Merge branch 'main' into inference_streaming

This commit is contained in:
Xiang Lyu
2024-08-29 23:48:02 +08:00
committed by GitHub
13 changed files with 750 additions and 1 deletions

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@@ -4,6 +4,36 @@
For `SenseVoice`, visit [SenseVoice repo](https://github.com/FunAudioLLM/SenseVoice) and [SenseVoice space](https://www.modelscope.cn/studios/iic/SenseVoice).
## Roadmap
- [x] 2024/07
- [x] Flow matching training support
- [x] WeTextProcessing support when ttsfrd is not avaliable
- [x] Fastapi server and client
- [ ] 2024/08
- [ ] Repetition Aware Sampling(RAS) inference for llm stability
- [ ] Streaming inference mode support, including kv cache and sdpa for rtf optimization
- [ ] 2024/09
- [ ] 50hz llm model which supports 10 language
- [ ] 2024/10
- [ ] 50hz llama based llm model which supports lora finetune
- [ ] TBD
- [ ] Support more instruction mode
- [ ] Voice conversion
- [ ] Music generation
- [ ] Training script sample based on Mandarin
- [ ] CosyVoice-500M trained with more multi-lingual data
- [ ] More...
## Install
**Clone and install**

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@@ -12,6 +12,7 @@
# See the License for the specific language governing permissions and
# limitations under the License.
import logging
import random
from typing import Dict, Optional
import torch
import torch.nn as nn
@@ -77,6 +78,11 @@ class MaskedDiffWithXvec(torch.nn.Module):
# get conditions
conds = torch.zeros(feat.shape, device=token.device)
for i, j in enumerate(feat_len):
if random.random() < 0.5:
continue
index = random.randint(0, int(0.3 * j))
conds[i, :index] = feat[i, :index]
conds = conds.transpose(1, 2)
mask = (~make_pad_mask(feat_len)).to(h)

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@@ -299,7 +299,7 @@ class BaseEncoder(torch.nn.Module):
rate.
3. Currently, nn.Sequential is used to stack all the convolution
layers in subsampling, we need to rewrite it to make it work
with cache, which is not prefered.
with cache, which is not preferred.
Args:
xs (torch.Tensor): (1, max_len, dim)
chunk_size (int): decoding chunk size

