download(url, md5sum, target_dir)
+        print(f"start to download the vox2 zip package to {target_dir}")
+        for zip_part in data_list.keys():
+            download_url = " --no-check-certificate " + base_url + "/" + zip_part
+            download(
+                url=download_url,
+                md5sum=data_list[zip_part],
+                target_dir=target_dir)
+
+        # pack the all part to target zip file
+        all_target_part, target_name, target_md5sum = target_data.split()
+        target_name = os.path.join(target_dir, target_name)
+        if not os.path.exists(target_name):
+            pack_part_cmd = "cat {}/{} > {}".format(target_dir, all_target_part,
+                                                    target_name)
+            subprocess.call(pack_part_cmd, shell=True)
+
+        # check the target zip file md5sum
+        if not check_md5sum(target_name, target_md5sum):
+            raise RuntimeError("{} MD5 checkssum failed".format(target_name))
+        else:
+            print("Check {} md5sum successfully".format(target_name))
+
         if dataset == "test":
-            unzip(filepath, os.path.join(target_dir, "test"))
+            # we need make the test directory
+            unzip(target_name, os.path.join(target_dir, "test"))
+        else:
+            # upzip dev zip pacakge and will create the dev directory
+            unzip(target_name, target_dir)
 
 
 def main():
@@ -142,14 +199,16 @@ def main():
     print("download: {}".format(args.download))
     if args.download:
         download_dataset(
-            url=DEV_DATA_URL,
-            md5sum=DEV_MD5SUM,
+            base_url=BASE_URL,
+            data_list=DEV_LIST,
+            target_data=DEV_TARGET_DATA,
             target_dir=args.target_dir,
             dataset="dev")
 
         download_dataset(
-            url=TEST_DATA_URL,
-            md5sum=TEST_MD5SUM,
+            base_url=BASE_URL,
+            data_list=TEST_LIST,
+            target_data=TEST_TARGET_DATA,
             target_dir=args.target_dir,
             dataset="test")
 
diff --git a/examples/voxceleb/sv0/conf/ecapa_tdnn.yaml b/examples/voxceleb/sv0/conf/ecapa_tdnn.yaml
index e58dca82dcc..4715c5a3c0b 100644
--- a/examples/voxceleb/sv0/conf/ecapa_tdnn.yaml
+++ b/examples/voxceleb/sv0/conf/ecapa_tdnn.yaml
@@ -1,14 +1,16 @@
 ###########################################
 #                Data                 #
 ###########################################
-# we should explicitly specify the wav path of vox2 audio data converted from m4a
-vox2_base_path: 
 augment: True
-batch_size: 16
+batch_size: 32
 num_workers: 2
-num_speakers: 7205 # 1211 vox1, 5994 vox2, 7205 vox1+2, test speakers: 41
+num_speakers: 1211 # 1211 vox1, 5994 vox2, 7205 vox1+2, test speakers: 41
 shuffle: True
+skip_prep: False
+split_ratio: 0.9
+chunk_duration: 3.0 # seconds
 random_chunk: True
+verification_file: data/vox1/veri_test2.txt
 
 ###########################################################
 #                FEATURE EXTRACTION SETTING               #
@@ -26,7 +28,6 @@ hop_size: 160        #10ms, sample rate 16000, 10 * 16000 / 1000 = 160
 # if we want use another model, please choose another configuration yaml file
 model:
   input_size: 80
-  # "channels": [512, 512, 512, 512, 1536],
   channels: [1024, 1024, 1024, 1024, 3072]
   kernel_sizes: [5, 3, 3, 3, 1]
   dilations: [1, 2, 3, 4, 1]
@@ -38,8 +39,8 @@ model:
 ###########################################
 seed: 1986 # according from speechbrain configuration
 epochs: 10
-save_interval: 1
-log_interval: 1
+save_interval: 10
+log_interval: 10
 learning_rate: 1e-8
 
 
diff --git a/examples/voxceleb/sv0/conf/ecapa_tdnn_small.yaml b/examples/voxceleb/sv0/conf/ecapa_tdnn_small.yaml
new file mode 100644
index 00000000000..5ad5ea28536
--- /dev/null
+++ b/examples/voxceleb/sv0/conf/ecapa_tdnn_small.yaml
@@ -0,0 +1,53 @@
+###########################################
+#                Data                 #
+###########################################
+augment: True
+batch_size: 16
+num_workers: 2
+num_speakers: 1211 # 1211 vox1, 5994 vox2, 7205 vox1+2, test speakers: 41
+shuffle: True
+skip_prep: False
+split_ratio: 0.9
+chunk_duration: 3.0 # seconds
+random_chunk: True
+verification_file: data/vox1/veri_test2.txt
+
+###########################################################
+#                FEATURE EXTRACTION SETTING               #
+###########################################################
+# currently, we only support fbank
+sr: 16000           # sample rate
+n_mels: 80
+window_size: 400     #25ms, sample rate 16000, 25 * 16000 / 1000 = 400 
+hop_size: 160        #10ms, sample rate 16000, 10 * 16000 / 1000 = 160
+
+###########################################################
+#                       MODEL SETTING                     #
+###########################################################
+# currently, we only support ecapa-tdnn in the ecapa_tdnn.yaml
+# if we want use another model, please choose another configuration yaml file
+model:
+  input_size: 80
+  channels: [512, 512, 512, 512, 1536]
+  kernel_sizes: [5, 3, 3, 3, 1]
+  dilations: [1, 2, 3, 4, 1]
+  attention_channels: 128
+  lin_neurons: 192
+
+###########################################
+#                Training                 #
+###########################################
+seed: 1986 # according from speechbrain configuration
+epochs: 100
+save_interval: 10
+log_interval: 10
+learning_rate: 1e-8
+
+
+###########################################
+#                Testing                  #
+###########################################
+global_embedding_norm: True
+embedding_mean_norm: True
+embedding_std_norm: False
+
diff --git a/examples/voxceleb/sv0/local/data.sh b/examples/voxceleb/sv0/local/data.sh
index a3ff1c486bd..d6010ec6627 100755
--- a/examples/voxceleb/sv0/local/data.sh
+++ b/examples/voxceleb/sv0/local/data.sh
@@ -12,7 +12,7 @@
 # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 # See the License for the specific language governing permissions and
 # limitations under the License.
-stage=1
+stage=0
 stop_stage=100
 
 . ${MAIN_ROOT}/utils/parse_options.sh || exit -1;
@@ -30,29 +30,114 @@ dir=$1
 conf_path=$2
 mkdir -p ${dir}
 
-if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
-    # data prepare for vox1 and vox2, vox2 must be converted from m4a to wav
-    # we should use the local/convert.sh convert m4a to wav
-    python3 local/data_prepare.py \
-                        --data-dir ${dir} \
-                        --config ${conf_path}
-fi 
-
+# Generally the `MAIN_ROOT` refers to the root of PaddleSpeech,
+# which is defined in the path.sh
+# And we will download the voxceleb data and rirs noise to ${MAIN_ROOT}/dataset
 TARGET_DIR=${MAIN_ROOT}/dataset
 mkdir -p ${TARGET_DIR}
 
 if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
-    # download data, generate manifests
-    python3 ${TARGET_DIR}/voxceleb/voxceleb1.py \
-      --manifest_prefix="data/vox1/manifest" \
+   # download data, generate manifests
+   # we will generate the manifest.{dev,test} file from ${TARGET_DIR}/voxceleb/vox1/{dev,test} directory
+   # and generate the meta info and download the trial file
+   # manifest.dev: 148642
+   # manifest.test: 4847
+   echo "Start to download vox1 dataset and generate the manifest files "
+   python3 ${TARGET_DIR}/voxceleb/voxceleb1.py \
+      --manifest_prefix="${dir}/vox1/manifest" \
       --target_dir="${TARGET_DIR}/voxceleb/vox1/"
 
-    if [ $? -ne 0 ]; then
-        echo "Prepare voxceleb failed. Terminated."
-        exit 1
-    fi
+   if [ $? -ne 0 ]; then
+      echo "Prepare voxceleb1 failed. Terminated."
+      exit 1
+   fi
+
+fi
+
+if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
+   # download voxceleb2 data
+   # we will download the data and unzip the package
+   # and we will store the m4a file in ${TARGET_DIR}/voxceleb/vox2/{dev,test}
+   echo "start to download vox2 dataset"
+   python3 ${TARGET_DIR}/voxceleb/voxceleb2.py \
+      --download \
+      --target_dir="${TARGET_DIR}/voxceleb/vox2/"
+
+   if [ $? -ne 0 ]; then
+      echo "Download voxceleb2 dataset failed. Terminated."
+      exit 1
+   fi
+
+fi
+
+if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
+   # convert the m4a to wav
+   # and we will not delete the original m4a file
+   echo "start to convert the m4a to wav"
+   bash local/convert.sh ${TARGET_DIR}/voxceleb/vox2/test/ || exit 1;
+   
+   if [ $? -ne 0 ]; then
+      echo "Convert voxceleb2 dataset from m4a to wav failed. Terminated."
+      exit 1
+   fi
+   echo "m4a convert to wav operation finished"
+fi
+
+if [ ${stage} -le 4 ] && [ ${stop_stage} -ge 4 ]; then
+   # generate the vox2 manifest file from wav file
+   # we will generate the ${dir}/vox2/manifest.vox2
+   # because we use all the vox2 dataset to train, so collect all the vox2 data in one file
+   echo "start generate the vox2 manifest files"
+   python3 ${TARGET_DIR}/voxceleb/voxceleb2.py \
+      --generate \
+      --manifest_prefix="${dir}/vox2/manifest" \
+      --target_dir="${TARGET_DIR}/voxceleb/vox2/"
 
