Update processing_gemma3_omni.py
Browse files- processing_gemma3_omni.py +115 -111
processing_gemma3_omni.py
CHANGED
@@ -1,5 +1,5 @@
|
|
1 |
import re
|
2 |
-
from typing import List, Optional, Union, Dict, Any
|
3 |
|
4 |
import math
|
5 |
import numpy as np
|
@@ -15,17 +15,16 @@ AudioInput = Union[np.ndarray, List[float], Tuple[np.ndarray, int]]
|
|
15 |
|
16 |
from transformers.feature_extraction_sequence_utils import SequenceFeatureExtractor
|
17 |
from transformers.feature_extraction_utils import BatchFeature
|
18 |
-
from transformers.processing_utils import ProcessorMixin, ProcessingKwargs
|
19 |
from transformers.utils import TensorType, to_py_obj, logging
|
20 |
# For AutoImageProcessor, AutoTokenizer if needed for default loading
|
21 |
from transformers import AutoImageProcessor, AutoTokenizer
|
22 |
|
23 |
-
|
24 |
# Constants (as defined before)
|
25 |
DEFAULT_SAMPLING_RATE = 16000
|
26 |
DEFAULT_N_FFT = 512
|
27 |
-
DEFAULT_WIN_LENGTH = 400
|
28 |
-
DEFAULT_HOP_LENGTH = 160
|
29 |
DEFAULT_N_MELS = 80
|
30 |
DEFAULT_COMPRESSION_RATE = 4
|
31 |
DEFAULT_QFORMER_RATE = 2
|
@@ -48,15 +47,15 @@ def create_mel_filterbank(sampling_rate: int, n_fft: int, n_mels: int, fmin: flo
|
|
48 |
if fmin >= fmax:
|
49 |
raise ValueError(f"fmin ({fmin}) must be smaller than fmax ({fmax}).")
|
50 |
|
51 |
-
def hz_to_mel(f: float) -> float:
|
52 |
return 2595.0 * math.log10(1 + f / 700.0)
|
53 |
|
54 |
def mel_to_hz(mel: float) -> float:
|
55 |
-
return 700.0 * (10**(mel / 2595.0) - 1)
|
56 |
|
57 |
mel_points = np.linspace(hz_to_mel(fmin), hz_to_mel(fmax), n_mels + 2)
|
58 |
freq_points = mel_to_hz(mel_points)
|
59 |
-
|
60 |
freq_points = np.clip(freq_points, 0, sampling_rate / 2.0)
|
61 |
bins = np.floor((n_fft / 2.0) * freq_points / (sampling_rate / 2.0)).astype(int)
|
62 |
bins = np.clip(bins, 0, n_fft // 2)
|
@@ -64,43 +63,44 @@ def create_mel_filterbank(sampling_rate: int, n_fft: int, n_mels: int, fmin: flo
|
|
64 |
filterbank = np.zeros((n_mels, n_fft // 2 + 1), dtype=np.float32)
|
65 |
for m in range(n_mels):
|
66 |
left, center, right = bins[m], bins[m + 1], bins[m + 2]
|
67 |
-
|
68 |
# Simplified triangle creation logic (more robust versions exist in libraries like librosa)
|
69 |
if center > left:
|
70 |
-
filterbank[m, left:center+1] = (np.arange(left, center + 1) - left) / (center - left)
|
71 |
if right > center:
|
72 |
-
filterbank[m, center:right+1] = (right - np.arange(center, right + 1)) / (right - center)
|
73 |
# Ensure peak is 1 if multiple points coincide at center (can happen with narrow filters/low resolution)
|
74 |
-
if left <= center <= right and filterbank[m,center] < 1.0 and (
|
|
|
75 |
# if filterbank[m,center] is not properly set to 1 by slopes (e.g. left==center or right==center)
|
76 |
-
|
77 |
-
|
78 |
-
|
79 |
-
|
80 |
-
|
81 |
-
|
82 |
-
|
83 |
|
84 |
return filterbank
|
85 |
|
|
|
86 |
# Gemma3AudioFeatureExtractor class (assuming it's correctly defined from previous response)
|
87 |
# ... (Gemma3AudioFeatureExtractor class from the previous corrected response) ...
|
88 |
class Gemma3AudioFeatureExtractor(SequenceFeatureExtractor):
|
89 |
-
model_input_names = ["audio_values", "audio_attention_mask"]
|
90 |
|
91 |
def __init__(
|
92 |
self,
|
93 |
compression_rate: int = DEFAULT_COMPRESSION_RATE,
|
94 |
qformer_rate: int = DEFAULT_QFORMER_RATE,
|
95 |
feat_stride: int = DEFAULT_FEAT_STRIDE,
|
96 |
-
sampling_rate: int = DEFAULT_SAMPLING_RATE,
|
97 |
n_fft: int = DEFAULT_N_FFT,
|
98 |
-
win_length: Optional[int] = None,
|
99 |
-
hop_length: Optional[int] = None,
|
100 |
n_mels: int = DEFAULT_N_MELS,
|
101 |
f_min: float = 0.0,
|
102 |
f_max: Optional[float] = None,
|
103 |
-
padding_value: float = 0.0,
|
104 |
**kwargs
|
105 |
):
|
106 |
super().__init__(feature_size=n_mels, sampling_rate=sampling_rate, padding_value=padding_value, **kwargs)
|
@@ -128,16 +128,16 @@ class Gemma3AudioFeatureExtractor(SequenceFeatureExtractor):
|
|
128 |
def __call__(
|
129 |
self,
|
130 |
audios: Union[AudioInput, List[AudioInput]],
|
131 |
-
sampling_rate: Optional[int] = None,
|
132 |
return_tensors: Union[TensorType, str, None] = TensorType.PYTORCH
|
133 |
) -> BatchFeature:
|
134 |
-
|
135 |
if not isinstance(audios, list):
|
136 |
audios = [audios]
|
137 |
|
138 |
processed_mel_spectrograms: List[torch.Tensor] = []
|
139 |
actual_mel_lengths: List[int] = []
|
140 |
-
downstream_sizes_for_token_calc: List[torch.Tensor] = []
|
141 |
downstream_frames_scaled_for_token_calc: List[int] = []
|
142 |
|
143 |
for audio_input_item in audios:
|
@@ -161,11 +161,11 @@ class Gemma3AudioFeatureExtractor(SequenceFeatureExtractor):
|
|
161 |
)
|
162 |
|
163 |
processed_wav = self._preprocess_audio(current_wav_array, source_sr)
|
164 |
-
mel_spectrogram = self._compute_log_mel_spectrogram(processed_wav)
|
165 |
-
|
166 |
feature_tensor = torch.from_numpy(mel_spectrogram)
|
167 |
processed_mel_spectrograms.append(feature_tensor)
