| |
| """ |
| Caption audio files for ACE-Step v1.5 training. |
| |
| Produces .txt files containing all training metadata: |
| - caption (from acestep-captioner) |
| - lyrics (from acestep-transcriber) |
| - bpm, keyscale, timesignature (from librosa) |
| - duration, language |
| |
| Requirements: |
| pip install torch torchaudio transformers librosa numpy |
| |
| Usage: |
| python caption_dir.py input_dir/ |
| python caption_dir.py input_dir/ --low_vram --skip_existing |
| """ |
|
|
| import argparse |
| import gc |
| import os |
| import glob |
| import logging |
| import warnings |
|
|
| import librosa |
| import numpy as np |
| import torch |
| import torchaudio |
| from tqdm import tqdm |
| from transformers import Qwen2_5OmniForConditionalGeneration, Qwen2_5OmniProcessor |
|
|
| warnings.filterwarnings("ignore") |
| logging.disable(logging.WARNING) |
|
|
| TARGET_SAMPLE_RATE = 16000 |
| CAPTIONER_ID = "ACE-Step/acestep-captioner" |
| TRANSCRIBER_ID = "ACE-Step/acestep-transcriber" |
|
|
| |
| MAJOR_PROFILE = np.array([6.35, 2.23, 3.48, 2.33, 4.38, 4.09, 2.52, 5.19, 2.39, 3.66, 2.29, 2.88]) |
| MINOR_PROFILE = np.array([6.33, 2.68, 3.52, 5.38, 2.60, 3.53, 2.54, 4.75, 3.98, 2.69, 3.34, 3.17]) |
| KEY_NAMES = ["C", "C#", "D", "D#", "E", "F", "F#", "G", "G#", "A", "A#", "B"] |
|
|
|
|
| def get_audio_files(input_dir): |
| extensions = ["*.wav", "*.mp3", "*.flac", "*.ogg", "*.WAV", "*.MP3", "*.FLAC"] |
| files = [] |
| for ext in extensions: |
| files.extend(glob.glob(os.path.join(input_dir, ext))) |
| return sorted(set(files)) |
|
|
|
|
| def load_audio_mono_16k(audio_path): |
| waveform, sr = torchaudio.load(audio_path) |
| if waveform.shape[0] > 1: |
| waveform = waveform.mean(dim=0, keepdim=True) |
| if sr != TARGET_SAMPLE_RATE: |
| waveform = torchaudio.functional.resample(waveform, sr, TARGET_SAMPLE_RATE) |
| return waveform.squeeze(0).numpy(), TARGET_SAMPLE_RATE |
|
|
|
|
| |
| |
| |
|
|
| def analyze_audio(audio_path): |
| """Extract BPM, key, and time signature from audio using librosa.""" |
| y, sr = librosa.load(audio_path, sr=22050, mono=True) |
| duration = librosa.get_duration(y=y, sr=sr) |
|
|
| |
| tempo, _ = librosa.beat.beat_track(y=y, sr=sr) |
| if hasattr(tempo, '__len__'): |
| tempo = tempo[0] |
| bpm = int(round(float(tempo))) |
|
|
| |
| chroma = librosa.feature.chroma_cqt(y=y, sr=sr) |
| chroma_avg = chroma.mean(axis=1) |
| major_corrs = np.array([np.corrcoef(np.roll(MAJOR_PROFILE, i), chroma_avg)[0, 1] for i in range(12)]) |
| minor_corrs = np.array([np.corrcoef(np.roll(MINOR_PROFILE, i), chroma_avg)[0, 1] for i in range(12)]) |
|
|
| best_major_idx = major_corrs.argmax() |
| best_minor_idx = minor_corrs.argmax() |
| if major_corrs[best_major_idx] >= minor_corrs[best_minor_idx]: |
| keyscale = f"{KEY_NAMES[best_major_idx]} major" |
| else: |
| keyscale = f"{KEY_NAMES[best_minor_idx]} minor" |
|
|
| |
| onset_env = librosa.onset.onset_strength(y=y, sr=sr) |
| tempo_est, beats = librosa.beat.beat_track(onset_envelope=onset_env, sr=sr) |
| if len(beats) >= 8: |
| beat_strengths = onset_env[beats] |
| |
| acf = np.correlate(beat_strengths - beat_strengths.mean(), |
| beat_strengths - beat_strengths.mean(), mode='full') |
| acf = acf[len(acf) // 2:] |
| if len(acf) > 6: |
| |
| score_3 = acf[3] if len(acf) > 3 else 0 |
| score_4 = acf[4] if len(acf) > 4 else 0 |