-   #  for dataset in train dev test; do
-   #      mv data/manifest.${dataset} data/manifest.${dataset}.raw
-   #  done
-fi
\ No newline at end of file
+   if [ $? -ne 0 ]; then
+      echo "Prepare voxceleb2 dataset failed. Terminated."
+      exit 1
+   fi
+fi
+
+if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
+   # generate the vox csv file
+   # Currently, our training system use csv file for dataset
+   echo "convert the json format to csv format to be compatible with training process"
+   python3 local/make_vox_csv_dataset_from_json.py\
+      --train "${dir}/vox1/manifest.dev" "${dir}/vox2/manifest.vox2"\
+      --test "${dir}/vox1/manifest.test" \
+      --target_dir "${dir}/vox/" \
+      --config ${conf_path}
+
+   if [ $? -ne 0 ]; then
+      echo "Prepare voxceleb failed. Terminated."
+      exit 1
+   fi
+fi
+
+if [ ${stage} -le 6 ] && [ ${stop_stage} -ge 6 ]; then
+   # generate the open rir noise manifest file
+   echo "generate the open rir noise manifest file"
+   python3 ${TARGET_DIR}/rir_noise/rir_noise.py\
+      --manifest_prefix="${dir}/rir_noise/manifest" \
+      --target_dir="${TARGET_DIR}/rir_noise/"
+
+   if [ $? -ne 0 ]; then
+      echo "Prepare rir_noise failed. Terminated."
+      exit 1
+   fi
+fi
+
+if [ ${stage} -le 7 ] && [ ${stop_stage} -ge 7 ]; then
+   # generate the open rir noise manifest file
+   echo "generate the open rir noise csv file"
+   python3 local/make_rirs_noise_csv_dataset_from_json.py \
+      --noise_dir="${TARGET_DIR}/rir_noise/" \
+      --data_dir="${dir}/rir_noise/" \
+      --config ${conf_path}
+
+   if [ $? -ne 0 ]; then
+      echo "Prepare rir_noise failed. Terminated."
+      exit 1
+   fi
+fi
diff --git a/examples/voxceleb/sv0/local/make_rirs_noise_csv_dataset_from_json.py b/examples/voxceleb/sv0/local/make_rirs_noise_csv_dataset_from_json.py
new file mode 100644
index 00000000000..b25a9d49a19
--- /dev/null
+++ b/examples/voxceleb/sv0/local/make_rirs_noise_csv_dataset_from_json.py
@@ -0,0 +1,167 @@
+# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+"""
+Convert the PaddleSpeech jsonline format data to csv format data in voxceleb experiment.
+Currently, Speaker Identificaton Training process use csv format.
+"""
+import argparse
+import csv
+import os
+from typing import List
+
+import tqdm
+from yacs.config import CfgNode
+
+from paddleaudio import load as load_audio
+from paddlespeech.s2t.utils.log import Log
+from paddlespeech.vector.utils.vector_utils import get_chunks
+
+logger = Log(__name__).getlog()
+
+
+def get_chunks_list(wav_file: str,
+                    split_chunks: bool,
+                    base_path: str,
+                    chunk_duration: float=3.0) -> List[List[str]]:
+    """Get the single audio file info 
+
+    Args:
+        wav_file (list): the wav audio file and get this audio segment info list
+        split_chunks (bool): audio split flag
+        base_path (str): the audio base path 
+        chunk_duration (float): the chunk duration. 
+                                if set the split_chunks, we split the audio into multi-chunks segment.
+    """
+    waveform, sr = load_audio(wav_file)
+    audio_id = wav_file.split("/rir_noise/")[-1].split(".")[0]
+    audio_duration = waveform.shape[0] / sr
+
+    ret = []
+    if split_chunks and audio_duration > chunk_duration:  # Split into pieces of self.chunk_duration seconds.
+        uniq_chunks_list = get_chunks(chunk_duration, audio_id, audio_duration)
+
+        for idx, chunk in enumerate(uniq_chunks_list):
+            s, e = chunk.split("_")[-2:]  # Timestamps of start and end
+            start_sample = int(float(s) * sr)
+            end_sample = int(float(e) * sr)
+
+            # currently, all vector csv data format use one representation
+            # id, duration, wav, start, stop, label
+            # in rirs noise, all the label name is 'noise'
+            # the label is string type and we will convert it to integer type in training
+            ret.append([
+                chunk, audio_duration, wav_file, start_sample, end_sample,
+                "noise"
+            ])
+    else:  # Keep whole audio.
+        ret.append(
+            [audio_id, audio_duration, wav_file, 0, waveform.shape[0], "noise"])
+    return ret
+
+
+def generate_csv(wav_files,
+                 output_file: str,
+                 base_path: str,
+                 split_chunks: bool=True):
+    """Prepare the csv file according the wav files
+
+    Args:
+        wav_files (list): all the audio list to prepare the csv file
+        output_file (str): the output csv file
+        config (CfgNode): yaml configuration content
+        split_chunks (bool): audio split flag
+    """
+    logger.info(f'Generating csv: {output_file}')
+    header = ["utt_id", "duration", "wav", "start", "stop", "label"]
+    csv_lines = []
+    for item in tqdm.tqdm(wav_files):
+        csv_lines.extend(
+            get_chunks_list(
+                item, base_path=base_path, split_chunks=split_chunks))
+
+    if not os.path.exists(os.path.dirname(output_file)):
+        os.makedirs(os.path.dirname(output_file))
+
+    with open(output_file, mode="w") as csv_f:
+        csv_writer = csv.writer(
+            csv_f, delimiter=",", quotechar='"', quoting=csv.QUOTE_MINIMAL)
+        csv_writer.writerow(header)
+        for line in csv_lines:
+            csv_writer.writerow(line)
+
+
+def prepare_data(args, config):
+    """Convert the jsonline format to csv format
+
+    Args:
+        args (argparse.Namespace): scripts args
+        config (CfgNode): yaml configuration content
+    """
+    # if external config set the skip_prep flat, we will do nothing
+    if config.skip_prep:
+        return
+
+    base_path = args.noise_dir
+    wav_path = os.path.join(base_path, "RIRS_NOISES")
+    logger.info(f"base path: {base_path}")
+    logger.info(f"wav path: {wav_path}")
+    rir_list = os.path.join(wav_path, "real_rirs_isotropic_noises", "rir_list")
+    rir_files = []
+    with open(rir_list, 'r') as f:
+        for line in f.readlines():
+            rir_file = line.strip().split(' ')[-1]
+            rir_files.append(os.path.join(base_path, rir_file))
+
+    noise_list = os.path.join(wav_path, "pointsource_noises", "noise_list")
+    noise_files = []
+    with open(noise_list, 'r') as f:
+        for line in f.readlines():
+            noise_file = line.strip().split(' ')[-1]
+            noise_files.append(os.path.join(base_path, noise_file))
+
+    csv_path = os.path.join(args.data_dir, 'csv')
+    logger.info(f"csv path: {csv_path}")
+    generate_csv(
+        rir_files, os.path.join(csv_path, 'rir.csv'), base_path=base_path)
+    generate_csv(
+        noise_files, os.path.join(csv_path, 'noise.csv'), base_path=base_path)
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser(description=__doc__)
+    parser.add_argument(
+        "--noise_dir",
+        default=None,
+        required=True,
+        help="The noise dataset dataset directory.")
+    parser.add_argument(
+        "--data_dir",
+        default=None,
+        required=True,
+        help="The target directory stores the csv files")
+    parser.add_argument(
+        "--config",
+        default=None,
+        required=True,
+        type=str,
+        help="configuration file")
+    args = parser.parse_args()
+
+    # parse the yaml config file
+    config = CfgNode(new_allowed=True)
+    if args.config:
+        config.merge_from_file(args.config)
+
+    # prepare the csv file from jsonlines files
+    prepare_data(args, config)
diff --git a/examples/voxceleb/sv0/local/make_vox_csv_dataset_from_json.py b/examples/voxceleb/sv0/local/make_vox_csv_dataset_from_json.py
new file mode 100644
index 00000000000..4e64c306712
--- /dev/null
+++ b/examples/voxceleb/sv0/local/make_vox_csv_dataset_from_json.py
@@ -0,0 +1,251 @@
+# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+"""
+Convert the PaddleSpeech jsonline format data to csv format data in voxceleb experiment.
+Currently, Speaker Identificaton Training process use csv format.
+"""
+import argparse
+import csv
+import json
+import os
+import random
+
+import tqdm
+from yacs.config import CfgNode
+
+from paddleaudio import load as load_audio
+from paddlespeech.s2t.utils.log import Log
+from paddlespeech.vector.utils.vector_utils import get_chunks
+
+logger = Log(__name__).getlog()
+
+
+def prepare_csv(wav_files, output_file, config, split_chunks=True):
+    """Prepare the csv file according the wav files
+
+    Args:
+        wav_files (list): all the audio list to prepare the csv file
+        output_file (str): the output csv file
+        config (CfgNode): yaml configuration content
+        split_chunks (bool, optional): audio split flag. Defaults to True.
+    """
+    if not os.path.exists(os.path.dirname(output_file)):
+        os.makedirs(os.path.dirname(output_file))
+    csv_lines = []
+    header = ["utt_id", "duration", "wav", "start", "stop", "label"]
+    # voxceleb meta info for each training utterance segment
+    # we extract a segment from a utterance to train 
+    # and the segment' period is between start and stop time point in the original wav file
+    # each field in the meta info means as follows:
+    # utt_id: the utterance segment name, which is uniq in training dataset
+    # duration: the total utterance time
+    # wav: utterance file path, which should be absoulute path
+    # start: start point in the original wav file sample point range
+    # stop: stop point in the original wav file sample point range
+    # label: the utterance segment's label name, 
+    #        which is speaker name in speaker verification domain
+    for item in tqdm.tqdm(wav_files, total=len(wav_files)):
+        item = json.loads(item.strip())
+        audio_id = item['utt'].replace(".wav",
+                                       "")  # we remove the wav suffix name
+        audio_duration = item['feat_shape'][0]
+        wav_file = item['feat']
+        label = audio_id.split('-')[
+            0]  # speaker name in speaker verification domain
+        waveform, sr = load_audio(wav_file)
+        if split_chunks:
+            uniq_chunks_list = get_chunks(config.chunk_duration, audio_id,
+                                          audio_duration)
+            for chunk in uniq_chunks_list:
+                s, e = chunk.split("_")[-2:]  # Timestamps of start and end
+                start_sample = int(float(s) * sr)
+                end_sample = int(float(e) * sr)
+                # id, duration, wav, start, stop, label
+                # in vector, the label in speaker id
+                csv_lines.append([
+                    chunk, audio_duration, wav_file, start_sample, end_sample,
+                    label
+                ])
+        else:
+            csv_lines.append([
+                audio_id, audio_duration, wav_file, 0, waveform.shape[0], label
+            ])
+
+    with open(output_file, mode="w") as csv_f:
+        csv_writer = csv.writer(
+            csv_f, delimiter=',', quotechar='"', quoting=csv.QUOTE_MINIMAL)
+        csv_writer.writerow(header)
+        for line in csv_lines:
+            csv_writer.writerow(line)
+
+
+def get_enroll_test_list(dataset_list, verification_file):
+    """Get the enroll and test utterance list from all the voxceleb1 test utterance dataset.
+       Generally, we get the enroll and test utterances from the verfification file.
+       The verification file format as follows:
+       target/nontarget enroll-utt test-utt,
+       we set 0 as nontarget and 1 as target, eg:
+       0 a.wav b.wav
+       1 a.wav a.wav
+
+    Args:
+        dataset_list (list): all the dataset to get the test utterances
+        verification_file (str): voxceleb1 trial file
+    """
+    logger.info(f"verification file: {verification_file}")
+    enroll_audios = set()
+    test_audios = set()
+    with open(verification_file, 'r') as f:
+        for line in f:
+            _, enroll_file, test_file = line.strip().split(' ')
+            enroll_audios.add('-'.join(enroll_file.split('/')))
+            test_audios.add('-'.join(test_file.split('/')))
+
+    enroll_files = []
+    test_files = []
+    for dataset in dataset_list:
+        with open(dataset, 'r') as f:
+            for line in f:
+                # audio_id may be in enroll and test at the same time
+                # eg: 1 a.wav a.wav
+                # the audio a.wav is enroll and test file at the same time
+                audio_id = json.loads(line.strip())['utt']
+                if audio_id in enroll_audios:
+                    enroll_files.append(line)
+                if audio_id in test_audios:
+                    test_files.append(line)
+
+    enroll_files = sorted(enroll_files)
+    test_files = sorted(test_files)
+
+    return enroll_files, test_files
+
+
+def get_train_dev_list(dataset_list, target_dir, split_ratio):
+    """Get the train and dev utterance list from all the training utterance dataset.
+       Generally, we use the split_ratio as the train dataset ratio,
+       and the remaining utterance (ratio is 1 - split_ratio) is the dev dataset
+
+    Args:
+        dataset_list (list): all the dataset to get the all utterances
+        target_dir (str): the target train and dev directory, 
+                          we will create the csv directory to store the {train,dev}.csv file
+        split_ratio (float): train dataset ratio in all utterance list
+    """
+    logger.info("start to get train and dev utt list")
+    if not os.path.exists(os.path.join(target_dir, "meta")):
+        os.makedirs(os.path.join(target_dir, "meta"))
+
+    audio_files = []
+    speakers = set()
+    for dataset in dataset_list:
+        with open(dataset, 'r') as f:
+            for line in f:
+                # the label is speaker name
+                label_name = json.loads(line.strip())['utt2spk']
+                speakers.add(label_name)
+                audio_files.append(line.strip())
+    speakers = sorted(speakers)
+    logger.info(f"we get {len(speakers)} speakers from all the train dataset")
+
+    with open(os.path.join(target_dir, "meta", "label2id.txt"), 'w') as f:
+        for label_id, label_name in enumerate(speakers):
+            f.write(f'{label_name} {label_id}\n')
+    logger.info(
+        f'we store the speakers to {os.path.join(target_dir, "meta", "label2id.txt")}'
+    )
+
+    # the split_ratio is for train dataset 
+    # the remaining is for dev dataset
+    split_idx = int(split_ratio * len(audio_files))
+    audio_files = sorted(audio_files)
+    random.shuffle(audio_files)
+    train_files, dev_files = audio_files[:split_idx], audio_files[split_idx:]
+    logger.info(
+        f"we get train utterances: {len(train_files)}, dev utterance: {len(dev_files)}"
+    )
+    return train_files, dev_files
+
+
+def prepare_data(args, config):
+    """Convert the jsonline format to csv format
+
+    Args:
+        args (argparse.Namespace): scripts args
+        config (CfgNode): yaml configuration content
+    """
+    # stage0: set the random seed
+    random.seed(config.seed)
+
+    # if external config set the skip_prep flat, we will do nothing
+    if config.skip_prep:
+        return
+
+    # stage 1: prepare the enroll and test csv file
+    #          And we generate the speaker to label file label2id.txt
+    logger.info("start to prepare the data csv file")
+    enroll_files, test_files = get_enroll_test_list(
+        [args.test], verification_file=config.verification_file)
+    prepare_csv(
+        enroll_files,
+        os.path.join(args.target_dir, "csv", "enroll.csv"),
+        config,
+        split_chunks=False)
+    prepare_csv(
+        test_files,
+        os.path.join(args.target_dir, "csv", "test.csv"),
+        config,
+        split_chunks=False)
+
+    # stage 2: prepare the train and dev csv file
+    #          we get the train dataset ratio as config.split_ratio
+    #          and the remaining is dev dataset
+    logger.info("start to prepare the data csv file")
+    train_files, dev_files = get_train_dev_list(
+        args.train, target_dir=args.target_dir, split_ratio=config.split_ratio)
+    prepare_csv(train_files,
+                os.path.join(args.target_dir, "csv", "train.csv"), config)
+    prepare_csv(dev_files,
+                os.path.join(args.target_dir, "csv", "dev.csv"), config)
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser(description=__doc__)
+    parser.add_argument(
+        "--train",
+        required=True,
+        nargs='+',
+        help="The jsonline files list for train.")
+    parser.add_argument(
+        "--test", required=True, help="The jsonline file for test")
+    parser.add_argument(
+        "--target_dir",
+        default=None,
+        required=True,
+        help="The target directory stores the csv files and meta file.")
+    parser.add_argument(
+        "--config",
+        default=None,
+        required=True,
+        type=str,
+        help="configuration file")
+    args = parser.parse_args()
+
+    # parse the yaml config file
+    config = CfgNode(new_allowed=True)
+    if args.config:
+        config.merge_from_file(args.config)
+
+    # prepare the csv file from jsonlines files
+    prepare_data(args, config)
diff --git a/examples/voxceleb/sv0/run.sh b/examples/voxceleb/sv0/run.sh
index bbc9e3dbb66..e1dccf2ae9e 100755
--- a/examples/voxceleb/sv0/run.sh
+++ b/examples/voxceleb/sv0/run.sh
@@ -18,24 +18,22 @@ set -e
 