|
168 |
-
actual_mel_lengths.append(feature_tensor.shape[0])
|
169 |
|
170 |
downstream_sizes_for_token_calc.append(torch.tensor(self._calculate_embed_length(feature_tensor.shape[0])))
|
171 |
downstream_frames_scaled_for_token_calc.append(feature_tensor.shape[0] * self.feat_stride)
|
@@ -173,16 +173,17 @@ class Gemma3AudioFeatureExtractor(SequenceFeatureExtractor):
|
|
173 |
audio_values = pad_sequence(processed_mel_spectrograms, batch_first=True, padding_value=self.padding_value)
|
174 |
max_mel_len = audio_values.shape[1]
|
175 |
lengths_tensor = torch.tensor(actual_mel_lengths, dtype=torch.long)
|
176 |
-
audio_attention_mask = torch.arange(max_mel_len).unsqueeze(0).expand(len(audios),
|
177 |
-
|
|
|
178 |
output_data = {
|
179 |
"audio_values": audio_values,
|
180 |
"audio_attention_mask": audio_attention_mask
|
181 |
}
|
182 |
-
|
183 |
if downstream_sizes_for_token_calc:
|
184 |
-
|
185 |
-
|
186 |
return BatchFeature(data=output_data, tensor_type=return_tensors)
|
187 |
|
188 |
def _preprocess_audio(self, wav: np.ndarray, source_sr: int) -> np.ndarray:
|
@@ -190,22 +191,22 @@ class Gemma3AudioFeatureExtractor(SequenceFeatureExtractor):
|
|
190 |
if np.issubdtype(wav.dtype, np.integer):
|
191 |
max_val = np.iinfo(wav.dtype).max
|
192 |
wav = wav.astype(np.float32) / max_val
|
193 |
-
else:
|
194 |
wav = wav.astype(np.float32)
|
195 |
-
|
196 |
if wav.ndim > 1:
|
197 |
wav = wav.mean(axis=0)
|
198 |
-
|
199 |
if source_sr != self.sampling_rate:
|
200 |
gcd = math.gcd(self.sampling_rate, source_sr)
|
201 |
up_factor = self.sampling_rate // gcd
|
202 |
down_factor = source_sr // gcd
|
203 |
-
if up_factor != down_factor:
|
204 |
-
|
205 |
-
|
206 |
-
|
207 |
norm_factor = np.abs(wav).max()
|
208 |
-
if norm_factor > 1e-9:
|
209 |
wav = wav / norm_factor
|
210 |
return wav
|
211 |
|
@@ -216,7 +217,8 @@ class Gemma3AudioFeatureExtractor(SequenceFeatureExtractor):
|
|
216 |
|
217 |
num_frames = 1 + (len(wav) - self.win_length) // self.hop_length
|
218 |
if num_frames <= 0:
|
219 |
-
logger.warning(
|
|
|
220 |
return np.zeros((0, self.n_mels), dtype=np.float32)
|
221 |
|
222 |
frames = np.lib.stride_tricks.as_strided(
|
@@ -225,21 +227,22 @@ class Gemma3AudioFeatureExtractor(SequenceFeatureExtractor):
|
|
225 |
strides=(wav.strides[0] * self.hop_length, wav.strides[0]),
|
226 |
writeable=False
|
227 |
)
|
228 |
-
|
229 |
windowed_frames = frames * self.window
|
230 |
stft_matrix = np.fft.rfft(windowed_frames, n=self.n_fft, axis=-1)
|
231 |
-
powers = np.abs(stft_matrix)**2
|
232 |
mel_spectrogram = np.dot(powers, self.mel_filterbank)
|
233 |
mel_spectrogram = np.clip(mel_spectrogram, LOG_MEL_CLIP_EPSILON, None)
|
234 |
log_mel_spectrogram = np.log(mel_spectrogram)
|
235 |
-
|
236 |
return log_mel_spectrogram.astype(np.float32)
|
237 |
|
238 |
def _calculate_embed_length(self, frame_count: int) -> int:
|
239 |
compressed = math.ceil(frame_count / self.compression_rate)
|
240 |
return math.ceil(compressed / self.qformer_rate)
|
241 |
|
242 |
-
|
|
|
243 |
images_kwargs: Dict[str, Any]
|
244 |
audio_kwargs: Dict[str, Any]
|
245 |
text_kwargs: Dict[str, Any]
|
@@ -249,24 +252,25 @@ class Gemma3DummyProcessorKwargs(ProcessingKwargs, total=False): # Dummy for tes
|
|
249 |
"audio_kwargs": {}
|
250 |
}
|
251 |
|
|
|
252 |
class Gemma3OmniProcessor(ProcessorMixin):
|
253 |
attributes = ["image_processor", "audio_processor", "tokenizer"]
|
254 |
# Define class attributes for ProcessorMixin to find/use them
|
255 |
image_processor_class = "AutoImageProcessor" # Or the specific class string if not auto
|
256 |
-
audio_processor_class = Gemma3AudioFeatureExtractor
|
257 |
-
tokenizer_class = "AutoTokenizer"
|
258 |
|
259 |
# valid_kwargs was in user's code, its role depends on ProcessorMixin internal usage
|
260 |
-
valid_kwargs = ["chat_template", "image_seq_length"]
|
261 |
|
262 |
def __init__(
|
263 |
self,
|
264 |
-
tokenizer,
|
265 |
audio_processor: Optional[Union[Gemma3AudioFeatureExtractor, Dict]] = None,
|
266 |
-
image_processor
|
267 |
chat_template=None,
|
268 |
image_seq_length: int = 256,
|
269 |
-
audio_prompt_compression_rate: int = 8,
|
270 |
audio_prompt_qformer_rate: int = 1,
|
271 |
audio_prompt_feat_stride: int = 1,
|
272 |
audio_placeholder_token: str = "<|audio_placeholder|>",
|
@@ -279,48 +283,50 @@ class Gemma3OmniProcessor(ProcessorMixin):
|
|
279 |
audio_processor = Gemma3AudioFeatureExtractor()
|
280 |
elif isinstance(audio_processor, Dict):
|
281 |
audio_processor = Gemma3AudioFeatureExtractor(**audio_processor)
|
282 |
-
elif not isinstance(audio_processor, Gemma3AudioFeatureExtractor):
|
283 |
-
raise TypeError(
|
|
|
284 |
|
285 |
# Handle image_processor similarly if it can be None or a dict
|
286 |
if image_processor is None and self.image_processor_class:
|
287 |
-
|
288 |
if isinstance(self.image_processor_class, str) and self.image_processor_class == "AutoImageProcessor":
