游雁
2024-06-07 b2be308de0a4d75c3645e55c26d33a58446a16ff
auto frontend
1个文件已修改
9 ■■■■ 已修改文件
funasr/models/llm_asr/model.py 9 ●●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/models/llm_asr/model.py
@@ -468,7 +468,7 @@
        if len(speech_lengths.size()) > 1:
            speech_lengths = speech_lengths[:, 0]
        batch_size = speech.shape[0]
        batch_size, frames, _ = speech.shape
        # audio encoder
        encoder_out, encoder_out_lens = self.audio_encoder(speech.permute(0, 2, 1), speech_lengths)
@@ -499,6 +499,13 @@
            stats["acc"] = acc_att
        stats["loss"] = torch.clone(loss.detach())
        stats["batch_size"] = batch_size
        stats["batch_size_x_frames"] = frames * batch_size
        stats["batch_size_real_frames"] = speech_lengths.sum().item()
        stats["padding_frames"] = stats["batch_size_x_frames"] - stats["batch_size_real_frames"]
        stats["batch_size_x_tokens"] = token_num * batch_size
        stats["batch_size_real_tokens"] = attention_mask.sum().item()
        stats["padding_tokens"] = stats["batch_size_x_tokens"] - stats["batch_size_real_tokens"]
        # force_gatherable: to-device and to-tensor if scalar for DataParallel
        if self.length_normalized_loss: