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clean up and update README #68

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7 changes: 5 additions & 2 deletions README.md
Original file line number Diff line number Diff line change
Expand Up @@ -27,8 +27,10 @@ developers to train custom multimodal large language model (MLLM), focusing on <
5. [Acknowledge](#acknowledge)

# News
- [Update Apr. 28, 2024] Recipes for automated audio captioning (AAC) with SOTA performance has been supported.
- [Update Mar. 31, 2024] Recipes for automatic speech recognition (ASR) with SOTA performance has been supported.
- [Update May. 8, 2024] Recipes for [visual speech recognition (VSR)](examples/vsr_LRS3/README.md) has been supported.
- [Update May. 4, 2024] Recipes for [zero-shot text-to-speech (TTS)](examples/vallex/README.md) has been supported.
- [Update Apr. 28, 2024] Recipes for [automated audio captioning (AAC)](examples/aac_audiocaps/README.md) has been supported.
- [Update Mar. 31, 2024] Recipes for [automatic speech recognition (ASR)](examples/asr_librispeech/README.md) has been supported.

# Installation
```bash
Expand Down Expand Up @@ -61,6 +63,7 @@ We provide reference implementations of various LLM-based speech, audio, and mus
- **Speech Task**
- [Automatic Speech Recognition (ASR)](examples/asr_librispeech/README.md)
- [Text-to-Speech (TTS)](examples/vallex/README.md)
- [Visual Speech Recognition (VSR)](examples/vsr_LRS3/README.md)
- **Audio Task**
- [Automated Audio Captioning (AAC)](examples/aac_audiocaps/README.md)

Expand Down
80 changes: 1 addition & 79 deletions examples/vsr_LRS3/model/slam_model_vsr.py
Original file line number Diff line number Diff line change
Expand Up @@ -74,82 +74,4 @@ def __init__(
train_config,
model_config,
**kwargs,
)


@torch.no_grad()
def inference(
self,
wav_path=None,
prompt=None,
generation_config=None,
logits_processor=None,
stopping_criteria=None,
prefix_allowed_tokens_fn=None,
synced_gpus=None,
assistant_model=None,
streamer=None,
negative_prompt_ids=None,
negative_prompt_attention_mask=None,
**kwargs,
):
# inference for asr model

device = kwargs.get("device", "cuda")
if os.path.exists(wav_path): # Audio-Text QA
import whisper

audio_raw = whisper.load_audio(wav_path)
audio_raw = whisper.pad_or_trim(audio_raw)

mel_size = getattr(
self.dataset_config, "mel_size", 80
) # 80 for large v1 and v2, 128 for large v3
audio_mel = (
whisper.log_mel_spectrogram(audio_raw, n_mels=mel_size)
.permute(1, 0)[None, :, :]
.to(device)
)

encoder_outs = self.encoder.extract_variable_length_features(
audio_mel.permute(0, 2, 1)
)

if self.model_config.encoder_projector == "q-former":
audio_mel_post_mask = torch.ones(
encoder_outs.size()[:-1], dtype=torch.long
).to(encoder_outs.device)
encoder_outs = self.encoder_projector(encoder_outs, audio_mel_post_mask)
if self.model_config.encoder_projector == "linear":
encoder_outs = self.encoder_projector(encoder_outs)
else: # Text QA
encoder_outs = torch.empty(
1, 0, self.llm.model.embed_tokens.embedding_dim
).to(device)

prompt = "USER: {}\n ASSISTANT:".format(prompt)
prompt_ids = self.tokenizer.encode(prompt)
prompt_length = len(prompt_ids)
prompt_ids = torch.tensor(prompt_ids, dtype=torch.int64).to(device)

if hasattr(self.llm.model, "embed_tokens"):
inputs_embeds = self.llm.model.embed_tokens(prompt_ids)
elif hasattr(self.llm.model.model, "embed_tokens"):
inputs_embeds = self.llm.model.model.embed_tokens(prompt_ids)
else:
inputs_embeds = self.llm.model.model.model.embed_tokens(prompt_ids)

inputs_embeds = torch.cat(
(encoder_outs, inputs_embeds[None, :, :]), dim=1
) # [audio,prompt]

attention_mask = torch.ones(inputs_embeds.size()[:-1], dtype=torch.long).to(
inputs_embeds.device
)

# generate
model_outputs = self.generate(
inputs_embeds=inputs_embeds, attention_mask=attention_mask, **kwargs
)

return model_outputs
)
228 changes: 0 additions & 228 deletions src/slam_llm/models/AV/av_net.py

This file was deleted.

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