Add model files

ailia-ai · ooe1123 · May 26, 2025 · May 28, 2025 · May 29, 2025 · May 30, 2025
commit 021f32af2698503741b3df4487f63f7f75dc6506
diff --git a/lipsync/hallo/demo.jpg b/lipsync/hallo/demo.jpg
diff --git a/lipsync/hallo/demo.wav b/lipsync/hallo/demo.wav
diff --git a/lipsync/hallo/face_analysis/models/face_landmarker_v2_with_blendshapes.task b/lipsync/hallo/face_analysis/models/face_landmarker_v2_with_blendshapes.task
diff --git a/lipsync/hallo/hallo.py b/lipsync/hallo/hallo.py
@@ -48,13 +48,16 @@
 MODEL_IMAGE_PROJ_PATH = "image_proj.onnx.prototxt"
 WEIGHT_DENOISE_PB_PATH = "denoising_unet_weights.pb"

+WEIGHT_FACE_ANALYSIS_DET_PATH = "./face_analysis/models/scrfd_10g_bnkps.onnx"
+WEIGHT_FACE_ANALYSIS_REG_PATH = "./face_analysis/models/glintr100.onnx"
+WEIGHT_WAV2VEC_PATH = "./wav2vec/wav2vec2-base-960h/model.safetensors"
+
 REMOTE_PATH = "https://storage.googleapis.com/ailia-models/hallo/"

 IMAGE_SIZE = 512
 SAMPLING_RATE = 16000
 IMAGE_PATH = "demo.jpg"
 WAV_PATH = "demo.wav"
-VIDEO_PATH = "demo.wav"
 SAVE_VIDEO_PATH = "output.mp4"

 # ======================
@@ -65,8 +68,8 @@
    "Hallo: Hierarchical Audio-Driven Visual Synthesis for Portrait Image Animation",
    IMAGE_PATH,
    SAVE_VIDEO_PATH,
-    input_ftype="audio",
 )
+parser.add_argument("--driving_audio", default=WAV_PATH, help="Input audio")
 parser.add_argument("--onnx", action="store_true", help="execute onnxruntime version.")
 args = update_parser(parser, check_input_type=False)

@@ -657,6 +660,7 @@ def progress_bar(self, iterable=None, total=None):

 def recognize_from_video(pipe: FaceAnimatePipeline):
    image_path = args.input[0]
+    driving_audio_path = args.driving_audio
    # prepare input data
    image = load_image(image_path)
    image = cv2.cvtColor(image, cv2.COLOR_BGRA2BGR)
@@ -758,6 +762,12 @@ def main():
    )
    check_and_download_file(WEIGHT_DENOISE_PB_PATH, REMOTE_PATH)

+    # for insightface
+    check_and_download_file(WEIGHT_FACE_ANALYSIS_DET_PATH, REMOTE_PATH)
+    check_and_download_file(WEIGHT_FACE_ANALYSIS_REG_PATH, REMOTE_PATH)
+    # wav2vec
+    check_and_download_file(WEIGHT_WAV2VEC_PATH, REMOTE_PATH)
+
    env_id = args.env_id

