Update app.py
Browse files
app.py
CHANGED
@@ -46,8 +46,6 @@ except FileNotFoundError:
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sys.path.append(os.path.join(os.path.dirname(os.path.abspath(__file__)), 'xcodec_mini_infer'))
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sys.path.append(os.path.join(os.path.dirname(os.path.abspath(__file__)), 'xcodec_mini_infer', 'descriptaudiocodec'))
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# don't change above code
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import argparse
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import numpy as np
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import json
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@@ -66,8 +64,8 @@ import time
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import copy
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from collections import Counter
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from models.soundstream_hubert_new import SoundStream
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#
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device = "cuda:0"
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@@ -81,9 +79,6 @@ model.eval()
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basic_model_config = './xcodec_mini_infer/final_ckpt/config.yaml'
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resume_path = './xcodec_mini_infer/final_ckpt/ckpt_00360000.pth'
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#config_path = './xcodec_mini_infer/decoders/config.yaml' # removed vocoder
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#vocal_decoder_path = './xcodec_mini_infer/decoders/decoder_131000.pth' # removed vocoder
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#inst_decoder_path = './xcodec_mini_infer/decoders/decoder_151000.pth' # removed vocoder
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mmtokenizer = _MMSentencePieceTokenizer("./mm_tokenizer_v0.2_hf/tokenizer.model")
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@@ -93,18 +88,8 @@ model_config = OmegaConf.load(basic_model_config)
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codec_model = eval(model_config.generator.name)(**model_config.generator.config).to(device)
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parameter_dict = torch.load(resume_path, map_location='cpu')
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codec_model.load_state_dict(parameter_dict['codec_model'])
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# codec_model = torch.compile(codec_model)
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codec_model.eval()
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# Preload and compile vocoders # removed vocoder
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#vocal_decoder, inst_decoder = build_codec_model(config_path, vocal_decoder_path, inst_decoder_path)
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#vocal_decoder.to(device)
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#inst_decoder.to(device)
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#vocal_decoder = torch.compile(vocal_decoder)
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#inst_decoder = torch.compile(inst_decoder)
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#vocal_decoder.eval()
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#inst_decoder.eval()
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@spaces.GPU(duration=120)
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def generate_music(
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max_new_tokens=5,
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@@ -117,234 +102,174 @@ def generate_music(
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prompt_end_time=30.0,
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cuda_idx=0,
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rescale=False,
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batch_size=1
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):
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if use_audio_prompt and not audio_prompt_path:
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raise FileNotFoundError("Please offer audio prompt filepath using '--audio_prompt_path', when you enable 'use_audio_prompt'!")
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cuda_idx = cuda_idx
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max_new_tokens = max_new_tokens * 100
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audio =
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vocals_ids = codectool.ids2npy(rearrange(codec_ids, "(n b) -> b n", b=2)[0])
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vocals_batch.append(vocals_ids)
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instrumentals_ids = codectool.ids2npy(rearrange(codec_ids, "(n b) -> b n", b=2)[1])
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instrumentals_batch.append(instrumentals_ids)
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vocals_batch = np.concatenate(vocals_batch, axis=1)
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instrumentals_batch = np.concatenate(instrumentals_batch, axis=1)
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vocals_list.append(vocals_batch)
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instrumentals_list.append(instrumentals_batch)
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vocals = np.concatenate(vocals_list, axis=1)
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instrumentals = np.concatenate(instrumentals_list, axis=1)
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vocal_save_path = os.path.join(stage1_output_dir, f"vocal_{random_ids[0]}".replace('.', '@') + '.npy')
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inst_save_path = os.path.join(stage1_output_dir, f"instrumental_{random_ids[0]}".replace('.', '@') + '.npy')
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np.save(vocal_save_path, vocals)
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np.save(inst_save_path, instrumentals)
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stage1_output_set.append(vocal_save_path)
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stage1_output_set.append(inst_save_path)
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print("Converting to Audio...")