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@@ -0,0 +1,198 @@
# set random seed, so that you may reproduce your result.
__set_seed1: !apply:random.seed [1986]
__set_seed2: !apply:numpy.random.seed [1986]
__set_seed3: !apply:torch.manual_seed [1986]
__set_seed4: !apply:torch.cuda.manual_seed_all [1986]
# fixed params
sample_rate: 22050
text_encoder_input_size: 512
llm_input_size: 1024
llm_output_size: 1024
spk_embed_dim: 192
# model params
# for all class/function included in this repo, we use !<name> or !<new> for intialization, so that user may find all corresponding class/function according to one single yaml.
# for system/third_party class/function, we do not require this.
llm: !new:cosyvoice.llm.llm.TransformerLM
text_encoder_input_size: !ref <text_encoder_input_size>
llm_input_size: !ref <llm_input_size>
llm_output_size: !ref <llm_output_size>
text_token_size: 51866
speech_token_size: 4096
length_normalized_loss: True
lsm_weight: 0
spk_embed_dim: !ref <spk_embed_dim>
text_encoder: !new:cosyvoice.transformer.encoder.ConformerEncoder
input_size: !ref <text_encoder_input_size>
output_size: 1024
attention_heads: 8
linear_units: 2048
num_blocks: 3
dropout_rate: 0.1
positional_dropout_rate: 0.1
attention_dropout_rate: 0.0
normalize_before: True
input_layer: 'linear'
pos_enc_layer_type: 'rel_pos_espnet'
selfattention_layer_type: 'rel_selfattn'
use_cnn_module: False
macaron_style: False
use_dynamic_chunk: False
use_dynamic_left_chunk: False
static_chunk_size: 1
llm: !new:cosyvoice.transformer.encoder.TransformerEncoder
input_size: !ref <llm_input_size>
output_size: !ref <llm_output_size>
attention_heads: 8
linear_units: 2048
num_blocks: 7
dropout_rate: 0.1
positional_dropout_rate: 0.1
attention_dropout_rate: 0.0
input_layer: 'linear_legacy'
pos_enc_layer_type: 'rel_pos_espnet'
selfattention_layer_type: 'rel_selfattn'
static_chunk_size: 1
flow: !new:cosyvoice.flow.flow.MaskedDiffWithXvec
input_size: 512
output_size: 80
spk_embed_dim: !ref <spk_embed_dim>
output_type: 'mel'
vocab_size: 4096
input_frame_rate: 50
only_mask_loss: True
encoder: !new:cosyvoice.transformer.encoder.ConformerEncoder
output_size: 512
attention_heads: 4
linear_units: 1024
num_blocks: 3
dropout_rate: 0.1
positional_dropout_rate: 0.1
attention_dropout_rate: 0.1
normalize_before: True
input_layer: 'linear'
pos_enc_layer_type: 'rel_pos_espnet'
selfattention_layer_type: 'rel_selfattn'
input_size: 512
use_cnn_module: False
macaron_style: False
length_regulator: !new:cosyvoice.flow.length_regulator.InterpolateRegulator
channels: 80
sampling_ratios: [1, 1, 1, 1]
decoder: !new:cosyvoice.flow.flow_matching.ConditionalCFM
in_channels: 240
n_spks: 1
spk_emb_dim: 80
cfm_params: !new:omegaconf.DictConfig
content:
sigma_min: 1e-06
solver: 'euler'
t_scheduler: 'cosine'
training_cfg_rate: 0.2
inference_cfg_rate: 0.7
reg_loss_type: 'l1'
estimator: !new:cosyvoice.flow.decoder.ConditionalDecoder
in_channels: 320
out_channels: 80
channels: [256, 256]
dropout: 0.0
attention_head_dim: 64
n_blocks: 4
num_mid_blocks: 8
num_heads: 8
act_fn: 'gelu'
hift: !new:cosyvoice.hifigan.generator.HiFTGenerator
in_channels: 80
base_channels: 512
nb_harmonics: 8
sampling_rate: !ref <sample_rate>
nsf_alpha: 0.1
nsf_sigma: 0.003
nsf_voiced_threshold: 10
upsample_rates: [8, 8]
upsample_kernel_sizes: [16, 16]
istft_params:
n_fft: 16
hop_len: 4
resblock_kernel_sizes: [3, 7, 11]
resblock_dilation_sizes: [[1, 3, 5], [1, 3, 5], [1, 3, 5]]
source_resblock_kernel_sizes: [7, 11]
source_resblock_dilation_sizes: [[1, 3, 5], [1, 3, 5]]
lrelu_slope: 0.1
audio_limit: 0.99
f0_predictor: !new:cosyvoice.hifigan.f0_predictor.ConvRNNF0Predictor
num_class: 1
in_channels: 80
cond_channels: 512
# processor functions
parquet_opener: !name:cosyvoice.dataset.processor.parquet_opener
get_tokenizer: !name:whisper.tokenizer.get_tokenizer
multilingual: True
num_languages: 100
language: 'en'
task: 'transcribe'
allowed_special: 'all'
tokenize: !name:cosyvoice.dataset.processor.tokenize
get_tokenizer: !ref <get_tokenizer>
allowed_special: !ref <allowed_special>
filter: !name:cosyvoice.dataset.processor.filter
max_length: 40960
min_length: 0
token_max_length: 200
token_min_length: 1
resample: !name:cosyvoice.dataset.processor.resample
resample_rate: !ref <sample_rate>
feat_extractor: !name:matcha.utils.audio.mel_spectrogram
n_fft: 1024
num_mels: 80
sampling_rate: !ref <sample_rate>
hop_size: 256
win_size: 1024
fmin: 0
fmax: 8000
center: False
compute_fbank: !name:cosyvoice.dataset.processor.compute_fbank
feat_extractor: !ref <feat_extractor>
parse_embedding: !name:cosyvoice.dataset.processor.parse_embedding
normalize: True
shuffle: !name:cosyvoice.dataset.processor.shuffle
shuffle_size: 1000
sort: !name:cosyvoice.dataset.processor.sort
sort_size: 500 # sort_size should be less than shuffle_size
batch: !name:cosyvoice.dataset.processor.batch
batch_type: 'dynamic'
max_frames_in_batch: 12000
padding: !name:cosyvoice.dataset.processor.padding
use_spk_embedding: False # change to True during sft
# dataset processor pipeline
data_pipeline: [
!ref <parquet_opener>,
!ref <tokenize>,
!ref <filter>,
!ref <resample>,
!ref <compute_fbank>,
!ref <parse_embedding>,
!ref <shuffle>,
!ref <sort>,
!ref <batch>,
!ref <padding>,
]
# train conf
train_conf:
optim: adam
optim_conf:
lr: 0.002 # change to 0.001 if you want to train flow from scratch
scheduler: warmuplr
scheduler_conf:
warmup_steps: 25000
max_epoch: 200
grad_clip: 5
accum_grad: 2
log_interval: 100
save_per_step: -1