    # initialize

diff --git a/lipsync/hallo/wav2vec/wav2vec2-base-960h/README.md b/lipsync/hallo/wav2vec/wav2vec2-base-960h/README.md
@@ -0,0 +1,128 @@
+---
+language: en
+datasets:
+- librispeech_asr
+tags:
+- audio
+- automatic-speech-recognition
+- hf-asr-leaderboard
+license: apache-2.0
+widget:
+- example_title: Librispeech sample 1
+  src: https://cdn-media.huggingface.co/speech_samples/sample1.flac
+- example_title: Librispeech sample 2
+  src: https://cdn-media.huggingface.co/speech_samples/sample2.flac
+model-index:
+- name: wav2vec2-base-960h
+  results:
+  - task:
+      name: Automatic Speech Recognition
+      type: automatic-speech-recognition
+    dataset:
+      name: LibriSpeech (clean)
+      type: librispeech_asr
+      config: clean
+      split: test
+      args: 
+        language: en
+    metrics:
+    - name: Test WER
+      type: wer
+      value: 3.4
+  - task:
+      name: Automatic Speech Recognition
+      type: automatic-speech-recognition
+    dataset:
+      name: LibriSpeech (other)
+      type: librispeech_asr
+      config: other
+      split: test
+      args: 
+        language: en
+    metrics:
+    - name: Test WER
+      type: wer
+      value: 8.6
+---
+
+# Wav2Vec2-Base-960h
+
+[Facebook's Wav2Vec2](https://ai.facebook.com/blog/wav2vec-20-learning-the-structure-of-speech-from-raw-audio/)
+
+The base model pretrained and fine-tuned on 960 hours of Librispeech on 16kHz sampled speech audio. When using the model
+make sure that your speech input is also sampled at 16Khz.
+
+[Paper](https://arxiv.org/abs/2006.11477)
+
+Authors: Alexei Baevski, Henry Zhou, Abdelrahman Mohamed, Michael Auli
+
+**Abstract**
+
+We show for the first time that learning powerful representations from speech audio alone followed by fine-tuning on transcribed speech can outperform the best semi-supervised methods while being conceptually simpler. wav2vec 2.0 masks the speech input in the latent space and solves a contrastive task defined over a quantization of the latent representations which are jointly learned. Experiments using all labeled data of Librispeech achieve 1.8/3.3 WER on the clean/other test sets. When lowering the amount of labeled data to one hour, wav2vec 2.0 outperforms the previous state of the art on the 100 hour subset while using 100 times less labeled data. Using just ten minutes of labeled data and pre-training on 53k hours of unlabeled data still achieves 4.8/8.2 WER. This demonstrates the feasibility of speech recognition with limited amounts of labeled data.
+
+The original model can be found under https://github.com/pytorch/fairseq/tree/master/examples/wav2vec#wav2vec-20.
+
+
+# Usage
+
+To transcribe audio files the model can be used as a standalone acoustic model as follows:
+
+```python
+ from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
+ from datasets import load_dataset
+ import torch
+
+ # load model and tokenizer
+ processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-base-960h")
+ model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-base-960h")
+
+ # load dummy dataset and read soundfiles
+ ds = load_dataset("patrickvonplaten/librispeech_asr_dummy", "clean", split="validation")
+
+ # tokenize
+ input_values = processor(ds[0]["audio"]["array"], return_tensors="pt", padding="longest").input_values  # Batch size 1
+
+ # retrieve logits
+ logits = model(input_values).logits
+
+ # take argmax and decode
+ predicted_ids = torch.argmax(logits, dim=-1)
+ transcription = processor.batch_decode(predicted_ids)
+ ```
+
+ ## Evaluation
+
+ This code snippet shows how to evaluate **facebook/wav2vec2-base-960h** on LibriSpeech's "clean" and "other" test data.
+
+```python
+from datasets import load_dataset
+from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
+import torch
+from jiwer import wer
+
+
+librispeech_eval = load_dataset("librispeech_asr", "clean", split="test")
+
+model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-base-960h").to("cuda")
+processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-base-960h")
+
+def map_to_pred(batch):
+    input_values = processor(batch["audio"]["array"], return_tensors="pt", padding="longest").input_values
+    with torch.no_grad():
+        logits = model(input_values.to("cuda")).logits
+
+    predicted_ids = torch.argmax(logits, dim=-1)
+    transcription = processor.batch_decode(predicted_ids)
+    batch["transcription"] = transcription
+    return batch
+