 #######################################################################
 # stage 0: data prepare, including voxceleb1 download and generate {train,dev,enroll,test}.csv
-#          voxceleb2 data is m4a format, so we need user to convert the m4a to wav yourselves as described in Readme.md with the script local/convert.sh
+#          voxceleb2 data is m4a format, so we need convert the m4a to wav yourselves with the script local/convert.sh
 # stage 1: train the speaker identification model
 # stage 2: test speaker identification 
-# stage 3: extract the training embeding to train the LDA and PLDA
+# stage 3: (todo)extract the training embeding to train the LDA and PLDA
 ######################################################################
 
-# we can set the variable PPAUDIO_HOME to specifiy the root directory of the downloaded vox1 and vox2 dataset 
-# default the dataset will be stored in the ~/.paddleaudio/
 # the vox2 dataset is stored in m4a format, we need to convert the audio from m4a to wav yourself
-# and put all of them to ${PPAUDIO_HOME}/datasets/vox2
-# we will find the wav from ${PPAUDIO_HOME}/datasets/vox1/wav and ${PPAUDIO_HOME}/datasets/vox2/wav
-# export PPAUDIO_HOME=
+# and put all of them to ${MAIN_ROOT}/datasets/vox2
+# we will find the wav from ${MAIN_ROOT}/datasets/vox1/{dev,test}/wav and ${MAIN_ROOT}/datasets/vox2/wav
+
 stage=0
 stop_stage=50
 
 # data directory
 # if we set the variable ${dir}, we will store the wav info to this directory
-# otherwise, we will store the wav info to vox1 and vox2 directory respectively
+# otherwise, we will store the wav info to data/vox1 and data/vox2 directory respectively
 # vox2 wav path, we must convert the m4a format to wav format    
 dir=data/                                 # data info directory   
 
@@ -64,6 +62,6 @@ if [ $stage -le 2 ] && [ ${stop_stage} -ge 2 ]; then
 fi
 
 # if [ $stage -le 3 ]; then
-#      # stage 2: extract the training embeding to train the LDA and PLDA
+#      # stage 3: extract the training embeding to train the LDA and PLDA
 #      # todo: extract the training embedding
 # fi 
diff --git a/paddleaudio/paddleaudio/datasets/voxceleb.py b/paddleaudio/paddleaudio/datasets/voxceleb.py
index 3f72b5f2eb0..07f44e0c1fc 100644
--- a/paddleaudio/paddleaudio/datasets/voxceleb.py
+++ b/paddleaudio/paddleaudio/datasets/voxceleb.py
@@ -261,7 +261,7 @@ def generate_csv(self,
                      output_file: str,
                      split_chunks: bool=True):
         print(f'Generating csv: {output_file}')
-        header = ["ID", "duration", "wav", "start", "stop", "spk_id"]
+        header = ["id", "duration", "wav", "start", "stop", "spk_id"]
         # Note: this may occurs c++ execption, but the program will execute fine
         # so we can ignore the execption 
         with Pool(cpu_count()) as p:
diff --git a/paddlespeech/vector/exps/ecapa_tdnn/test.py b/paddlespeech/vector/exps/ecapa_tdnn/test.py
index d0de6dc51a8..70b1521ed67 100644
--- a/paddlespeech/vector/exps/ecapa_tdnn/test.py
+++ b/paddlespeech/vector/exps/ecapa_tdnn/test.py
@@ -21,10 +21,11 @@
 from tqdm import tqdm
 from yacs.config import CfgNode
 
-from paddleaudio.datasets import VoxCeleb
 from paddleaudio.metric import compute_eer
 from paddlespeech.s2t.utils.log import Log
 from paddlespeech.vector.io.batch import batch_feature_normalize
+from paddlespeech.vector.io.dataset import CSVDataset
+from paddlespeech.vector.io.embedding_norm import InputNormalization
 from paddlespeech.vector.models.ecapa_tdnn import EcapaTdnn
 from paddlespeech.vector.modules.sid_model import SpeakerIdetification
 from paddlespeech.vector.training.seeding import seed_everything
@@ -32,6 +33,91 @@
 logger = Log(__name__).getlog()
 
 
+def compute_dataset_embedding(data_loader, model, mean_var_norm_emb, config,
+                              id2embedding):
+    """compute the dataset embeddings
+
+    Args:
+        data_loader (_type_): _description_
+        model (_type_): _description_
+        mean_var_norm_emb (_type_): _description_
+        config (_type_): _description_
+    """
+    logger.info(
+        f'Computing embeddings on {data_loader.dataset.csv_path} dataset')
+    with paddle.no_grad():
+        for batch_idx, batch in enumerate(tqdm(data_loader)):
+
+            # stage 8-1: extrac the audio embedding
+            ids, feats, lengths = batch['ids'], batch['feats'], batch['lengths']
+            embeddings = model.backbone(feats, lengths).squeeze(
+                -1)  # (N, emb_size, 1) -> (N, emb_size)
+
+            # Global embedding normalization.
+            # if we use the global embedding norm
+            # eer can reduece about relative 10%
+            if config.global_embedding_norm and mean_var_norm_emb:
+                lengths = paddle.ones([embeddings.shape[0]])
+                embeddings = mean_var_norm_emb(embeddings, lengths)
+
+            # Update embedding dict.
+            id2embedding.update(dict(zip(ids, embeddings)))
+
+
+def compute_verification_scores(id2embedding, train_cohort, config):
+    labels = []
+    enroll_ids = []
+    test_ids = []
+    logger.info(f"read the trial from {config.verification_file}")
+    cos_sim_func = paddle.nn.CosineSimilarity(axis=-1)
+    scores = []
+    with open(config.verification_file, 'r') as f:
+        for line in f.readlines():
+            label, enroll_id, test_id = line.strip().split(' ')
+            enroll_id = enroll_id.split('.')[0].replace('/', '-')
+            test_id = test_id.split('.')[0].replace('/', '-')
+            labels.append(int(label))
+
+            enroll_emb = id2embedding[enroll_id]
+            test_emb = id2embedding[test_id]
+            score = cos_sim_func(enroll_emb, test_emb).item()
+
+            if "score_norm" in config:
+                # Getting norm stats for enroll impostors
+                enroll_rep = paddle.tile(
+                    enroll_emb, repeat_times=[train_cohort.shape[0], 1])
+                score_e_c = cos_sim_func(enroll_rep, train_cohort)
+                if "cohort_size" in config:
+                    score_e_c, _ = paddle.topk(
+                        score_e_c, k=config.cohort_size, axis=0)
+                mean_e_c = paddle.mean(score_e_c, axis=0)
+                std_e_c = paddle.std(score_e_c, axis=0)
+
+                # Getting norm stats for test impostors
+                test_rep = paddle.tile(
+                    test_emb, repeat_times=[train_cohort.shape[0], 1])
+                score_t_c = cos_sim_func(test_rep, train_cohort)
+                if "cohort_size" in config:
+                    score_t_c, _ = paddle.topk(
+                        score_t_c, k=config.cohort_size, axis=0)
+                mean_t_c = paddle.mean(score_t_c, axis=0)
+                std_t_c = paddle.std(score_t_c, axis=0)
+
+                if config.score_norm == "s-norm":
+                    score_e = (score - mean_e_c) / std_e_c
+                    score_t = (score - mean_t_c) / std_t_c
+
+                    score = 0.5 * (score_e + score_t)
+                elif config.score_norm == "z-norm":
+                    score = (score - mean_e_c) / std_e_c
+                elif config.score_norm == "t-norm":
+                    score = (score - mean_t_c) / std_t_c
+
+            scores.append(score)
+
+    return scores, labels
+
+
 def main(args, config):
     # stage0: set the training device, cpu or gpu
     paddle.set_device(args.device)
@@ -58,9 +144,8 @@ def main(args, config):
 