|
289 |
-
logger.info(
|
|
|
290 |
# image_processor = AutoImageProcessor.from_pretrained("openai/clip-vit-base-patch32") # Example default
|
291 |
# else if self.image_processor_class is an actual class, instantiate it.
|
292 |
elif isinstance(image_processor, Dict):
|
293 |
# image_processor = AutoImageProcessor.from_config(config_class(**image_processor)) # Example
|
294 |
-
pass
|
295 |
|
296 |
# Ensure tokenizer is an instantiated object
|
297 |
-
if isinstance(tokenizer, str):
|
298 |
logger.info(f"Loading tokenizer from {tokenizer}")
|
299 |
# tokenizer = AutoTokenizer.from_pretrained(tokenizer) # This is how it's usually done
|
300 |
elif tokenizer is None:
|
301 |
-
|
302 |
-
|
303 |
|
304 |
super().__init__(
|
305 |
image_processor=image_processor,
|
306 |
audio_processor=audio_processor,
|
307 |
tokenizer=tokenizer,
|
308 |
chat_template=chat_template,
|
309 |
-
**kwargs
|
310 |
)
|
311 |
-
|
312 |
self.image_seq_length = image_seq_length
|
313 |
-
self.image_token_id = getattr(self.tokenizer, "image_token_id",
|
314 |
-
|
|
|
315 |
self.image_token = getattr(self.tokenizer, "image_token", "<|image|>")
|
316 |
-
self.eoi_token = getattr(self.tokenizer, "eoi_token", "")
|
317 |
|
318 |
self.audio_placeholder_token = audio_placeholder_token
|
319 |
self.audio_soft_token_str = audio_soft_token_str
|
320 |
-
|
321 |
self.audio_soft_token_id = self.tokenizer.convert_tokens_to_ids(self.audio_soft_token_str)
|
322 |
-
if self.audio_soft_token_id == self.tokenizer.unk_token_id:
|
323 |
-
|
324 |
f"The audio soft token string '{self.audio_soft_token_str}' maps to UNK token (ID: {self.audio_soft_token_id}). "
|
325 |
"Ensure it is added to the tokenizer's vocabulary as a special token."
|
326 |
)
|
@@ -331,7 +337,6 @@ class Gemma3OmniProcessor(ProcessorMixin):
|
|
331 |
self.audio_prompt_qformer_rate = audio_prompt_qformer_rate
|
332 |
self.audio_prompt_feat_stride = audio_prompt_feat_stride
|
333 |
|
334 |
-
|
335 |
def _merge_kwargs(self, KwargsClassWithDefaults, tokenizer_init_kwargs, **kwargs_passed_to_call):
|
336 |
final_kwargs = {}
|
337 |
# Initialize with _defaults from the Kwargs class
|
@@ -342,24 +347,24 @@ class Gemma3OmniProcessor(ProcessorMixin):
|
|
342 |
|
343 |
# Override with tokenizer's init_kwargs if they exist for a given key
|
344 |
for modality_key, modality_dict in final_kwargs.items():
|
345 |
-
for key in list(modality_dict.keys()):
|
346 |
if key in tokenizer_init_kwargs:
|
347 |
modality_dict[key] = tokenizer_init_kwargs[key]
|
348 |
-
|
349 |
# Override with kwargs passed directly to __call__
|
350 |
for modality_key_from_call, modality_dict_from_call in kwargs_passed_to_call.items():
|
351 |
if modality_key_from_call in final_kwargs and isinstance(modality_dict_from_call, dict):
|
352 |
final_kwargs[modality_key_from_call].update(modality_dict_from_call)
|
353 |
# If a new modality_kwargs (e.g., "video_kwargs") is passed, add it
|
354 |
elif modality_key_from_call not in final_kwargs and isinstance(modality_dict_from_call, dict):
|
355 |
-
|
356 |
|
357 |
# Specific handling for text_kwargs
|
358 |
if "text_kwargs" not in final_kwargs:
|
359 |
-
|
360 |
final_kwargs["text_kwargs"]["truncation"] = final_kwargs["text_kwargs"].get("truncation", False)
|
361 |
final_kwargs["text_kwargs"]["max_length"] = final_kwargs["text_kwargs"].get("max_length", DEFAULT_MAX_LENGTH)
|
362 |
-
|
363 |
return final_kwargs
|
364 |
|
365 |
def _compute_audio_prompt_token_count(self, actual_mel_frames_count: int) -> int:
|
@@ -371,13 +376,13 @@ class Gemma3OmniProcessor(ProcessorMixin):
|
|
371 |
def __call__(
|
372 |
self,
|
373 |
text: Union[str, List[str]] = None,
|
374 |
-
images: Optional[Any] = None,
|
375 |
audios: Optional[Union[AudioInput, List[AudioInput]]] = None,
|
376 |
-
sampling_rate: Optional[int] = None,
|
377 |
-
return_tensors: Optional[Union[str, TensorType]] = None,
|
378 |
-
**kwargs: Any
|
379 |
) -> BatchFeature:
|
380 |
-
|
381 |
if text is None and images is None and audios is None:
|
382 |
raise ValueError("Provide at least one of `text`, `images`, or `audios`.")
|
383 |
|
@@ -385,27 +390,27 @@ class Gemma3OmniProcessor(ProcessorMixin):
|
|
385 |
# Priority: 1. Explicit return_tensors, 2. from text_kwargs in **kwargs, 3. Default (PT)
|
386 |
final_rt = return_tensors
|
387 |
merged_call_kwargs = self._merge_kwargs(
|
388 |
-
Gemma3DummyProcessorKwargs,
|
389 |
self.tokenizer.init_kwargs if hasattr(self.tokenizer, 'init_kwargs') else {},
|
390 |
-
**kwargs
|
391 |
)
|
392 |
-
|
393 |
-
if final_rt is None:
|
394 |
final_rt = merged_call_kwargs.get("text_kwargs", {}).pop("return_tensors", TensorType.PYTORCH)
|
395 |
-
else:
|
396 |
merged_call_kwargs.get("text_kwargs", {}).pop("return_tensors", None)
|
397 |
|
398 |
-
|
399 |
if text is None:
|
400 |
num_samples = 0
|
401 |
if images is not None:
|
402 |
-
_images_list = images if isinstance(images, list) and (
|
|
|
403 |
num_samples = len(_images_list)
|
404 |
elif audios is not None:
|
405 |
_audios_list = audios if isinstance(audios, list) else [audios]
|
406 |
num_samples = len(_audios_list)
|
407 |
text = [""] * num_samples if num_samples > 0 else [""]
|
408 |
-
|
409 |
if isinstance(text, str):
|
410 |
text = [text]
|
411 |
if not (isinstance(text, list) and all(isinstance(t, str) for t in text)):
|
@@ -419,17 +424,16 @@ class Gemma3OmniProcessor(ProcessorMixin):
|
|
419 |
# text = self._handle_image_text_replacement(text, images, image_features_dict)
|
420 |
pass
|
421 |
|
422 |
-
|
423 |
audio_features_dict = {}
|
424 |
if audios is not None and self.audio_processor is not None:
|
425 |
logger.info("Processing audio...")
|
426 |
audio_call_kwargs = merged_call_kwargs.get("audio_kwargs", {})
|
427 |
-
if sampling_rate:
|
428 |
-
|
429 |
-
|
430 |
# audio_processor.__call__ returns BatchFeature, we need its .data attribute
|
431 |
audio_features_batch_feature = self.audio_processor(audios=audios, return_tensors=None, **audio_call_kwargs)
|
432 |
-
audio_features_dict = audio_features_batch_feature.data
|
433 |
|
434 |
new_text_with_audio = []
|
435 |
# audio_attention_mask shape is (B, Max_T_mel)
|
@@ -438,42 +442,42 @@ class Gemma3OmniProcessor(ProcessorMixin):
|
|
438 |
for i, prompt in enumerate(text):
|
439 |
num_soft_tokens = self._compute_audio_prompt_token_count(audio_sample_mel_lengths[i])
|
440 |
audio_token_sequence_str = self.audio_soft_token_str * num_soft_tokens
|
441 |
-
|
442 |
if self.audio_placeholder_token in prompt:
|
443 |
prompt = prompt.replace(self.audio_placeholder_token, audio_token_sequence_str, 1)
|
444 |
-
else:
|
445 |
-
prompt += audio_token_sequence_str
|
446 |
new_text_with_audio.append(prompt)
|
447 |
text = new_text_with_audio
|
448 |
-
|
449 |
logger.info("Tokenizing text...")
|
450 |
text_call_kwargs = merged_call_kwargs.get("text_kwargs", {})
|
451 |
text_features_dict = self.tokenizer(text, return_tensors=None, **text_call_kwargs)
|
452 |
|
453 |
input_ids_list = text_features_dict["input_ids"]
|
454 |
if not isinstance(input_ids_list, list) or not (input_ids_list and isinstance(input_ids_list[0], list)):
|
455 |
-
|
456 |
-
|
457 |
-
|
458 |
-
|
459 |
|
460 |
token_type_ids_list = []
|
461 |
for ids_sample in input_ids_list:
|
462 |
-
types = [0] * len(ids_sample)
|
463 |
for j, token_id in enumerate(ids_sample):
|
464 |
if self.image_token_id is not None and token_id == self.image_token_id:
|
465 |
-
types[j] = 1
|
466 |
-
elif token_id == self.audio_soft_token_id:
|
467 |
-
types[j] = 2
|
468 |
token_type_ids_list.append(types)
|
469 |
text_features_dict["token_type_ids"] = token_type_ids_list
|
470 |
-
|
471 |
combined_features = {**text_features_dict}
|
472 |
-
if image_features_dict:
|
473 |
combined_features.update(image_features_dict)
|
474 |
-
if audio_features_dict:
|
475 |
combined_features.update(audio_features_dict)
|
476 |
-
|
477 |
return BatchFeature(data=combined_features, tensor_type=final_rt)
|
478 |
|
479 |
def batch_decode(self, *args, **kwargs):
|
@@ -489,6 +493,6 @@ class Gemma3OmniProcessor(ProcessorMixin):
|
|
489 |
input_names.update(self.image_processor.model_input_names)
|
490 |
if self.audio_processor is not None:
|
491 |
# From Gemma3AudioFeatureExtractor's output_data keys
|
492 |
-
input_names.update(["audio_values", "audio_attention_mask"])
|
493 |
# "audio_token_calc_sizes" is internal to processor, not model.
|
494 |
return list(input_names)
|
|
|
1 |
import re
|
2 |
+
from typing import List, Optional, Union, Dict, Any, Tuple
|
3 |
|
4 |
import math
|
5 |
import numpy as np
|
|
|
15 |
|
16 |
from transformers.feature_extraction_sequence_utils import SequenceFeatureExtractor
|
17 |
from transformers.feature_extraction_utils import BatchFeature
|
18 |
+
from transformers.processing_utils import ProcessorMixin, ProcessingKwargs
|
19 |
from transformers.utils import TensorType, to_py_obj, logging
|
20 |
# For AutoImageProcessor, AutoTokenizer if needed for default loading
|
21 |
from transformers import AutoImageProcessor, AutoTokenizer
|
22 |
|
|
|
23 |
# Constants (as defined before)
|
24 |
DEFAULT_SAMPLING_RATE = 16000
|
25 |
DEFAULT_N_FFT = 512
|
26 |
+
DEFAULT_WIN_LENGTH = 400 # Will be n_fft if None in __init__
|
27 |
+
DEFAULT_HOP_LENGTH = 160 # Will be win_length // 4 if None in __init__
|
28 |
DEFAULT_N_MELS = 80
|
29 |
DEFAULT_COMPRESSION_RATE = 4
|
30 |
DEFAULT_QFORMER_RATE = 2
|
|
|
47 |
if fmin >= fmax:
|
48 |
raise ValueError(f"fmin ({fmin}) must be smaller than fmax ({fmax}).")
|
49 |
|
50 |
+
def hz_to_mel(f: float) -> float: # Using HTK formula (as in librosa default)
|
51 |
return 2595.0 * math.log10(1 + f / 700.0)
|
52 |
|
53 |
def mel_to_hz(mel: float) -> float:
|
54 |
+
return 700.0 * (10 ** (mel / 2595.0) - 1)
|
55 |
|
56 |
mel_points = np.linspace(hz_to_mel(fmin), hz_to_mel(fmax), n_mels + 2)
|
57 |
freq_points = mel_to_hz(mel_points)
|
58 |
+
|
59 |
freq_points = np.clip(freq_points, 0, sampling_rate / 2.0)
|
60 |
bins = np.floor((n_fft / 2.0) * freq_points / (sampling_rate / 2.0)).astype(int)
|
61 |
bins = np.clip(bins, 0, n_fft // 2)
|
|
|
63 |
filterbank = np.zeros((n_mels, n_fft // 2 + 1), dtype=np.float32)
|
64 |
for m in range(n_mels):
|
65 |
left, center, right = bins[m], bins[m + 1], bins[m + 2]
|
66 |
+
|
67 |
# Simplified triangle creation logic (more robust versions exist in libraries like librosa)
|
68 |
if center > left:
|
69 |
+
filterbank[m, left:center + 1] = (np.arange(left, center + 1) - left) / (center - left)
|
70 |
if right > center:
|
71 |
+
filterbank[m, center:right + 1] = (right - np.arange(center, right + 1)) / (right - center)
|
72 |
# Ensure peak is 1 if multiple points coincide at center (can happen with narrow filters/low resolution)
|
73 |
+
if left <= center <= right and filterbank[m, center] < 1.0 and (
|
74 |
+
center > left or center < right): # check if it's a valid point for a peak
|
75 |
# if filterbank[m,center] is not properly set to 1 by slopes (e.g. left==center or right==center)
|
76 |
+
filterbank[m, center] = 1.0
|
77 |
+
if left == center and right > center: # only falling slope
|
78 |
+
# Ensure it doesn't double-dip if already set
|
79 |
+
pass
|
80 |
+
elif right == center and left < center: # only rising slope
|
81 |
+
pass
|
|
|
82 |
|
83 |
return filterbank
|
84 |
|
85 |
+
|
86 |
# Gemma3AudioFeatureExtractor class (assuming it's correctly defined from previous response)
|
87 |
# ... (Gemma3AudioFeatureExtractor class from the previous corrected response) ...
|
88 |
class Gemma3AudioFeatureExtractor(SequenceFeatureExtractor):
|
89 |
+
model_input_names = ["audio_values", "audio_attention_mask"]
|
90 |
|
91 |
def __init__(
|
92 |
self,
|
93 |
compression_rate: int = DEFAULT_COMPRESSION_RATE,
|
94 |
qformer_rate: int = DEFAULT_QFORMER_RATE,
|
95 |
feat_stride: int = DEFAULT_FEAT_STRIDE,
|
96 |
+
sampling_rate: int = DEFAULT_SAMPLING_RATE,
|
97 |
n_fft: int = DEFAULT_N_FFT,
|
98 |
+
win_length: Optional[int] = None,
|
99 |
+
hop_length: Optional[int] = None,
|
100 |
n_mels: int = DEFAULT_N_MELS,
|
101 |
f_min: float = 0.0,
|
102 |
f_max: Optional[float] = None,
|
103 |
+
padding_value: float = 0.0,
|
104 |
**kwargs
|
105 |
):
|
106 |
super().__init__(feature_size=n_mels, sampling_rate=sampling_rate, padding_value=padding_value, **kwargs)
|
|
|
128 |
def __call__(
|
129 |
self,
|
130 |
audios: Union[AudioInput, List[AudioInput]],
|
131 |
+
sampling_rate: Optional[int] = None,
|
132 |
return_tensors: Union[TensorType, str, None] = TensorType.PYTORCH
|
133 |
) -> BatchFeature:
|
134 |
+
|
135 |
if not isinstance(audios, list):
|
136 |
audios = [audios]
|
137 |
|
138 |
processed_mel_spectrograms: List[torch.Tensor] = []
|
139 |
actual_mel_lengths: List[int] = []
|
140 |
+
downstream_sizes_for_token_calc: List[torch.Tensor] = []
|
141 |
downstream_frames_scaled_for_token_calc: List[int] = []
|
142 |
|
143 |
for audio_input_item in audios:
|
|
|
161 |
)
|
162 |
|
163 |
processed_wav = self._preprocess_audio(current_wav_array, source_sr)
|
164 |
+
mel_spectrogram = self._compute_log_mel_spectrogram(processed_wav)
|
165 |
+
|
166 |
feature_tensor = torch.from_numpy(mel_spectrogram)
|
167 |
processed_mel_spectrograms.append(feature_tensor)
|
168 |
+
actual_mel_lengths.append(feature_tensor.shape[0])
|
169 |
|
170 |
downstream_sizes_for_token_calc.append(torch.tensor(self._calculate_embed_length(feature_tensor.shape[0])))
|
171 |
downstream_frames_scaled_for_token_calc.append(feature_tensor.shape[0] * self.feat_stride)
|
|
|
173 |
audio_values = pad_sequence(processed_mel_spectrograms, batch_first=True, padding_value=self.padding_value)
|
174 |
max_mel_len = audio_values.shape[1]
|
175 |
lengths_tensor = torch.tensor(actual_mel_lengths, dtype=torch.long)
|
176 |
+
audio_attention_mask = torch.arange(max_mel_len).unsqueeze(0).expand(len(audios),
|
177 |
+
-1) < lengths_tensor.unsqueeze(1)
|
178 |
+
|
179 |
output_data = {
|
180 |
"audio_values": audio_values,
|
181 |
"audio_attention_mask": audio_attention_mask
|
182 |
}
|
183 |
+
|
184 |
if downstream_sizes_for_token_calc:
|
185 |
+
output_data["audio_token_calc_sizes"] = torch.stack(downstream_sizes_for_token_calc)
|
186 |
+
|
187 |
return BatchFeature(data=output_data, tensor_type=return_tensors)
|
188 |
|
189 |
def _preprocess_audio(self, wav: np.ndarray, source_sr: int) -> np.ndarray:
|
|
|
191 |
if np.issubdtype(wav.dtype, np.integer):
|
192 |
max_val = np.iinfo(wav.dtype).max
|
193 |
wav = wav.astype(np.float32) / max_val
|
194 |
+
else:
|
195 |
wav = wav.astype(np.float32)
|
196 |
+
|
197 |
if wav.ndim > 1:
|
198 |
wav = wav.mean(axis=0)
|
199 |
+
|
200 |
if source_sr != self.sampling_rate:
|
201 |
gcd = math.gcd(self.sampling_rate, source_sr)
|
202 |
up_factor = self.sampling_rate // gcd
|
203 |
down_factor = source_sr // gcd
|
204 |
+
if up_factor != down_factor:
|
205 |
+
logger.info(f"Resampling audio from {source_sr} Hz to {self.sampling_rate} Hz.")
|
206 |
+
wav = scipy.signal.resample_poly(wav, up=up_factor, down=down_factor)
|
207 |
+
|
208 |
norm_factor = np.abs(wav).max()
|
209 |
+
if norm_factor > 1e-9:
|
210 |
wav = wav / norm_factor
|
211 |
return wav
|
212 |
|
|
|
217 |
|
218 |
num_frames = 1 + (len(wav) - self.win_length) // self.hop_length
|
219 |
if num_frames <= 0:
|
220 |
+
logger.warning(
|
221 |
+
f"Audio of length {len(wav)} is too short to produce frames with win_length {self.win_length} and hop_length {self.hop_length}. Returning empty mel spectrogram.")
|
222 |
return np.zeros((0, self.n_mels), dtype=np.float32)
|
223 |
|
224 |
frames = np.lib.stride_tricks.as_strided(
|
|
|
227 |
strides=(wav.strides[0] * self.hop_length, wav.strides[0]),
|
228 |
writeable=False
|
229 |
)
|
230 |
+
|
231 |
windowed_frames = frames * self.window
|
232 |
stft_matrix = np.fft.rfft(windowed_frames, n=self.n_fft, axis=-1)
|
233 |
+
powers = np.abs(stft_matrix) ** 2
|
234 |
mel_spectrogram = np.dot(powers, self.mel_filterbank)
|
235 |
mel_spectrogram = np.clip(mel_spectrogram, LOG_MEL_CLIP_EPSILON, None)
|
236 |
log_mel_spectrogram = np.log(mel_spectrogram)
|
237 |
+
|
238 |
return log_mel_spectrogram.astype(np.float32)
|
239 |
|
240 |
def _calculate_embed_length(self, frame_count: int) -> int:
|
241 |
compressed = math.ceil(frame_count / self.compression_rate)
|
242 |
return math.ceil(compressed / self.qformer_rate)
|
243 |
|
244 |
+
|
245 |
+
class Gemma3DummyProcessorKwargs(ProcessingKwargs, total=False): # Dummy for testing structure
|
246 |
images_kwargs: Dict[str, Any]
|
247 |
audio_kwargs: Dict[str, Any]
|
248 |
text_kwargs: Dict[str, Any]
|
|
|
252 |
"audio_kwargs": {}
|
253 |
}
|
254 |
|
255 |
+
|
256 |
class Gemma3OmniProcessor(ProcessorMixin):
|
257 |
attributes = ["image_processor", "audio_processor", "tokenizer"]
|
258 |
# Define class attributes for ProcessorMixin to find/use them
|
259 |
image_processor_class = "AutoImageProcessor" # Or the specific class string if not auto
|
260 |
+
audio_processor_class = Gemma3AudioFeatureExtractor # Correctly points to your custom class
|
261 |
+
tokenizer_class = "AutoTokenizer" # Or the specific class string
|
262 |
|
263 |
# valid_kwargs was in user's code, its role depends on ProcessorMixin internal usage
|
264 |
+
valid_kwargs = ["chat_template", "image_seq_length"]
|
265 |
|
266 |
def __init__(
|
267 |
self,
|
268 |
+
tokenizer,
|
269 |
audio_processor: Optional[Union[Gemma3AudioFeatureExtractor, Dict]] = None,
|
270 |
+
image_processor=None,
|
271 |
chat_template=None,
|
272 |
image_seq_length: int = 256,
|
273 |
+
audio_prompt_compression_rate: int = 8,
|
274 |
audio_prompt_qformer_rate: int = 1,
|
275 |
audio_prompt_feat_stride: int = 1,
|
276 |
audio_placeholder_token: str = "<|audio_placeholder|>",
|
|
|
283 |
audio_processor = Gemma3AudioFeatureExtractor()
|
284 |
elif isinstance(audio_processor, Dict):
|
285 |
audio_processor = Gemma3AudioFeatureExtractor(**audio_processor)
|
286 |
+
elif not isinstance(audio_processor, Gemma3AudioFeatureExtractor): # Check type if instance is passed
|
287 |
+
raise TypeError(
|
288 |
+
f"audio_processor must be an instance of Gemma3AudioFeatureExtractor or a config dict, got {type(audio_processor)}")
|
289 |
|
290 |
# Handle image_processor similarly if it can be None or a dict
|
291 |
if image_processor is None and self.image_processor_class:
|
292 |
+
# This is a basic way; from_pretrained usually handles complex loading
|
293 |
if isinstance(self.image_processor_class, str) and self.image_processor_class == "AutoImageProcessor":
|
294 |
+
logger.info(
|
295 |
+
f"Attempting to load a default {self.image_processor_class}. This might require a default model name or fail.")