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@@ -0,0 +1,198 @@
# set random seed, so that you may reproduce your result.
__set_seed1: !apply:random.seed [1986]
__set_seed2: !apply:numpy.random.seed [1986]
__set_seed3: !apply:torch.manual_seed [1986]
__set_seed4: !apply:torch.cuda.manual_seed_all [1986]
# fixed params
sample_rate: 22050
text_encoder_input_size: 512
llm_input_size: 1024
llm_output_size: 1024
spk_embed_dim: 192
# model params
# for all class/function included in this repo, we use !<name> or !<new> for intialization, so that user may find all corresponding class/function according to one single yaml.
# for system/third_party class/function, we do not require this.
llm: !new:cosyvoice.llm.llm.TransformerLM
text_encoder_input_size: !ref <text_encoder_input_size>
llm_input_size: !ref <llm_input_size>
llm_output_size: !ref <llm_output_size>
text_token_size: 51866
speech_token_size: 4096
length_normalized_loss: True
lsm_weight: 0
spk_embed_dim: !ref <spk_embed_dim>
text_encoder: !new:cosyvoice.transformer.encoder.ConformerEncoder
input_size: !ref <text_encoder_input_size>
output_size: 1024
attention_heads: 16
linear_units: 4096
num_blocks: 6
dropout_rate: 0.1
positional_dropout_rate: 0.1
attention_dropout_rate: 0.0
normalize_before: True
input_layer: 'linear'
pos_enc_layer_type: 'rel_pos_espnet'
selfattention_layer_type: 'rel_selfattn'
use_cnn_module: False
macaron_style: False
use_dynamic_chunk: False
use_dynamic_left_chunk: False
static_chunk_size: 1
llm: !new:cosyvoice.transformer.encoder.TransformerEncoder
input_size: !ref <llm_input_size>
output_size: !ref <llm_output_size>
attention_heads: 16
linear_units: 4096
num_blocks: 14
dropout_rate: 0.1
positional_dropout_rate: 0.1
attention_dropout_rate: 0.0
input_layer: 'linear_legacy'
pos_enc_layer_type: 'rel_pos_espnet'
selfattention_layer_type: 'rel_selfattn'
static_chunk_size: 1
flow: !new:cosyvoice.flow.flow.MaskedDiffWithXvec
input_size: 512
output_size: 80
spk_embed_dim: !ref <spk_embed_dim>
output_type: 'mel'
vocab_size: 4096
input_frame_rate: 50
only_mask_loss: True
encoder: !new:cosyvoice.transformer.encoder.ConformerEncoder
output_size: 512
attention_heads: 8
linear_units: 2048
num_blocks: 6
dropout_rate: 0.1
positional_dropout_rate: 0.1
attention_dropout_rate: 0.1
normalize_before: True
input_layer: 'linear'
pos_enc_layer_type: 'rel_pos_espnet'
selfattention_layer_type: 'rel_selfattn'
input_size: 512
use_cnn_module: False
macaron_style: False
length_regulator: !new:cosyvoice.flow.length_regulator.InterpolateRegulator
channels: 80
sampling_ratios: [1, 1, 1, 1]
decoder: !new:cosyvoice.flow.flow_matching.ConditionalCFM
in_channels: 240
n_spks: 1
spk_emb_dim: 80
cfm_params: !new:omegaconf.DictConfig
content:
