From 59bc02b089f7a626fe67907dcfc695eae6883f82 Mon Sep 17 00:00:00 2001
From: 游雁 <zhifu.gzf@alibaba-inc.com>
Date: 星期五, 14 六月 2024 13:59:49 +0800
Subject: [PATCH] decoding
---
funasr/models/sense_voice/model.py | 2 ++
funasr/models/llm_asr/model.py | 31 +++++++++++++++++++++++++++----
examples/industrial_data_pretraining/llm_asr/conf/whisper_qwen_linear2.yaml | 2 +-
funasr/datasets/openai_datasets/datasets.py | 11 ++++++++---
4 files changed, 38 insertions(+), 8 deletions(-)
diff --git a/examples/industrial_data_pretraining/llm_asr/conf/whisper_qwen_linear2.yaml b/examples/industrial_data_pretraining/llm_asr/conf/whisper_qwen_linear2.yaml
index 483f219..48bd0cf 100644
--- a/examples/industrial_data_pretraining/llm_asr/conf/whisper_qwen_linear2.yaml
+++ b/examples/industrial_data_pretraining/llm_asr/conf/whisper_qwen_linear2.yaml
@@ -69,7 +69,7 @@
batch_size_scale_ratio_max: 2
num_workers: 4
audio_adaptor_downsample_rate: ${audio_adaptor_conf.downsample_rate}
- audio_encoder_downsample_rate: 2
+ audio_encoder_downsample_rate: 4
data_split_num: 512
batch_size_sample_max: 15
retry: 20
diff --git a/funasr/datasets/openai_datasets/datasets.py b/funasr/datasets/openai_datasets/datasets.py
index 39b8453..7300b9d 100644
--- a/funasr/datasets/openai_datasets/datasets.py
+++ b/funasr/datasets/openai_datasets/datasets.py
@@ -64,6 +64,8 @@
self.max_token_length = kwargs.get("max_token_length", 1024)
self.batch_size_scale_ratio_max = kwargs.get("batch_size_scale_ratio_max", 1.5)
self.batch_size_token_max = kwargs.get("batch_size_token_max", 2500)
+ self.audio_adaptor_downsample_rate = kwargs.get("audio_adaptor_downsample_rate", 2)
+ self.audio_encoder_downsample_rate = kwargs.get("audio_encoder_downsample_rate", 4)
def get_source_len(self, index):
item = self.index_ds[index]
@@ -136,10 +138,13 @@
speech = speech.permute(0, 2, 1)
# if speech_lengths > self.batch_size:
# continue
+ if self.audio_encoder_downsample_rate == 4:
+ olens = 1 + (speech_lengths[0].item() - 3 + 2 * 1) // 2
+ olens = 1 + (olens - 3 + 2 * 1) // 2
+ elif self.audio_encoder_downsample_rate == 1:
+ olens = speech_lengths[0].item()
- olens = 1 + (speech_lengths[0].item() - 3 + 2 * 1) // 2
- olens = 1 + (olens - 3 + 2 * 1) // 2
- sub_token_len = (olens - 1) // 2 + 1
+ sub_token_len = (olens - 1) // self.audio_adaptor_downsample_rate + 1
sub_token = [0] * sub_token_len
fbank_beg_i = [len(source_ids)]
source_ids += sub_token
diff --git a/funasr/models/llm_asr/model.py b/funasr/models/llm_asr/model.py
index 45e56c3..84d7d33 100644
--- a/funasr/models/llm_asr/model.py
+++ b/funasr/models/llm_asr/model.py
@@ -498,9 +498,7 @@
with torch.cuda.amp.autocast(enabled=False):
# audio encoder
- encoder_out, encoder_out_lens = self.audio_encoder(
- speech.permute(0, 2, 1), speech_lengths
- )
+ encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
# audio_adaptor
encoder_out, encoder_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
@@ -565,6 +563,12 @@
batch_size = int((labels_ids > 0 + 1).sum())
loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device)
return loss, stats, weight
+
+ def encode(self, speech, speech_lengths):
+ # audio encoder
+ encoder_out, encoder_out_lens = self.audio_encoder(speech.permute(0, 2, 1), speech_lengths)
+
+ return encoder_out, encoder_out_lens
def data_template(self, data):
system, user, assistant = [], [], []
@@ -721,7 +725,8 @@
speech = speech.to(torch.float16)
elif kwargs.get("bf16", False):
speech = speech.to(torch.bfloat16)
- encoder_out, encoder_out_lens = self.audio_encoder(speech.permute(0, 2, 1), speech_lengths)
+ # audio encoder
+ encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
# audio_adaptor
encoder_out, encoder_out_lens = self.audio_adaptor(encoder_out, encoder_out_lens)
@@ -806,3 +811,21 @@
ibest_writer["text_tn"][key[0]] = response_clean
return results, meta_data
+
+
+@tables.register("model_classes", "LLMASR3")
+class LLMASR3(nn.Module):
+ """ """
+
+ def __init__(
+ self,
+ *args,
+ **kwargs,
+ ):
+
+ super().__init__(*args, **kwargs)
+
+ def encode(self, speech, speech_lengths):
+ # audio encoder
+ encoder_out, encoder_out_lens = self.audio_encoder(speech, speech_lengths)
+ return encoder_out, encoder_out_lens
diff --git a/funasr/models/sense_voice/model.py b/funasr/models/sense_voice/model.py
index 9be5abe..c77930d 100644
--- a/funasr/models/sense_voice/model.py
+++ b/funasr/models/sense_voice/model.py
@@ -1042,6 +1042,7 @@
self.length_normalized_loss = length_normalized_loss
self.beam_search = None
self.activation_checkpoint = kwargs.get("activation_checkpoint", False)
+ self.encoder_output_size = encoder_output_size
def forward(
self,
@@ -1451,6 +1452,7 @@
self.ctc = ctc
self.length_normalized_loss = length_normalized_loss
+ self.encoder_output_size = encoder_output_size
def forward(
self,
--
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