| timesig = "3" if score_3 > score_4 * 1.2 else "4" |
| else: |
| timesig = "4" |
| else: |
| timesig = "4" |
|
|
| return { |
| "bpm": bpm, |
| "keyscale": keyscale, |
| "timesignature": timesig, |
| "duration": int(round(duration)), |
| } |
|
|
|
|
| |
| |
| |
|
|
| def offload_to_cpu(model): |
| """Move model to CPU and free GPU memory.""" |
| if model is not None: |
| model.to("cpu") |
| gc.collect() |
| if torch.cuda.is_available(): |
| torch.cuda.empty_cache() |
|
|
|
|
| def load_qwen_model(model_id, device="cuda", dtype=torch.bfloat16): |
| """Load a Qwen2.5-Omni model.""" |
| model = Qwen2_5OmniForConditionalGeneration.from_pretrained( |
| model_id, torch_dtype=dtype, device_map=device, |
| ) |
| model.disable_talker() |
| processor = Qwen2_5OmniProcessor.from_pretrained(model_id) |
| return model, processor |
|
|
|
|
| def run_qwen_audio(model, processor, audio_data, sr, prompt_text): |
| """Run a Qwen2.5-Omni model on audio with a text prompt.""" |
| conversation = [ |
| { |
| "role": "user", |
| "content": [ |
| {"type": "audio", "audio": "<|audio_bos|><|AUDIO|><|audio_eos|>"}, |
| {"type": "text", "text": prompt_text}, |
| ], |
| } |
| ] |
| text = processor.apply_chat_template(conversation, add_generation_prompt=True, tokenize=False) |
| inputs = processor( |
| text=text, audio=[audio_data], images=None, videos=None, |
| return_tensors="pt", padding=True, sampling_rate=sr, |
| ) |
| inputs = inputs.to(model.device).to(model.dtype) |
| text_ids = model.generate(**inputs, return_audio=False) |
| output = processor.batch_decode(text_ids, skip_special_tokens=True, clean_up_tokenization_spaces=False) |
| result = output[0] |
| marker = "assistant\n" |
| if marker in result: |
| result = result[result.rfind(marker) + len(marker):] |
| return result.strip() |
|
|
|
|
| |
| |
| |
|
|
| def format_output(caption, lyrics, analysis, language="en"): |
| """Format all metadata into tagged format for easy parsing.""" |
| return ( |
| f"<CAPTION>\n{caption}\n</CAPTION>\n" |
| f"<LYRICS>\n{lyrics}\n</LYRICS>\n" |
| f"<BPM>{analysis['bpm']}</BPM>\n" |
| f"<KEYSCALE>{analysis['keyscale']}</KEYSCALE>\n" |
| f"<TIMESIGNATURE>{analysis['timesignature']}</TIMESIGNATURE>\n" |
| f"<DURATION>{analysis['duration']}</DURATION>\n" |
| f"<LANGUAGE>{language}</LANGUAGE>" |
| ) |
|
|
|
|
| def parse_caption_file(path): |
| """Parse a tagged caption file back into a dict.""" |
| import re |
| text = open(path, "r", encoding="utf-8").read() |
| def tag(name): |
| m = re.search(rf"<{name}>(.*?)</{name}>", text, re.DOTALL) |
| return m.group(1).strip() if m else "" |
| return { |
| "caption": tag("CAPTION"), |
| "lyrics": tag("LYRICS"), |
| "bpm": tag("BPM"), |
| "keyscale": tag("KEYSCALE"), |
| "timesignature": tag("TIMESIGNATURE"), |
| "duration": tag("DURATION"), |
| "language": tag("LANGUAGE"), |
| } |
|
|
|
|
| |
| |
| |
|
|
| def main(): |
| parser = argparse.ArgumentParser(description="Caption audio files for ACE-Step training") |
| parser.add_argument("input_dir", type=str, help="Directory containing audio files") |
| parser.add_argument("--skip_existing", action="store_true", help="Skip files that already have captions") |
| parser.add_argument("--low_vram", action="store_true", help="Offload models to CPU between stages") |
| parser.add_argument("--language", default="en", help="Default language code (default: en)") |