     # stage4: construct the enroll and test dataloader
 
-    enroll_dataset = VoxCeleb(
-        subset='enroll',
-        target_dir=args.data_dir,
+    enroll_dataset = CSVDataset(
+        os.path.join(args.data_dir, "vox/csv/enroll.csv"),
         feat_type='melspectrogram',
         random_chunk=False,
         n_mels=config.n_mels,
@@ -68,16 +153,15 @@ def main(args, config):
         hop_length=config.hop_size)
     enroll_sampler = BatchSampler(
         enroll_dataset, batch_size=config.batch_size,
-        shuffle=True)  # Shuffle to make embedding normalization more robust.
-    enrol_loader = DataLoader(enroll_dataset,
+        shuffle=False)  # Shuffle to make embedding normalization more robust.
+    enroll_loader = DataLoader(enroll_dataset,
                     batch_sampler=enroll_sampler,
                     collate_fn=lambda x: batch_feature_normalize(
-                            x, mean_norm=True, std_norm=False),
+                                x, mean_norm=True, std_norm=False),
                     num_workers=config.num_workers,
                     return_list=True,)
-    test_dataset = VoxCeleb(
-        subset='test',
-        target_dir=args.data_dir,
+    test_dataset = CSVDataset(
+        os.path.join(args.data_dir, "vox/csv/test.csv"),
         feat_type='melspectrogram',
         random_chunk=False,
         n_mels=config.n_mels,
@@ -85,7 +169,7 @@ def main(args, config):
         hop_length=config.hop_size)
 
     test_sampler = BatchSampler(
-        test_dataset, batch_size=config.batch_size, shuffle=True)
+        test_dataset, batch_size=config.batch_size, shuffle=False)
     test_loader = DataLoader(test_dataset,
                             batch_sampler=test_sampler,
                             collate_fn=lambda x: batch_feature_normalize(
@@ -97,75 +181,65 @@ def main(args, config):
 
     # stage6: global embedding norm to imporve the performance
     logger.info(f"global embedding norm: {config.global_embedding_norm}")
-    if config.global_embedding_norm:
-        global_embedding_mean = None
-        global_embedding_std = None
-        mean_norm_flag = config.embedding_mean_norm
-        std_norm_flag = config.embedding_std_norm
-        batch_count = 0
 
     # stage7: Compute embeddings of audios in enrol and test dataset from model.
+
+    if config.global_embedding_norm:
+        mean_var_norm_emb = InputNormalization(
+            norm_type="global",
+            mean_norm=config.embedding_mean_norm,
+            std_norm=config.embedding_std_norm)
+
+    if "score_norm" in config:
+        logger.info(f"we will do score norm: {config.score_norm}")
+        train_dataset = CSVDataset(
+            os.path.join(args.data_dir, "vox/csv/train.csv"),
+            feat_type='melspectrogram',
+            n_train_snts=config.n_train_snts,
+            random_chunk=False,
+            n_mels=config.n_mels,
+            window_size=config.window_size,
+            hop_length=config.hop_size)
+        train_sampler = BatchSampler(
+            train_dataset, batch_size=config.batch_size, shuffle=False)
+        train_loader = DataLoader(train_dataset,
+                            batch_sampler=train_sampler,
+                            collate_fn=lambda x: batch_feature_normalize(
+                                x, mean_norm=True, std_norm=False),
+                            num_workers=config.num_workers,
+                            return_list=True,)
+
     id2embedding = {}
     # Run multi times to make embedding normalization more stable.
-    for i in range(2):
-        for dl in [enrol_loader, test_loader]:
-            logger.info(
-                f'Loop {[i+1]}: Computing embeddings on {dl.dataset.subset} dataset'
-            )
-            with paddle.no_grad():
-                for batch_idx, batch in enumerate(tqdm(dl)):
-
-                    # stage 8-1: extrac the audio embedding
-                    ids, feats, lengths = batch['ids'], batch['feats'], batch[
-                        'lengths']
-                    embeddings = model.backbone(feats, lengths).squeeze(
-                        -1).numpy()  # (N, emb_size, 1) -> (N, emb_size)
-
-                    # Global embedding normalization.
-                    # if we use the global embedding norm
-                    # eer can reduece about relative 10%
-                    if config.global_embedding_norm:
-                        batch_count += 1
-                        current_mean = embeddings.mean(
-                            axis=0) if mean_norm_flag else 0
-                        current_std = embeddings.std(
-                            axis=0) if std_norm_flag else 1
-                        # Update global mean and std.
-                        if global_embedding_mean is None and global_embedding_std is None:
-                            global_embedding_mean, global_embedding_std = current_mean, current_std
-                        else:
-                            weight = 1 / batch_count  # Weight decay by batches.
-                            global_embedding_mean = (
-                                1 - weight
-                            ) * global_embedding_mean + weight * current_mean
-                            global_embedding_std = (
-                                1 - weight
-                            ) * global_embedding_std + weight * current_std
-                        # Apply global embedding normalization.
-                        embeddings = (embeddings - global_embedding_mean
-                                      ) / global_embedding_std
-
-                    # Update embedding dict.
-                    id2embedding.update(dict(zip(ids, embeddings)))
+    logger.info("First loop for enroll and test dataset")
+    compute_dataset_embedding(enroll_loader, model, mean_var_norm_emb, config,
+                              id2embedding)
+    compute_dataset_embedding(test_loader, model, mean_var_norm_emb, config,
+                              id2embedding)
+
+    logger.info("Second loop for enroll and test dataset")
+    compute_dataset_embedding(enroll_loader, model, mean_var_norm_emb, config,
+                              id2embedding)
+    compute_dataset_embedding(test_loader, model, mean_var_norm_emb, config,
+                              id2embedding)
+    mean_var_norm_emb.save(
+        os.path.join(args.load_checkpoint, "mean_var_norm_emb"))
 