|
296 |
# image_processor = AutoImageProcessor.from_pretrained("openai/clip-vit-base-patch32") # Example default
|
297 |
# else if self.image_processor_class is an actual class, instantiate it.
|
298 |
elif isinstance(image_processor, Dict):
|
299 |
# image_processor = AutoImageProcessor.from_config(config_class(**image_processor)) # Example
|
300 |
+
pass # Actual instantiation from dict would be more complex
|
301 |
|
302 |
# Ensure tokenizer is an instantiated object
|
303 |
+
if isinstance(tokenizer, str): # If tokenizer is a string (model name/path)
|
304 |
logger.info(f"Loading tokenizer from {tokenizer}")
|
305 |
# tokenizer = AutoTokenizer.from_pretrained(tokenizer) # This is how it's usually done
|
306 |
elif tokenizer is None:
|
307 |
+
raise ValueError("A tokenizer instance or identifier must be provided.")
|
|
|
308 |
|
309 |
super().__init__(
|
310 |
image_processor=image_processor,
|
311 |
audio_processor=audio_processor,
|
312 |
tokenizer=tokenizer,
|
313 |
chat_template=chat_template,
|
314 |
+
**kwargs # Pass other kwargs to super
|
315 |
)
|
316 |
+
|
317 |
self.image_seq_length = image_seq_length
|
318 |
+
self.image_token_id = getattr(self.tokenizer, "image_token_id",
|
319 |
+
self.tokenizer.unk_token_id if hasattr(self.tokenizer, "unk_token_id") else None)
|
320 |
+
self.boi_token = getattr(self.tokenizer, "boi_token", "<|image|>")
|
321 |
self.image_token = getattr(self.tokenizer, "image_token", "<|image|>")
|
322 |
+
self.eoi_token = getattr(self.tokenizer, "eoi_token", "")
|
323 |
|
324 |
self.audio_placeholder_token = audio_placeholder_token
|
325 |
self.audio_soft_token_str = audio_soft_token_str
|
326 |
+
|
327 |
self.audio_soft_token_id = self.tokenizer.convert_tokens_to_ids(self.audio_soft_token_str)
|
328 |
+
if self.audio_soft_token_id == self.tokenizer.unk_token_id: # Check if UNK
|
329 |
+
logger.warning(
|
330 |
f"The audio soft token string '{self.audio_soft_token_str}' maps to UNK token (ID: {self.audio_soft_token_id}). "
|
331 |
"Ensure it is added to the tokenizer's vocabulary as a special token."
|
332 |
)
|
|
|
337 |
self.audio_prompt_qformer_rate = audio_prompt_qformer_rate
|
338 |
self.audio_prompt_feat_stride = audio_prompt_feat_stride
|
339 |
|
|
|
340 |
def _merge_kwargs(self, KwargsClassWithDefaults, tokenizer_init_kwargs, **kwargs_passed_to_call):
|
341 |
final_kwargs = {}
|
342 |
# Initialize with _defaults from the Kwargs class
|
|
|
347 |
|
348 |
# Override with tokenizer's init_kwargs if they exist for a given key
|
349 |
for modality_key, modality_dict in final_kwargs.items():
|
350 |
+
for key in list(modality_dict.keys()):
|
351 |
if key in tokenizer_init_kwargs:
|
352 |
modality_dict[key] = tokenizer_init_kwargs[key]
|
353 |
+
|
354 |
# Override with kwargs passed directly to __call__
|
355 |
for modality_key_from_call, modality_dict_from_call in kwargs_passed_to_call.items():
|
356 |
if modality_key_from_call in final_kwargs and isinstance(modality_dict_from_call, dict):
|
357 |
final_kwargs[modality_key_from_call].update(modality_dict_from_call)
|
358 |
# If a new modality_kwargs (e.g., "video_kwargs") is passed, add it
|
359 |
elif modality_key_from_call not in final_kwargs and isinstance(modality_dict_from_call, dict):
|
360 |
+
final_kwargs[modality_key_from_call] = modality_dict_from_call.copy()
|
361 |
|
362 |
# Specific handling for text_kwargs
|
363 |
if "text_kwargs" not in final_kwargs:
|
364 |
+
final_kwargs["text_kwargs"] = {} # Ensure it exists
|
365 |
final_kwargs["text_kwargs"]["truncation"] = final_kwargs["text_kwargs"].get("truncation", False)
|
366 |
final_kwargs["text_kwargs"]["max_length"] = final_kwargs["text_kwargs"].get("max_length", DEFAULT_MAX_LENGTH)
|
367 |
+
|
368 |
return final_kwargs
|
369 |
|
370 |
def _compute_audio_prompt_token_count(self, actual_mel_frames_count: int) -> int:
|
|
|
376 |
def __call__(
|
377 |
self,
|
378 |
text: Union[str, List[str]] = None,
|
379 |
+
images: Optional[Any] = None,
|
380 |
audios: Optional[Union[AudioInput, List[AudioInput]]] = None,
|
381 |
+
sampling_rate: Optional[int] = None,
|
382 |
+
return_tensors: Optional[Union[str, TensorType]] = None,
|
383 |
+
**kwargs: Any
|
384 |
) -> BatchFeature:
|
385 |
+
|
386 |
if text is None and images is None and audios is None:
|
387 |
raise ValueError("Provide at least one of `text`, `images`, or `audios`.")