sigma_min: 1e-06
solver: 'euler'
t_scheduler: 'cosine'
training_cfg_rate: 0.2
inference_cfg_rate: 0.7
reg_loss_type: 'l1'
estimator: !new:cosyvoice.flow.decoder.ConditionalDecoder
in_channels: 320
out_channels: 80
channels: [256, 256]
dropout: 0.0
attention_head_dim: 64
n_blocks: 4
num_mid_blocks: 12
num_heads: 8
act_fn: 'gelu'
hift: !new:cosyvoice.hifigan.generator.HiFTGenerator
in_channels: 80
base_channels: 512
nb_harmonics: 8
sampling_rate: !ref <sample_rate>
nsf_alpha: 0.1
nsf_sigma: 0.003
nsf_voiced_threshold: 10
upsample_rates: [8, 8]
upsample_kernel_sizes: [16, 16]
istft_params:
n_fft: 16
hop_len: 4
resblock_kernel_sizes: [3, 7, 11]
resblock_dilation_sizes: [[1, 3, 5], [1, 3, 5], [1, 3, 5]]
source_resblock_kernel_sizes: [7, 11]
source_resblock_dilation_sizes: [[1, 3, 5], [1, 3, 5]]
lrelu_slope: 0.1
audio_limit: 0.99
f0_predictor: !new:cosyvoice.hifigan.f0_predictor.ConvRNNF0Predictor
num_class: 1
in_channels: 80
cond_channels: 512
# processor functions
parquet_opener: !name:cosyvoice.dataset.processor.parquet_opener
get_tokenizer: !name:whisper.tokenizer.get_tokenizer
multilingual: True
num_languages: 100
language: 'en'
task: 'transcribe'
allowed_special: 'all'
tokenize: !name:cosyvoice.dataset.processor.tokenize
get_tokenizer: !ref <get_tokenizer>
allowed_special: !ref <allowed_special>
filter: !name:cosyvoice.dataset.processor.filter
max_length: 40960
min_length: 0
token_max_length: 200
token_min_length: 1
resample: !name:cosyvoice.dataset.processor.resample
resample_rate: !ref <sample_rate>
feat_extractor: !name:matcha.utils.audio.mel_spectrogram
n_fft: 1024
num_mels: 80
sampling_rate: !ref <sample_rate>
hop_size: 256
win_size: 1024
fmin: 0
fmax: 8000
center: False
compute_fbank: !name:cosyvoice.dataset.processor.compute_fbank
feat_extractor: !ref <feat_extractor>
parse_embedding: !name:cosyvoice.dataset.processor.parse_embedding
normalize: True
shuffle: !name:cosyvoice.dataset.processor.shuffle
shuffle_size: 1000
sort: !name:cosyvoice.dataset.processor.sort
sort_size: 500 # sort_size should be less than shuffle_size
batch: !name:cosyvoice.dataset.processor.batch
batch_type: 'dynamic'
max_frames_in_batch: 2000
padding: !name:cosyvoice.dataset.processor.padding
use_spk_embedding: False # change to True during sft
# dataset processor pipeline
data_pipeline: [
!ref <parquet_opener>,
!ref <tokenize>,
!ref <filter>,
!ref <resample>,
!ref <compute_fbank>,
!ref <parse_embedding>,
!ref <shuffle>,
!ref <sort>,
!ref <batch>,
!ref <padding>,
]
# train conf
train_conf:
optim: adam
optim_conf:
lr: 0.001 # change to 1e-5 during sft
scheduler: warmuplr # change to constantlr during sft
scheduler_conf:
warmup_steps: 2500
max_epoch: 200
grad_clip: 5
accum_grad: 2
log_interval: 100
save_per_step: -1