| args = parser.parse_args() |
|
|
| if not os.path.isdir(args.input_dir): |
| print(f"Error: {args.input_dir} is not a valid directory") |
| return |
|
|
| audio_files = get_audio_files(args.input_dir) |
| if not audio_files: |
| print("No audio files found in the directory") |
| return |
|
|
| print(f"Found {len(audio_files)} audio files") |
|
|
| |
| print("\n[Stage 1/3] Analyzing audio (BPM, key, time signature)...") |
| analyses = {} |
| for audio_path in tqdm(audio_files, desc="Analyzing"): |
| base_name = os.path.splitext(audio_path)[0] |
| if args.skip_existing and os.path.exists(base_name + ".txt"): |
| continue |
| try: |
| analyses[audio_path] = analyze_audio(audio_path) |
| except Exception as e: |
| print(f"\n Error analyzing {os.path.basename(audio_path)}: {e}") |
| analyses[audio_path] = {"bpm": 120, "keyscale": "C major", "timesignature": "4", |
| "duration": 30} |
|
|
| |
| files_to_process = [f for f in audio_files if f in analyses] |
| if not files_to_process: |
| print("All files already captioned (use without --skip_existing to overwrite)") |
| return |
|
|
| |
| print(f"\n[Stage 2/3] Captioning {len(files_to_process)} files...") |
| print(" Loading captioner model...") |
| captioner, cap_processor = load_qwen_model(CAPTIONER_ID) |
|
|
| captions = {} |
| for audio_path in tqdm(files_to_process, desc="Captioning"): |
| try: |
| audio_data, sr = load_audio_mono_16k(audio_path) |
| caption = run_qwen_audio( |
| captioner, cap_processor, audio_data, sr, |
| "*Task* Describe this music in detail. Include genre, mood, instrumentation, tempo feel, and vocal style if present." |
| ) |
| captions[audio_path] = caption |
| except Exception as e: |
| print(f"\n Error captioning {os.path.basename(audio_path)}: {e}") |
| captions[audio_path] = "" |
|
|
| if args.low_vram: |
| print(" Offloading captioner to CPU...") |
| offload_to_cpu(captioner) |
| del captioner, cap_processor |
|
|
| |
| print(f"\n[Stage 3/3] Transcribing lyrics for {len(files_to_process)} files...") |
| print(" Loading transcriber model...") |
| transcriber, trans_processor = load_qwen_model(TRANSCRIBER_ID) |
|
|
| lyrics_map = {} |
| for audio_path in tqdm(files_to_process, desc="Transcribing"): |
| try: |
| audio_data, sr = load_audio_mono_16k(audio_path) |
| lyrics = run_qwen_audio( |
| transcriber, trans_processor, audio_data, sr, |
| "*Task* Transcribe this audio in detail" |
| ) |
| lyrics_map[audio_path] = lyrics |
| except Exception as e: |
| print(f"\n Error transcribing {os.path.basename(audio_path)}: {e}") |
| lyrics_map[audio_path] = "[Instrumental]" |
|
|
| if args.low_vram: |
| print(" Offloading transcriber to CPU...") |
| offload_to_cpu(transcriber) |
| del transcriber, trans_processor |
|
|
| |
| print("\nWriting output files...") |
| for audio_path in files_to_process: |
| base_name = os.path.splitext(audio_path)[0] |
| output_path = base_name + ".txt" |
|
|
| caption = captions.get(audio_path, "") |
| lyrics = lyrics_map.get(audio_path, "[Instrumental]") |
| analysis = analyses[audio_path] |
|
|
| output = format_output(caption, lyrics, analysis, args.language) |
| with open(output_path, "w", encoding="utf-8") as f: |
| f.write(output) |
|
|
| print(f"Done! Processed {len(files_to_process)} files.") |
|
|
|
|
| if __name__ == "__main__": |
| main() |
|
|