|
388 |
|
|
|
390 |
# Priority: 1. Explicit return_tensors, 2. from text_kwargs in **kwargs, 3. Default (PT)
|
391 |
final_rt = return_tensors
|
392 |
merged_call_kwargs = self._merge_kwargs(
|
393 |
+
Gemma3DummyProcessorKwargs, # Using dummy for _defaults structure
|
394 |
self.tokenizer.init_kwargs if hasattr(self.tokenizer, 'init_kwargs') else {},
|
395 |
+
**kwargs
|
396 |
)
|
397 |
+
|
398 |
+
if final_rt is None: # If not passed directly to __call__
|
399 |
final_rt = merged_call_kwargs.get("text_kwargs", {}).pop("return_tensors", TensorType.PYTORCH)
|
400 |
+
else: # If passed directly, remove from text_kwargs to avoid conflict
|
401 |
merged_call_kwargs.get("text_kwargs", {}).pop("return_tensors", None)
|
402 |
|
|
|
403 |
if text is None:
|
404 |
num_samples = 0
|
405 |
if images is not None:
|
406 |
+
_images_list = images if isinstance(images, list) and (
|
407 |
+
not images or not isinstance(images[0], (int, float))) else [images]
|
408 |
num_samples = len(_images_list)
|
409 |
elif audios is not None:
|
410 |
_audios_list = audios if isinstance(audios, list) else [audios]
|
411 |
num_samples = len(_audios_list)
|
412 |
text = [""] * num_samples if num_samples > 0 else [""]
|
413 |
+
|
414 |
if isinstance(text, str):
|
415 |
text = [text]
|
416 |
if not (isinstance(text, list) and all(isinstance(t, str) for t in text)):
|
|
|
424 |
# text = self._handle_image_text_replacement(text, images, image_features_dict)
|
425 |
pass
|
426 |
|
|
|
427 |
audio_features_dict = {}
|
428 |
if audios is not None and self.audio_processor is not None:
|
429 |
logger.info("Processing audio...")
|
430 |
audio_call_kwargs = merged_call_kwargs.get("audio_kwargs", {})
|
431 |
+
if sampling_rate:
|
432 |
+
audio_call_kwargs["sampling_rate"] = sampling_rate
|
433 |
+
|
434 |
# audio_processor.__call__ returns BatchFeature, we need its .data attribute
|
435 |
audio_features_batch_feature = self.audio_processor(audios=audios, return_tensors=None, **audio_call_kwargs)
|
436 |
+
audio_features_dict = audio_features_batch_feature.data # Get the dict
|
437 |
|
438 |
new_text_with_audio = []
|
439 |
# audio_attention_mask shape is (B, Max_T_mel)
|
|
|
442 |
for i, prompt in enumerate(text):
|
443 |
num_soft_tokens = self._compute_audio_prompt_token_count(audio_sample_mel_lengths[i])
|
444 |
audio_token_sequence_str = self.audio_soft_token_str * num_soft_tokens
|
445 |
+
|
446 |
if self.audio_placeholder_token in prompt:
|
447 |
prompt = prompt.replace(self.audio_placeholder_token, audio_token_sequence_str, 1)
|
448 |
+
else:
|
449 |
+
prompt += audio_token_sequence_str
|
450 |
new_text_with_audio.append(prompt)
|
451 |
text = new_text_with_audio
|
452 |
+
|
453 |
logger.info("Tokenizing text...")
|
454 |
text_call_kwargs = merged_call_kwargs.get("text_kwargs", {})
|
455 |
text_features_dict = self.tokenizer(text, return_tensors=None, **text_call_kwargs)
|
456 |
|
457 |
input_ids_list = text_features_dict["input_ids"]
|
458 |
if not isinstance(input_ids_list, list) or not (input_ids_list and isinstance(input_ids_list[0], list)):
|
459 |
+
if isinstance(input_ids_list, (torch.Tensor, np.ndarray)):
|
460 |
+
input_ids_list = to_py_obj(input_ids_list) # Convert tensor/np.array to list of lists
|
461 |
+
elif isinstance(input_ids_list, list) and (not input_ids_list or isinstance(input_ids_list[0], int)):
|
462 |
+
input_ids_list = [input_ids_list]
|
463 |
|
464 |
token_type_ids_list = []
|
465 |
for ids_sample in input_ids_list:
|
466 |
+
types = [0] * len(ids_sample)
|
467 |
for j, token_id in enumerate(ids_sample):
|
468 |
if self.image_token_id is not None and token_id == self.image_token_id:
|
469 |
+
types[j] = 1
|
470 |
+
elif token_id == self.audio_soft_token_id:
|
471 |
+
types[j] = 2
|
472 |
token_type_ids_list.append(types)
|
473 |
text_features_dict["token_type_ids"] = token_type_ids_list
|
474 |
+
|
475 |
combined_features = {**text_features_dict}
|
476 |
+
if image_features_dict:
|
477 |
combined_features.update(image_features_dict)
|
478 |
+
if audio_features_dict:
|
479 |
combined_features.update(audio_features_dict)
|
480 |
+
|
481 |
return BatchFeature(data=combined_features, tensor_type=final_rt)
|
482 |
|
483 |
def batch_decode(self, *args, **kwargs):
|
|
|
493 |
input_names.update(self.image_processor.model_input_names)
|
494 |
if self.audio_processor is not None:
|
495 |
# From Gemma3AudioFeatureExtractor's output_data keys
|
496 |
+
input_names.update(["audio_values", "audio_attention_mask"])
|
497 |
# "audio_token_calc_sizes" is internal to processor, not model.
|
498 |
return list(input_names)
|