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@@ -0,0 +1,42 @@
{
"train_micro_batch_size_per_gpu": 1,
"gradient_accumulation_steps": 1,
"steps_per_print": 100,
"gradient_clipping": 5,
"fp16": {
"enabled": false,
"auto_cast": false,
"loss_scale": 0,
"initial_scale_power": 16,
"loss_scale_window": 256,
"hysteresis": 2,
"consecutive_hysteresis": false,
"min_loss_scale": 1
},
"bf16": {
"enabled": false
},
"zero_force_ds_cpu_optimizer": false,
"zero_optimization": {
"stage": 2,
"offload_optimizer": {
"device": "none",
"pin_memory": true
},
"allgather_partitions": true,
"allgather_bucket_size": 5e8,
"overlap_comm": false,
"reduce_scatter": true,
"reduce_bucket_size": 5e8,
"contiguous_gradients" : true
},
"optimizer": {
"type": "AdamW",
"params": {
"lr": 0.001,
"weight_decay": 0.0001,
"torch_adam": true,
"adam_w_mode": true
}
}
}

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@@ -0,0 +1 @@
../../../cosyvoice

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@@ -0,0 +1,97 @@
#!/bin/bash
# Copyright 2014 Johns Hopkins University (author: Daniel Povey)
# Apache 2.0
remove_archive=false
if [ "$1" == --remove-archive ]; then
remove_archive=true
shift
fi
if [ $# -ne 3 ]; then
echo "Usage: $0 [--remove-archive] <data-base> <url-base> <corpus-part>"
echo "e.g.: $0 /export/a15/vpanayotov/data www.openslr.org/resources/11 dev-clean"
echo "With --remove-archive it will remove the archive after successfully un-tarring it."
echo "<corpus-part> can be one of: dev-clean, test-clean, dev-other, test-other,"
echo " train-clean-100, train-clean-360, train-other-500."
exit 1
fi
data=$1
url=$2
part=$3
if [ ! -d "$data" ]; then
echo "$0: no such directory $data"
exit 1
fi
part_ok=false
list="dev_set test_set train_set"
for x in $list; do
if [ "$part" == $x ]; then part_ok=true; fi
done
if ! $part_ok; then
echo "$0: expected <corpus-part> to be one of $list, but got '$part'"
exit 1
fi
if [ -z "$url" ]; then
echo "$0: empty URL base."
exit 1
fi
if [ -f $data/.$part.complete ]; then
echo "$0: data part $part was already successfully extracted, nothing to do."
exit 0
fi
# sizes of the archive files in bytes. This is some older versions.
sizes_old="1035537823 2201936013 52627842921"
# sizes_new is the archive file sizes of the final release. Some of these sizes are of
# things we probably won't download.
sizes_new="3886385"
if [ -f $data/$part.tar.gz ]; then
size=$(/bin/ls -l $data/$part.tar.gz | awk '{print $5}')
size_ok=false
for s in $sizes_old $sizes_new; do if [ $s == $size ]; then size_ok=true; fi; done
if ! $size_ok; then
echo "$0: removing existing file $data/$part.tar.gz because its size in bytes $size"
echo "does not equal the size of one of the archives."
rm $data/$part.tar.gz
else
echo "$data/$part.tar.gz exists and appears to be complete."
fi
fi
if [ ! -f $data/$part.tar.gz ]; then
if ! which wget >/dev/null; then
echo "$0: wget is not installed."
exit 1
fi
full_url=$url/$part.tar.gz
echo "$0: downloading data from $full_url. This may take some time, please be patient."
if ! wget -P $data --no-check-certificate $full_url; then
echo "$0: error executing wget $full_url"
exit 1
fi
fi
if ! tar -C $data -xvzf $data/$part.tar.gz; then
echo "$0: error un-tarring archive $data/$part.tar.gz"
exit 1
fi
touch $data/.$part.complete
echo "$0: Successfully downloaded and un-tarred $data/$part.tar.gz"
if $remove_archive; then
echo "$0: removing $data/$part.tar.gz file since --remove-archive option was supplied."
rm $data/$part.tar.gz
fi

