Training in progress - step 30500
Browse files- asr_config.py +3 -1
- asr_modeling.py +4 -5
- asr_pipeline.py +0 -28
- asr_processing.py +2 -0
- model.safetensors +1 -1
asr_config.py
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@@ -18,6 +18,7 @@ class ASRConfig(transformers.PretrainedConfig):
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user_prompt: str = "Please transcribe this English audio into text: <audio>",
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encoder_dim: Optional[int] = None,
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llm_dim: Optional[int] = None,
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# Encoder conv layers: list of (padding, kernel_size, stride) tuples
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# Default is Whisper/GLM-ASR structure: conv1(k=3,s=1,p=1) + conv2(k=3,s=2,p=1)
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encoder_conv_layers: Optional[list] = None,
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@@ -51,7 +52,7 @@ class ASRConfig(transformers.PretrainedConfig):
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# Set default generation parameters (greedy decoding only)
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generation_defaults = {
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"num_beams": 1,
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"max_new_tokens":
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"repetition_penalty": 1.0,
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"length_penalty": 1.0,
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"no_repeat_ngram_size": 0,
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@@ -69,6 +70,7 @@ class ASRConfig(transformers.PretrainedConfig):
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self.user_prompt = user_prompt
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self.encoder_dim = encoder_dim
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self.llm_dim = llm_dim
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# Default conv layers for Whisper/GLM-ASR: [(pad, kernel, stride), ...]
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self.encoder_conv_layers = encoder_conv_layers or [(1, 3, 1), (1, 3, 2)]
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self.audio_sample_rate = audio_sample_rate
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user_prompt: str = "Please transcribe this English audio into text: <audio>",
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encoder_dim: Optional[int] = None,
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llm_dim: Optional[int] = None,
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encoder_stride: int = 2, # Temporal downsampling factor of audio encoder (legacy, use encoder_conv_layers)
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# Encoder conv layers: list of (padding, kernel_size, stride) tuples
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# Default is Whisper/GLM-ASR structure: conv1(k=3,s=1,p=1) + conv2(k=3,s=2,p=1)
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encoder_conv_layers: Optional[list] = None,
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# Set default generation parameters (greedy decoding only)
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generation_defaults = {
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"num_beams": 1,
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"max_new_tokens": 96,
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"repetition_penalty": 1.0,
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"length_penalty": 1.0,
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"no_repeat_ngram_size": 0,
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self.user_prompt = user_prompt
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self.encoder_dim = encoder_dim
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self.llm_dim = llm_dim
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self.encoder_stride = encoder_stride
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# Default conv layers for Whisper/GLM-ASR: [(pad, kernel, stride), ...]
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self.encoder_conv_layers = encoder_conv_layers or [(1, 3, 1), (1, 3, 2)]
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self.audio_sample_rate = audio_sample_rate
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asr_modeling.py
CHANGED
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@@ -96,6 +96,7 @@ class ASRModel(PreTrainedModel, GenerationMixin):
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super().__init__(config)
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self.system_prompt = config.system_prompt
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target_dtype = getattr(torch, config.model_dtype)
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# Audio encoder (frozen)
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@@ -120,10 +121,7 @@ class ASRModel(PreTrainedModel, GenerationMixin):
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self.generation_config.length_penalty = config.length_penalty
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self.generation_config.repetition_penalty = config.repetition_penalty
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self.generation_config.no_repeat_ngram_size = config.no_repeat_ngram_size
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self.generation_config.eos_token_id =
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self.tokenizer.convert_tokens_to_ids("<|im_end|>"),
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self.tokenizer.convert_tokens_to_ids("<|endoftext|>"),
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]
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self.generation_config.pad_token_id = self.tokenizer.pad_token_id
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# Feature extractor for audio preprocessing
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@@ -147,7 +145,7 @@ class ASRModel(PreTrainedModel, GenerationMixin):
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encoder_kwargs = {
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"attn_implementation": config.attn_implementation,
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"low_cpu_mem_usage": True,
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"
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}
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if "whisper" in config.audio_model_id.lower():
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@@ -298,6 +296,7 @@ class ASRModel(PreTrainedModel, GenerationMixin):
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feature_extractor=self.feature_extractor,
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tokenizer=self.tokenizer,
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projector=self.projector,
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encoder_conv_layers=self.config.encoder_conv_layers,
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)
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super().__init__(config)
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self.system_prompt = config.system_prompt
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self.encoder_stride = config.encoder_stride
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target_dtype = getattr(torch, config.model_dtype)
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# Audio encoder (frozen)
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self.generation_config.length_penalty = config.length_penalty
