zhifu gao
2024-02-01 2ddfc27d5b69e5c1a38021554e97dae958328c20
Funasr1.0 (#1343)

* funasr1.0.5

* funasr1.0.5 audio samples input

* batch_type token

* batch_type token
4个文件已修改
23 ■■■■■ 已修改文件
examples/industrial_data_pretraining/seaco_paraformer/demo.py 3 ●●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/bin/train.py 2 ●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/datasets/audio_datasets/samplers.py 10 ●●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/models/whisper/model.py 8 ●●●● 补丁 | 查看 | 原始文档 | blame | 历史
examples/industrial_data_pretraining/seaco_paraformer/demo.py
@@ -25,6 +25,7 @@
# example2
import torchaudio
import os
wav_file = os.path.join(model.model_path, "example/asr_example.wav")
input_tensor, sample_rate = torchaudio.load(wav_file)
input_tensor = input_tensor.mean(0)
@@ -33,7 +34,7 @@
# example3
import soundfile
import os
wav_file = os.path.join(model.model_path, "example/asr_example.wav")
speech, sample_rate = soundfile.read(wav_file)
res = model.generate(input=[speech], batch_size_s=300, is_final=True)
funasr/bin/train.py
@@ -154,7 +154,7 @@
    if batch_sampler is not None:
        batch_sampler_class = tables.batch_sampler_classes.get(batch_sampler)
        batch_sampler = batch_sampler_class(dataset_tr, **kwargs.get("dataset_conf"))
        batch_sampler_val = batch_sampler_class(dataset_tr, is_training=False, **kwargs.get("dataset_conf"))
        batch_sampler_val = batch_sampler_class(dataset_val, is_training=False, **kwargs.get("dataset_conf"))
    dataloader_tr = torch.utils.data.DataLoader(dataset_tr,
                                                collate_fn=dataset_tr.collator,
                                                batch_sampler=batch_sampler,
funasr/datasets/audio_datasets/samplers.py
@@ -26,6 +26,8 @@
        self.max_token_length = kwargs.get("max_token_length", 5000)
        self.shuffle_idx = np.arange(self.total_samples)
        self.shuffle = shuffle and is_training
        self.length_scale_source = kwargs.get("length_scale_source", 1.0)
    
    def __len__(self):
        return (self.total_samples-1) // self.batch_size + 1
@@ -53,8 +55,10 @@
                
                idx_map = self.shuffle_idx[idx]
                # prompt = self.dataset.indexed_dataset[idx_map]["prompt"]
                sample_len_cur = self.dataset.get_source_len(idx_map) + \
                                 self.dataset.get_target_len(idx_map)
                target_len = self.dataset.get_target_len(idx_map) if self.batch_type == 'length' else 0.0
                source_len = self.dataset.get_source_len(idx_map) / self.length_scale_source
                sample_len_cur = source_len + target_len
                
                datalen_with_index.append([idx, sample_len_cur])
            
@@ -66,7 +70,7 @@
                
                max_token_cur = max(max_token, sample_len_cur_raw)
                max_token_padding = 1 + num_sample
                if self.batch_type == 'length':
                if self.batch_type != 'example':
                    max_token_padding *= max_token_cur
                if max_token_padding <= self.batch_size:
                    batch.append(idx)
funasr/models/whisper/model.py
@@ -10,6 +10,8 @@
from funasr.models.whisper.utils.decoding import detect_language as detect_language_function, decode as decode_function
from funasr.register import tables
@dataclass
class ModelDimensions:
@@ -128,6 +130,8 @@
        return x
@tables.register("encoder_classes", "WhisperEncoder")
class AudioEncoder(nn.Module):
    def __init__(self, n_mels: int, n_ctx: int, n_state: int, n_head: int, n_layer: int):
        super().__init__()
@@ -158,7 +162,7 @@
        x = self.ln_post(x)
        return x
@tables.register("decoder_classes", "WhisperDecoder")
class TextDecoder(nn.Module):
    def __init__(self, n_vocab: int, n_ctx: int, n_state: int, n_head: int, n_layer: int):
        super().__init__()
@@ -193,7 +197,7 @@
        return logits
@tables.register("model_classes", "Whisper")
class Whisper(nn.Module):
    def __init__(self, dims: dict):
        super().__init__()