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@@ -0,0 +1,50 @@
import argparse
import logging
import os
from tqdm import tqdm
logger = logging.getLogger()
def main():
utt2wav, utt2text, utt2spk, spk2utt = {}, {}, {}, {}
with open(os.path.join(args.src_dir, "TRANS.txt"), "r") as f:
lines = f.readlines()[1:]
lines = [l.split('\t') for l in lines]
for wav, spk, content in tqdm(lines):
wav, spk, content = wav.strip(), spk.strip(), content.strip()
content = content.replace('[FIL]', '')
content = content.replace('[SPK]', '')
wav = os.path.join(args.src_dir, spk, wav)
if not os.path.exists(wav):
continue
utt = os.path.basename(wav).replace('.wav', '')
utt2wav[utt] = wav
utt2text[utt] = content
utt2spk[utt] = spk
if spk not in spk2utt:
spk2utt[spk] = []
spk2utt[spk].append(utt)
with open('{}/wav.scp'.format(args.des_dir), 'w') as f:
for k, v in utt2wav.items():
f.write('{} {}\n'.format(k, v))
with open('{}/text'.format(args.des_dir), 'w') as f:
for k, v in utt2text.items():
f.write('{} {}\n'.format(k, v))
with open('{}/utt2spk'.format(args.des_dir), 'w') as f:
for k, v in utt2spk.items():
f.write('{} {}\n'.format(k, v))
with open('{}/spk2utt'.format(args.des_dir), 'w') as f:
for k, v in spk2utt.items():
f.write('{} {}\n'.format(k, ' '.join(v)))
return
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument('--src_dir',
type=str)
parser.add_argument('--des_dir',
type=str)
args = parser.parse_args()
main()

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@@ -0,0 +1,3 @@
# NOTE(kan-bayashi): Use UTF-8 in Python to avoid UnicodeDecodeError when LC_ALL=C
export PYTHONIOENCODING=UTF-8
export PYTHONPATH=../../../:../../../third_party/Matcha-TTS:$PYTHONPATH