     # stage 8: Compute cosine scores.
-    labels = []
-    enroll_ids = []
-    test_ids = []
-    logger.info(f"read the trial from {VoxCeleb.veri_test_file}")
-    with open(VoxCeleb.veri_test_file, 'r') as f:
-        for line in f.readlines():
-            label, enroll_id, test_id = line.strip().split(' ')
-            labels.append(int(label))
-            enroll_ids.append(enroll_id.split('.')[0].replace('/', '-'))
-            test_ids.append(test_id.split('.')[0].replace('/', '-'))
-
-    cos_sim_func = paddle.nn.CosineSimilarity(axis=1)
-    enrol_embeddings, test_embeddings = map(lambda ids: paddle.to_tensor(
-        np.asarray([id2embedding[uttid] for uttid in ids], dtype='float32')),
-                                            [enroll_ids, test_ids
-                                             ])  # (N, emb_size)
-    scores = cos_sim_func(enrol_embeddings, test_embeddings)
+    train_cohort = None
+    if "score_norm" in config:
+        train_embeddings = {}
+        # cohort embedding not do mean and std norm
+        compute_dataset_embedding(train_loader, model, None, config,
+                                  train_embeddings)
+        train_cohort = paddle.stack(list(train_embeddings.values()))
+
+    # compute the scores
+    scores, labels = compute_verification_scores(id2embedding, train_cohort,
+                                                 config)
+
+    # compute the EER and threshold
+    scores = paddle.to_tensor(scores)
     EER, threshold = compute_eer(np.asarray(labels), scores.numpy())
     logger.info(
         f'EER of verification test: {EER*100:.4f}%, score threshold: {threshold:.5f}'
diff --git a/paddlespeech/vector/exps/ecapa_tdnn/train.py b/paddlespeech/vector/exps/ecapa_tdnn/train.py
index 257b97abed7..b777dae8902 100644
--- a/paddlespeech/vector/exps/ecapa_tdnn/train.py
+++ b/paddlespeech/vector/exps/ecapa_tdnn/train.py
@@ -23,13 +23,13 @@
 from yacs.config import CfgNode
 
 from paddleaudio.compliance.librosa import melspectrogram
-from paddleaudio.datasets.voxceleb import VoxCeleb
 from paddlespeech.s2t.utils.log import Log
 from paddlespeech.vector.io.augment import build_augment_pipeline
 from paddlespeech.vector.io.augment import waveform_augment
 from paddlespeech.vector.io.batch import batch_pad_right
 from paddlespeech.vector.io.batch import feature_normalize
 from paddlespeech.vector.io.batch import waveform_collate_fn
+from paddlespeech.vector.io.dataset import CSVDataset
 from paddlespeech.vector.models.ecapa_tdnn import EcapaTdnn
 from paddlespeech.vector.modules.loss import AdditiveAngularMargin
 from paddlespeech.vector.modules.loss import LogSoftmaxWrapper
@@ -54,8 +54,12 @@ def main(args, config):
 
     # stage2: data prepare, such vox1 and vox2 data, and augment noise data and pipline
     # note: some cmd must do in rank==0, so wo will refactor the data prepare code
-    train_dataset = VoxCeleb('train', target_dir=args.data_dir)
-    dev_dataset = VoxCeleb('dev', target_dir=args.data_dir)
+    train_dataset = CSVDataset(
+        csv_path=os.path.join(args.data_dir, "vox/csv/train.csv"),
+        label2id_path=os.path.join(args.data_dir, "vox/meta/label2id.txt"))
+    dev_dataset = CSVDataset(
+        csv_path=os.path.join(args.data_dir, "vox/csv/dev.csv"),
+        label2id_path=os.path.join(args.data_dir, "vox/meta/label2id.txt"))
 
     if config.augment:
         augment_pipeline = build_augment_pipeline(target_dir=args.data_dir)
@@ -67,7 +71,7 @@ def main(args, config):
 
     # stage4: build the speaker verification train instance with backbone model
     model = SpeakerIdetification(
-        backbone=ecapa_tdnn, num_class=VoxCeleb.num_speakers)
+        backbone=ecapa_tdnn, num_class=config.num_speakers)
 
     # stage5: build the optimizer, we now only construct the AdamW optimizer
     #         140000 is single gpu steps
@@ -193,15 +197,15 @@ def main(args, config):
                           paddle.optimizer.lr.LRScheduler):
                 optimizer._learning_rate.step()
             optimizer.clear_grad()
-            train_run_cost += time.time() - train_start
 
             # stage 9-8: Calculate average loss per batch
-            avg_loss += loss.numpy()[0]
+            avg_loss = loss.item()
 
             # stage 9-9: Calculate metrics, which is one-best accuracy
             preds = paddle.argmax(logits, axis=1)
             num_corrects += (preds == labels).numpy().sum()
             num_samples += feats.shape[0]
+            train_run_cost += time.time() - train_start
             timer.count()  # step plus one in timer
 
             # stage 9-10: print the log information only on 0-rank per log-freq batchs
@@ -220,8 +224,9 @@ def main(args, config):
                     train_feat_cost / config.log_interval)
                 print_msg += ' avg_train_cost: {:.5f} sec,'.format(
                     train_run_cost / config.log_interval)
-                print_msg += ' lr={:.4E} step/sec={:.2f} | ETA {}'.format(
-                    lr, timer.timing, timer.eta)
+
+                print_msg += ' lr={:.4E} step/sec={:.2f} ips:{:.5f}| ETA {}'.format(
+                    lr, timer.timing, timer.ips, timer.eta)
                 logger.info(print_msg)
 
                 avg_loss = 0
diff --git a/paddlespeech/vector/io/augment.py b/paddlespeech/vector/io/augment.py
index 3baace13977..0aa89c6a344 100644
--- a/paddlespeech/vector/io/augment.py
+++ b/paddlespeech/vector/io/augment.py
@@ -14,6 +14,7 @@
 # this is modified from SpeechBrain
 # https://github.com/speechbrain/speechbrain/blob/085be635c07f16d42cd1295045bc46c407f1e15b/speechbrain/lobes/augment.py
 import math
+import os
 from typing import List
 
 import numpy as np
@@ -21,8 +22,8 @@
 import paddle.nn as nn
 import paddle.nn.functional as F
 
-from paddleaudio.datasets.rirs_noises import OpenRIRNoise
 from paddlespeech.s2t.utils.log import Log
+from paddlespeech.vector.io.dataset import CSVDataset
 from paddlespeech.vector.io.signal_processing import compute_amplitude
 from paddlespeech.vector.io.signal_processing import convolve1d
 from paddlespeech.vector.io.signal_processing import dB_to_amplitude
@@ -509,7 +510,7 @@ def pad(x, target_length, mode='constant', **kwargs):
                     assert w >= 0, f'Target length {target_length} is less than origin length {x.shape[0]}'
                     return np.pad(x, [0, w], mode=mode, **kwargs)
 
-                ids = [item['id'] for item in batch]
+                ids = [item['utt_id'] for item in batch]
                 lengths = np.asarray([item['feat'].shape[0] for item in batch])
                 waveforms = list(
                     map(lambda x: pad(x, max(max_length, lengths.max().item())),
@@ -589,7 +590,7 @@ def pad(x, target_length, mode='constant', **kwargs):
                 assert w >= 0, f'Target length {target_length} is less than origin length {x.shape[0]}'
                 return np.pad(x, [0, w], mode=mode, **kwargs)
 
-            ids = [item['id'] for item in batch]
+            ids = [item['utt_id'] for item in batch]
             lengths = np.asarray([item['feat'].shape[0] for item in batch])
             waveforms = list(
                 map(lambda x: pad(x, lengths.max().item()),
@@ -839,8 +840,10 @@ def build_augment_pipeline(target_dir=None) -> List[paddle.nn.Layer]:
         List[paddle.nn.Layer]: all augment process
     """
     logger.info("start to build the augment pipeline")
-    noise_dataset = OpenRIRNoise('noise', target_dir=target_dir)
-    rir_dataset = OpenRIRNoise('rir', target_dir=target_dir)
+    noise_dataset = CSVDataset(csv_path=os.path.join(target_dir,
+                                                     "rir_noise/csv/noise.csv"))
+    rir_dataset = CSVDataset(csv_path=os.path.join(target_dir,
+                                                   "rir_noise/csv/rir.csv"))
 
     wavedrop = TimeDomainSpecAugment(
         sample_rate=16000,
diff --git a/paddlespeech/vector/io/batch.py b/paddlespeech/vector/io/batch.py
index 92ca990cf2d..5049d1946ea 100644
--- a/paddlespeech/vector/io/batch.py
+++ b/paddlespeech/vector/io/batch.py
@@ -17,6 +17,17 @@
 
 
 def waveform_collate_fn(batch):
+    """Wrap the waveform into a batch form
+
+    Args:
+        batch (list): the waveform list from the dataloader
+                      the item of data include several field
+                      feat: the utterance waveform data
+                      label: the utterance label encoding data
+
+    Returns:
+        dict: the batch data to dataloader
+    """
     waveforms = np.stack([item['feat'] for item in batch])
     labels = np.stack([item['label'] for item in batch])
 