+result = librispeech_eval.map(map_to_pred, batched=True, batch_size=1, remove_columns=["audio"])
+
+print("WER:", wer(result["text"], result["transcription"]))
+```
+
+*Result (WER)*:
+
+| "clean" | "other" |
+|---|---|
+| 3.4 | 8.6 |
diff --git a/lipsync/hallo/wav2vec/wav2vec2-base-960h/config.json b/lipsync/hallo/wav2vec/wav2vec2-base-960h/config.json
@@ -0,0 +1,77 @@
+{
+  "_name_or_path": "facebook/wav2vec2-base-960h",
+  "activation_dropout": 0.1,
+  "apply_spec_augment": true,
+  "architectures": [
+    "Wav2Vec2ForCTC"
+  ],
+  "attention_dropout": 0.1,
+  "bos_token_id": 1,
+  "codevector_dim": 256,
+  "contrastive_logits_temperature": 0.1,
+  "conv_bias": false,
+  "conv_dim": [
+    512,
+    512,
+    512,
+    512,
+    512,
+    512,
+    512
+  ],
+  "conv_kernel": [
+    10,
+    3,
+    3,
+    3,
+    3,
+    2,
+    2
+  ],
+  "conv_stride": [
+    5,
+    2,
+    2,
+    2,
+    2,
+    2,
+    2
+  ],
+  "ctc_loss_reduction": "sum",
+  "ctc_zero_infinity": false,
+  "diversity_loss_weight": 0.1,
+  "do_stable_layer_norm": false,
+  "eos_token_id": 2,
+  "feat_extract_activation": "gelu",
+  "feat_extract_dropout": 0.0,
+  "feat_extract_norm": "group",
+  "feat_proj_dropout": 0.1,
+  "feat_quantizer_dropout": 0.0,
+  "final_dropout": 0.1,
+  "gradient_checkpointing": false,
+  "hidden_act": "gelu",
+  "hidden_dropout": 0.1,
+  "hidden_dropout_prob": 0.1,
+  "hidden_size": 768,
+  "initializer_range": 0.02,
+  "intermediate_size": 3072,
+  "layer_norm_eps": 1e-05,
+  "layerdrop": 0.1,
+  "mask_feature_length": 10,
+  "mask_feature_prob": 0.0,
+  "mask_time_length": 10,
+  "mask_time_prob": 0.05,
+  "model_type": "wav2vec2",
+  "num_attention_heads": 12,
+  "num_codevector_groups": 2,
+  "num_codevectors_per_group": 320,
+  "num_conv_pos_embedding_groups": 16,
+  "num_conv_pos_embeddings": 128,
+  "num_feat_extract_layers": 7,
+  "num_hidden_layers": 12,
+  "num_negatives": 100,
+  "pad_token_id": 0,
+  "proj_codevector_dim": 256,
+  "transformers_version": "4.7.0.dev0",
+  "vocab_size": 32
+}
diff --git a/lipsync/hallo/wav2vec/wav2vec2-base-960h/feature_extractor_config.json b/lipsync/hallo/wav2vec/wav2vec2-base-960h/feature_extractor_config.json
@@ -0,0 +1,8 @@
+{
+  "do_normalize": true,
+  "feature_dim": 1,
+  "padding_side": "right",
+  "padding_value": 0.0,
+  "return_attention_mask": false,
+  "sampling_rate": 16000
+}
diff --git a/lipsync/hallo/wav2vec/wav2vec2-base-960h/preprocessor_config.json b/lipsync/hallo/wav2vec/wav2vec2-base-960h/preprocessor_config.json
@@ -0,0 +1,8 @@
+{
+  "do_normalize": true,
+  "feature_size": 1,
+  "padding_side": "right",
+  "padding_value": 0.0,
+  "return_attention_mask": false,
+  "sampling_rate": 16000
+}
diff --git a/lipsync/hallo/wav2vec/wav2vec2-base-960h/special_tokens_map.json b/lipsync/hallo/wav2vec/wav2vec2-base-960h/special_tokens_map.json
@@ -0,0 +1 @@
+{"bos_token": "<s>", "eos_token": "</s>", "unk_token": "<unk>", "pad_token": "<pad>"}
diff --git a/lipsync/hallo/wav2vec/wav2vec2-base-960h/tokenizer_config.json b/lipsync/hallo/wav2vec/wav2vec2-base-960h/tokenizer_config.json
@@ -0,0 +1 @@
+{"unk_token": "<unk>", "bos_token": "<s>", "eos_token": "</s>", "pad_token": "<pad>", "do_lower_case": false, "return_attention_mask": false, "do_normalize": true}
diff --git a/lipsync/hallo/wav2vec/wav2vec2-base-960h/vocab.json b/lipsync/hallo/wav2vec/wav2vec2-base-960h/vocab.json
@@ -0,0 +1 @@
+{"<pad>": 0, "<s>": 1, "</s>": 2, "<unk>": 3, "|": 4, "E": 5, "T": 6, "A": 7, "O": 8, "N": 9, "I": 10, "H": 11, "S": 12, "R": 13, "D": 14, "L": 15, "U": 16, "M": 17, "W": 18, "C": 19, "F": 20, "G": 21, "Y": 22, "P": 23, "B": 24, "V": 25, "K": 26, "'": 27, "X": 28, "J": 29, "Q": 30, "Z": 31}
Original file line number	Diff line number	Diff line change
		@@ -0,0 +1 @@
		{"bos_token": "<s>", "eos_token": "</s>", "unk_token": "<unk>", "pad_token": "<pad>"}
Original file line number	Diff line number	Diff line change
		@@ -0,0 +1 @@
		{"unk_token": "<unk>", "bos_token": "<s>", "eos_token": "</s>", "pad_token": "<pad>", "do_lower_case": false, "return_attention_mask": false, "do_normalize": true}
Original file line number	Diff line number	Diff line change
		@@ -0,0 +1 @@
		{"<pad>": 0, "<s>": 1, "</s>": 2, "<unk>": 3, "\|": 4, "E": 5, "T": 6, "A": 7, "O": 8, "N": 9, "I": 10, "H": 11, "S": 12, "R": 13, "D": 14, "L": 15, "U": 16, "M": 17, "W": 18, "C": 19, "F": 20, "G": 21, "Y": 22, "P": 23, "B": 24, "V": 25, "K": 26, "'": 27, "X": 28, "J": 29, "Q": 30, "Z": 31}