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#!/bin/bash
# Copyright 2024 Alibaba Inc. All Rights Reserved.
. ./path.sh || exit 1;
stage=-1
stop_stage=3
data_url=www.openslr.org/resources/68
data_dir=/mnt/hengwu.zty/data/tts/openslr/magicdata-read
pretrained_model_dir=../../../pretrained_models/CosyVoice-300M
if [ ${stage} -le -1 ] && [ ${stop_stage} -ge -1 ]; then
echo "Data Download"
for part in dev_set test_set train_set; do
local/download_and_untar.sh ${data_dir} ${data_url} ${part}
done
fi
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
echo "Data preparation, prepare wav.scp/text/utt2spk/spk2utt"
for x in dev test train; do
mkdir -p data/$x
python local/prepare_data.py --src_dir $data_dir/$x --des_dir data/$x
done
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
echo "Extract campplus speaker embedding, you will get spk2embedding.pt and utt2embedding.pt in data/$x dir"
for x in dev test train; do
tools/extract_embedding.py --dir data/$x \
--onnx_path $pretrained_model_dir/campplus.onnx
done
fi
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
echo "Extract discrete speech token, you will get utt2speech_token.pt in data/$x dir"
for x in dev test train; do
tools/extract_speech_token.py --dir data/$x \
--onnx_path $pretrained_model_dir/speech_tokenizer_v1.onnx
done
fi
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
echo "Prepare required parquet format data, you should have prepared wav.scp/text/utt2spk/spk2utt/utt2embedding.pt/spk2embedding.pt/utt2speech_token.pt"
for x in dev test train; do
mkdir -p data/$x/parquet
tools/make_parquet_list.py --num_utts_per_parquet 1000 \
--num_processes 10 \
--src_dir data/$x \
--des_dir data/$x/parquet
done
fi
# inference
if [ ${stage} -le 4 ] && [ ${stop_stage} -ge 4 ]; then
echo "Run inference. Please make sure utt in tts_text is in prompt_data"
for mode in sft zero_shot; do
python cosyvoice/bin/inference.py --mode $mode \
--gpu 0 \
--config conf/cosyvoice.yaml \
--prompt_data data/test/parquet/data.list \
--prompt_utt2data data/test/parquet/utt2data.list \
--tts_text `pwd`/tts_text.json \
--llm_model $pretrained_model_dir/llm.pt \
--flow_model $pretrained_model_dir/flow.pt \
--hifigan_model $pretrained_model_dir/hift.pt \
--result_dir `pwd`/exp/cosyvoice/test/$mode
done
fi
# train llm
export CUDA_VISIBLE_DEVICES="0,1,2,3"
num_gpus=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
job_id=1986
dist_backend="nccl"
num_workers=2
prefetch=100
train_engine=torch_ddp
if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
echo "Run train. We only support llm traning for now. If your want to train from scratch, please use conf/cosyvoice.fromscratch.yaml"
if [ $train_engine == 'deepspeed' ]; then
echo "Notice deepspeed has its own optimizer config. Modify conf/ds_stage2.json if necessary"
fi
cp data/train/parquet/data.list data/train.data.list
cp data/dev/parquet/data.list data/dev.data.list
for model in llm; do
torchrun --nnodes=1 --nproc_per_node=$num_gpus \
--rdzv_id=$job_id --rdzv_backend="c10d" --rdzv_endpoint="localhost:0" \
cosyvoice/bin/train.py \
--train_engine $train_engine \
--config conf/cosyvoice.yaml \
--train_data data/train.data.list \
--cv_data data/dev.data.list \
--model $model \
--checkpoint $pretrained_model_dir/$model.pt \
--model_dir `pwd`/exp/cosyvoice/$model/$train_engine \
--tensorboard_dir `pwd`/tensorboard/cosyvoice/$model/$train_engine \
--ddp.dist_backend $dist_backend \
--num_workers ${num_workers} \
--prefetch ${prefetch} \
--pin_memory \
--deepspeed_config ./conf/ds_stage2.json \
--deepspeed.save_states model+optimizer
done
fi

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../../../tools

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{
"38_5718_20170915093303": [
"我想这出最好歌曲把歌词发到网上请别人帮我作曲急急",
"叫他明天早上差五分儿九点去机场"
],
"38_5721_20170915091235": [
"变温室调到零下两度档",
"交谈中请勿轻信汇款信息陌生电话请勿使用外挂软件"
],
"38_5733_20170915130323": [
"这是老鹰乐队的一首经典歌曲",
"我急用这段音乐我自己找到一段但是有现场杂音"
],
"38_5836_20170916221414": [
"给我播一个陶喆的专辑",
"这套餐好贵呀我发这么多短信贵死了"
]
}