@@ -27,6 +38,18 @@ def feature_normalize(feats: paddle.Tensor,
                       mean_norm: bool=True,
                       std_norm: bool=True,
                       convert_to_numpy: bool=False):
+    """Do one utterance feature normalization
+
+    Args:
+        feats (paddle.Tensor): the original utterance feat, such as fbank, mfcc
+        mean_norm (bool, optional): mean norm flag. Defaults to True.
+        std_norm (bool, optional): std norm flag. Defaults to True.
+        convert_to_numpy (bool, optional): convert the paddle.tensor to numpy 
+                                           and do feature norm with numpy. Defaults to False.
+
+    Returns:
+        paddle.Tensor : the normalized feats
+    """
     # Features normalization if needed
     # numpy.mean is a little with paddle.mean about 1e-6
     if convert_to_numpy:
@@ -60,7 +83,17 @@ def pad_right_2d(x, target_length, axis=-1, mode='constant', **kwargs):
 
 
 def batch_feature_normalize(batch, mean_norm: bool=True, std_norm: bool=True):
-    ids = [item['id'] for item in batch]
+    """Do batch utterance features normalization
+
+    Args:
+        batch (list): the batch feature from dataloader
+        mean_norm (bool, optional): mean normalization flag. Defaults to True.
+        std_norm (bool, optional): std normalization flag. Defaults to True.
+
+    Returns:
+        dict: the normalized batch features
+    """
+    ids = [item['utt_id'] for item in batch]
     lengths = np.asarray([item['feat'].shape[1] for item in batch])
     feats = list(
         map(lambda x: pad_right_2d(x, lengths.max()),
diff --git a/paddlespeech/vector/io/dataset.py b/paddlespeech/vector/io/dataset.py
new file mode 100644
index 00000000000..316c8ac341b
--- /dev/null
+++ b/paddlespeech/vector/io/dataset.py
@@ -0,0 +1,192 @@
+# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+from dataclasses import dataclass
+from dataclasses import fields
+from paddle.io import Dataset
+
+from paddleaudio import load as load_audio
+from paddleaudio.compliance.librosa import melspectrogram
+from paddlespeech.s2t.utils.log import Log
+logger = Log(__name__).getlog()
+
+# the audio meta info in the vector CSVDataset
+# utt_id: the utterance segment name
+# duration: utterance segment time
+# wav: utterance file path
+# start: start point in the original wav file
+# stop: stop point in the original wav file
+# label: the utterance segment's label id
+
+
+@dataclass
+class meta_info:
+    """the audio meta info in the vector CSVDataset
+
+    Args:
+        utt_id (str): the utterance segment name
+        duration (float): utterance segment time
+        wav (str): utterance file path
+        start (int): start point in the original wav file
+        stop (int): stop point in the original wav file
+        lab_id (str): the utterance segment's label id
+    """
+    utt_id: str
+    duration: float
+    wav: str
+    start: int
+    stop: int
+    label: str
+
+
+# csv dataset support feature type
+# raw: return the pcm data sample point
+# melspectrogram: fbank feature
+feat_funcs = {
+    'raw': None,
+    'melspectrogram': melspectrogram,
+}
+
+
+class CSVDataset(Dataset):
+    def __init__(self,
+                 csv_path,
+                 label2id_path=None,
+                 config=None,
+                 random_chunk=True,
+                 feat_type: str="raw",
+                 n_train_snts: int=-1,
+                 **kwargs):
+        """Implement the CSV Dataset
+
+        Args:
+            csv_path (str): csv dataset file path
+            label2id_path (str): the utterance label to integer id map file path
+            config (CfgNode): yaml config
+            feat_type (str): dataset feature type. if it is raw, it return pcm data.
+            n_train_snts (int): select the n_train_snts sample from the dataset. 
+                                if n_train_snts = -1, dataset will load all the sample.
+                                Default value is -1.
+            kwargs : feature type args
+        """
+        super().__init__()
+        self.csv_path = csv_path
+        self.label2id_path = label2id_path
+        self.config = config
+        self.random_chunk = random_chunk
+        self.feat_type = feat_type
+        self.n_train_snts = n_train_snts
+        self.feat_config = kwargs
+        self.id2label = {}
+        self.label2id = {}
+        self.data = self.load_data_csv()
+        self.load_speaker_to_label()
+
+    def load_data_csv(self):
+        """Load the csv dataset content and store them in the data property
+        the csv dataset's format has six fields, 
+        that is audio_id or utt_id, audio duration, segment start point, segment stop point 
+        and utterance label.
+        Note in training period, the utterance label must has a map to integer id in label2id_path 
+
+        Returns:
+            list: the csv data with meta_info type
+        """
+        data = []
+
+        with open(self.csv_path, 'r') as rf:
+            for line in rf.readlines()[1:]:
+                audio_id, duration, wav, start, stop, spk_id = line.strip(
+                ).split(',')
+                data.append(
+                    meta_info(audio_id,
+                              float(duration), wav,
+                              int(start), int(stop), spk_id))
+        if self.n_train_snts > 0:
+            sample_num = min(self.n_train_snts, len(data))
+            data = data[0:sample_num]
+
+        return data
+
+    def load_speaker_to_label(self):
+        """Load the utterance label map content.
+        In vector domain, we call the utterance label as speaker label.
+        The speaker label is real speaker label in speaker verification domain,
+        and in language identification is language label.
+        """
+        if not self.label2id_path:
+            logger.warning("No speaker id to label file")
+            return
+
+        with open(self.label2id_path, 'r') as f:
+            for line in f.readlines():
+                label_name, label_id = line.strip().split(' ')
+                self.label2id[label_name] = int(label_id)
+                self.id2label[int(label_id)] = label_name
+
+    def convert_to_record(self, idx: int):
+        """convert the dataset sample to training record the CSV Dataset
+
+        Args:
+            idx (int) : the request index in all the dataset
+        """
+        sample = self.data[idx]
+
+        record = {}
+        # To show all fields in a namedtuple: `type(sample)._fields`
+        for field in fields(sample):
+            record[field.name] = getattr(sample, field.name)
+
+        waveform, sr = load_audio(record['wav'])
+
+        # random select a chunk audio samples from the audio
+        if self.config and self.config.random_chunk:
+            num_wav_samples = waveform.shape[0]
+            num_chunk_samples = int(self.config.chunk_duration * sr)
+            start = random.randint(0, num_wav_samples - num_chunk_samples - 1)
+            stop = start + num_chunk_samples
+        else:
+            start = record['start']
+            stop = record['stop']
+
+        # we only return the waveform as feat
+        waveform = waveform[start:stop]
+
+        # all availabel feature type is in feat_funcs
+        assert self.feat_type in feat_funcs.keys(), \
+            f"Unknown feat_type: {self.feat_type}, it must be one in {list(feat_funcs.keys())}"
+        feat_func = feat_funcs[self.feat_type]
+        feat = feat_func(
+            waveform, sr=sr, **self.feat_config) if feat_func else waveform
+
+        record.update({'feat': feat})
+        if self.label2id:
+            record.update({'label': self.label2id[record['label']]})
+
+        return record
+
+    def __getitem__(self, idx):
+        """Return the specific index sample
+
+        Args:
+            idx (int) : the request index in all the dataset
+        """
+        return self.convert_to_record(idx)
+
+    def __len__(self):
+        """Return the dataset length
+
+        Returns:
+            int: the length num of the dataset
+        """
+        return len(self.data)
diff --git a/paddlespeech/vector/io/embedding_norm.py b/paddlespeech/vector/io/embedding_norm.py
new file mode 100644
index 00000000000..619f3710105
--- /dev/null
+++ b/paddlespeech/vector/io/embedding_norm.py
@@ -0,0 +1,214 @@
+# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+from typing import Dict
+
+import paddle
+
+
+class InputNormalization:
+    spk_dict_mean: Dict[int, paddle.Tensor]
+    spk_dict_std: Dict[int, paddle.Tensor]
+    spk_dict_count: Dict[int, int]
+
+    def __init__(
+            self,
+            mean_norm=True,
+            std_norm=True,
+            norm_type="global", ):
+        """Do feature or embedding mean and std norm
+
+        Args:
+            mean_norm (bool, optional): mean norm flag. Defaults to True.
+            std_norm (bool, optional): std norm flag. Defaults to True.
+            norm_type (str, optional): norm type. Defaults to "global".
+        """
+        super().__init__()
+        self.training = True
+        self.mean_norm = mean_norm
+        self.std_norm = std_norm
+        self.norm_type = norm_type
+        self.glob_mean = paddle.to_tensor([0], dtype="float32")
+        self.glob_std = paddle.to_tensor([0], dtype="float32")
+        self.spk_dict_mean = {}
+        self.spk_dict_std = {}
+        self.spk_dict_count = {}
+        self.weight = 1.0
+        self.count = 0
+        self.eps = 1e-10
+
+    def __call__(self,
+                 x,
+                 lengths,
+                 spk_ids=paddle.to_tensor([], dtype="float32")):
+        """Returns the tensor with the surrounding context.
+        Args:
+            x (paddle.Tensor): A batch of tensors.
+            lengths (paddle.Tensor): A batch of tensors containing the relative length of each
+                                    sentence (e.g, [0.7, 0.9, 1.0]). It is used to avoid
+                                    computing stats on zero-padded steps.
+            spk_ids (_type_, optional): tensor containing the ids of each speaker (e.g, [0 10 6]).
+                                        It is used to perform per-speaker normalization when
+                                        norm_type='speaker'. Defaults to paddle.to_tensor([], dtype="float32").
+        Returns:
+            paddle.Tensor: The normalized feature or embedding
+        """
+        N_batches = x.shape[0]
+        # print(f"x shape: {x.shape[1]}")
+        current_means = []
+        current_stds = []
+
+        for snt_id in range(N_batches):
+
+            # Avoiding padded time steps
+            # actual size is the actual time data length
+            actual_size = paddle.round(lengths[snt_id] *
+                                       x.shape[1]).astype("int32")
+            # computing actual time data statistics
+            current_mean, current_std = self._compute_current_stats(
+                x[snt_id, 0:actual_size, ...].unsqueeze(0))
+            current_means.append(current_mean)
+            current_stds.append(current_std)
+
+        if self.norm_type == "global":
+            current_mean = paddle.mean(paddle.stack(current_means), axis=0)
+            current_std = paddle.mean(paddle.stack(current_stds), axis=0)
+
+            if self.norm_type == "global":
+
+                if self.training:
+                    if self.count == 0:
+                        self.glob_mean = current_mean
+                        self.glob_std = current_std
+
+                    else:
+                        self.weight = 1 / (self.count + 1)
+
+                        self.glob_mean = (
+                            1 - self.weight
+                        ) * self.glob_mean + self.weight * current_mean
+
+                        self.glob_std = (
+                            1 - self.weight
+                        ) * self.glob_std + self.weight * current_std
+
+                    self.glob_mean.detach()
+                    self.glob_std.detach()
+
+                    self.count = self.count + 1
+                x = (x - self.glob_mean) / (self.glob_std)
+        return x
+
+    def _compute_current_stats(self, x):
+        """Returns the tensor with the surrounding context.
+
+        Args:
+            x (paddle.Tensor): A batch of tensors.
+
+        Returns:
+             the statistics of the data
+        """
+        # Compute current mean
+        if self.mean_norm:
+            current_mean = paddle.mean(x, axis=0).detach()
+        else:
+            current_mean = paddle.to_tensor([0.0], dtype="float32")
+
+        # Compute current std
+        if self.std_norm:
+            current_std = paddle.std(x, axis=0).detach()
+        else:
+            current_std = paddle.to_tensor([1.0], dtype="float32")
+
+        # Improving numerical stability of std
+        current_std = paddle.maximum(current_std,
+                                     self.eps * paddle.ones_like(current_std))
+
+        return current_mean, current_std
+
+    def _statistics_dict(self):
+        """Fills the dictionary containing the normalization statistics.
+        """
+        state = {}
+        state["count"] = self.count
+        state["glob_mean"] = self.glob_mean
+        state["glob_std"] = self.glob_std
+        state["spk_dict_mean"] = self.spk_dict_mean
+        state["spk_dict_std"] = self.spk_dict_std
+        state["spk_dict_count"] = self.spk_dict_count
+
+        return state
+
+    def _load_statistics_dict(self, state):
+        """Loads the dictionary containing the statistics.
+
+        Arguments
+        ---------
+        state : dict
+            A dictionary containing the normalization statistics.
+        """
+        self.count = state["count"]
+        if isinstance(state["glob_mean"], int):
+            self.glob_mean = state["glob_mean"]
+            self.glob_std = state["glob_std"]
+        else:
+            self.glob_mean = state["glob_mean"]  # .to(self.device_inp)
+            self.glob_std = state["glob_std"]  # .to(self.device_inp)
+
+        # Loading the spk_dict_mean in the right device
+        self.spk_dict_mean = {}
+        for spk in state["spk_dict_mean"]:
+            self.spk_dict_mean[spk] = state["spk_dict_mean"][spk]
+
+        # Loading the spk_dict_std in the right device
+        self.spk_dict_std = {}
+        for spk in state["spk_dict_std"]:
+            self.spk_dict_std[spk] = state["spk_dict_std"][spk]
+
+        self.spk_dict_count = state["spk_dict_count"]
+
+        return state
+
+    def to(self, device):
+        """Puts the needed tensors in the right device.
+        """
+        self = super(InputNormalization, self).to(device)
+        self.glob_mean = self.glob_mean.to(device)
+        self.glob_std = self.glob_std.to(device)
+        for spk in self.spk_dict_mean:
+            self.spk_dict_mean[spk] = self.spk_dict_mean[spk].to(device)
+            self.spk_dict_std[spk] = self.spk_dict_std[spk].to(device)
+        return self
+
+    def save(self, path):
+        """Save statistic dictionary.
+    
+        Args:
+            path (str): A path where to save the dictionary.
+        """
+        stats = self._statistics_dict()
+        paddle.save(stats, path)
+
+    def _load(self, path, end_of_epoch=False, device=None):
+        """Load statistic dictionary.
+
+        Arguments
+        ---------
+        path : str
+            The path of the statistic dictionary
+        device : str, None
+            Passed to paddle.load(..., map_location=device)
+        """
+        del end_of_epoch  # Unused here.
+        stats = paddle.load(path, map_location=device)
+        self._load_statistics_dict(stats)
diff --git a/paddlespeech/vector/utils/time.py b/paddlespeech/vector/utils/time.py
index 8e85b0e120a..9dfbbe1f7d5 100644
--- a/paddlespeech/vector/utils/time.py
+++ b/paddlespeech/vector/utils/time.py
@@ -23,6 +23,7 @@ def __init__(self, total_step: int):
         self.last_start_step = 0
         self.current_step = 0
         self._is_running = True
+        self.cur_ips = 0
 