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self.generation_config.repetition_penalty = config.repetition_penalty
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self.generation_config.no_repeat_ngram_size = config.no_repeat_ngram_size
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self.generation_config.eos_token_id = self.tokenizer.convert_tokens_to_ids("<|im_end|>")
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self.generation_config.pad_token_id = self.tokenizer.pad_token_id
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# Feature extractor for audio preprocessing
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encoder_kwargs = {
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"attn_implementation": config.attn_implementation,
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"low_cpu_mem_usage": True,
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"torch_dtype": dtype,
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}
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if "whisper" in config.audio_model_id.lower():
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feature_extractor=self.feature_extractor,
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tokenizer=self.tokenizer,
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projector=self.projector,
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encoder_stride=self.encoder_stride,
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encoder_conv_layers=self.config.encoder_conv_layers,
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)
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asr_pipeline.py
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@@ -476,32 +476,4 @@ class ASRPipeline(transformers.AutomaticSpeechRecognitionPipeline):
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text = self.tokenizer.decode(tokens, skip_special_tokens=True).strip()
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# Strip <think>...</think> tags (Qwen3 doesn't respect /no_think prompt)
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text = re.sub(r"<think>.*?</think>\s*", "", text, flags=re.DOTALL).strip()
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# Truncate if a word repeats more than 3 times consecutively
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text = self._truncate_repetitions(text, max_repeats=3)
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return {"text": text}
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def _truncate_repetitions(self, text: str, max_repeats: int = 3) -> str:
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"""Truncate text when a word repeats more than max_repeats times consecutively.
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Args:
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text: Input text to check for repetitions
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max_repeats: Maximum allowed consecutive repetitions (default 3)
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Returns:
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Truncated text if repetition detected, otherwise original text
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"""
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words = text.split()
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if len(words) <= max_repeats:
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return text
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repeat_count = 1
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for i in range(1, len(words)):
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if words[i].lower() == words[i - 1].lower():
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repeat_count += 1
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if repeat_count > max_repeats:
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# Keep up to max_repeats of the repeated word
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return " ".join(words[:i])
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else:
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repeat_count = 1
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return text
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text = self.tokenizer.decode(tokens, skip_special_tokens=True).strip()
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# Strip <think>...</think> tags (Qwen3 doesn't respect /no_think prompt)
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text = re.sub(r"<think>.*?</think>\s*", "", text, flags=re.DOTALL).strip()
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return {"text": text}
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asr_processing.py
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@@ -26,12 +26,14 @@ class ASRProcessor(ProcessorMixin):
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feature_extractor,
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tokenizer,
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projector=None,
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encoder_conv_layers: Optional[list] = None,
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):
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self.feature_extractor = feature_extractor
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self.tokenizer = tokenizer
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self.audio_token_id = tokenizer.convert_tokens_to_ids(self.AUDIO_TOKEN)
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self.projector = projector
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self.encoder_conv_layers = encoder_conv_layers or self.DEFAULT_ENCODER_CONV_LAYERS
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def _compute_encoder_output_length(self, mel_length: int) -> int:
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feature_extractor,
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tokenizer,
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projector=None,
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encoder_stride: int = 2,
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encoder_conv_layers: Optional[list] = None,
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):
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self.feature_extractor = feature_extractor
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self.tokenizer = tokenizer
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self.audio_token_id = tokenizer.convert_tokens_to_ids(self.AUDIO_TOKEN)
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self.projector = projector
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self.encoder_stride = encoder_stride # Legacy, kept for compatibility
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self.encoder_conv_layers = encoder_conv_layers or self.DEFAULT_ENCODER_CONV_LAYERS
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def _compute_encoder_output_length(self, mel_length: int) -> int:
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model.safetensors
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@@ -1,3 +1,3 @@
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version https://git-lfs.github.com/spec/v1
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-
oid sha256:
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size 58732960
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version https://git-lfs.github.com/spec/v1
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+
oid sha256:c5ce422bc8492d610e980da968fec9e97cc628c00a928b8d9cdc24197ab5910c
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size 58732960
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