     def start(self):
         self.last_time = time.time()
@@ -43,12 +44,17 @@ def timing(self) -> float:
         self.last_start_step = self.current_step
         time_used = time.time() - self.last_time
         self.last_time = time.time()
+        self.cur_ips = run_steps / time_used
         return time_used / run_steps
 
     @property
     def is_running(self) -> bool:
         return self._is_running
 
+    @property
+    def ips(self) -> float:
+        return self.cur_ips
+
     @property
     def eta(self) -> str:
         if not self.is_running:
diff --git a/paddlespeech/vector/utils/vector_utils.py b/paddlespeech/vector/utils/vector_utils.py
new file mode 100644
index 00000000000..46de7ffaaa9
--- /dev/null
+++ b/paddlespeech/vector/utils/vector_utils.py
@@ -0,0 +1,32 @@
+# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+
+def get_chunks(seg_dur, audio_id, audio_duration):
+    """Get all chunk segments from a utterance
+
+    Args:
+        seg_dur (float): segment chunk duration, seconds
+        audio_id (str): utterance name, 
+        audio_duration (float): utterance duration, seconds
+
+    Returns:
+        List: all the chunk segments 
+    """
+    num_chunks = int(audio_duration / seg_dur)  # all in seconds
+    chunk_lst = [
+        audio_id + "_" + str(i * seg_dur) + "_" + str(i * seg_dur + seg_dur)
+        for i in range(num_chunks)
+    ]
+    return chunk_lst