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# convert audio tokens to audio
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def save_audio(wav: torch.Tensor, path, sample_rate: int, rescale: bool = False):
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folder_path = os.path.dirname(path)
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if not os.path.exists(folder_path):
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os.makedirs(folder_path)
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limit = 0.99
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max_val = wav.abs().max()
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wav = wav * min(limit / max_val, 1) if rescale else wav.clamp(-limit, limit)
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torchaudio.save(str(path), wav, sample_rate=sample_rate, encoding='PCM_S', bits_per_sample=16)
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# reconstruct tracks
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recons_output_dir = os.path.join(output_dir, "recons")
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recons_mix_dir = os.path.join(recons_output_dir, 'mix')
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os.makedirs(recons_mix_dir, exist_ok=True)
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tracks = []
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vocal_stem = None
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instrumental_stem = None
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sr = None
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for npy in stage1_output_set:
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codec_result = np.load(npy)
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decodec_rlt = []
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with torch.no_grad():
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decoded_waveform = codec_model.decode(
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torch.as_tensor(codec_result.astype(np.int16), dtype=torch.long).unsqueeze(0).permute(1, 0, 2).to(
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device))
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decoded_waveform = decoded_waveform.cpu().squeeze(0)
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decodec_rlt.append(torch.as_tensor(decoded_waveform))
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decodec_rlt = torch.cat(decodec_rlt, dim=-1)
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#save_path = os.path.join(recons_output_dir, os.path.splitext(os.path.basename(npy))[0] + ".mp3")
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#tracks.append(save_path)
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#save_audio(decodec_rlt, save_path, 16000)
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if 'vocal' in npy:
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vocal_stem = decodec_rlt.numpy()
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elif 'instrumental' in npy:
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instrumental_stem = decodec_rlt.numpy()
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sr = 16000
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sr, (instrumental_stem * 32767).astype(np.int16))
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def infer(genre_txt_content, lyrics_txt_content, num_segments=2, max_new_tokens=15):
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# Execute the command
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try:
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mixed_audio_data, vocal_audio_data, instrumental_audio_data = generate_music(genre_txt=genre_txt_content, lyrics_txt=lyrics_txt_content, run_n_segments=num_segments,
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cuda_idx=0, max_new_tokens=max_new_tokens
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return mixed_audio_data, vocal_audio_data, instrumental_audio_data
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except Exception as e:
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gr.Warning("An Error Occured: " + str(e))
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@@ -430,7 +355,6 @@ Living out my dreams with this mic and a deal
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inputs=[genre_txt, lyrics_txt],
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outputs=[music_out, vocal_out, instrumental_out],
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cache_examples=True,
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cache_mode="eager",
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fn=infer
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)
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sys.path.append(os.path.join(os.path.dirname(os.path.abspath(__file__)), 'xcodec_mini_infer'))
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sys.path.append(os.path.join(os.path.dirname(os.path.abspath(__file__)), 'xcodec_mini_infer', 'descriptaudiocodec'))
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import argparse
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import numpy as np
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import json
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import copy
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from collections import Counter
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from models.soundstream_hubert_new import SoundStream
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# don't change above code
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device = "cuda:0"
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basic_model_config = './xcodec_mini_infer/final_ckpt/config.yaml'
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resume_path = './xcodec_mini_infer/final_ckpt/ckpt_00360000.pth'
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mmtokenizer = _MMSentencePieceTokenizer("./mm_tokenizer_v0.2_hf/tokenizer.model")
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codec_model = eval(model_config.generator.name)(**model_config.generator.config).to(device)
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parameter_dict = torch.load(resume_path, map_location='cpu')
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codec_model.load_state_dict(parameter_dict['codec_model'])
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codec_model.eval()
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@spaces.GPU(duration=120)
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def generate_music(
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max_new_tokens=5,
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prompt_end_time=30.0,
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cuda_idx=0,
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rescale=False,
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):
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if use_audio_prompt and not audio_prompt_path:
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raise FileNotFoundError("Please offer audio prompt filepath using '--audio_prompt_path', when you enable 'use_audio_prompt'!")
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cuda_idx = cuda_idx
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max_new_tokens = max_new_tokens * 100
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class BlockTokenRangeProcessor(LogitsProcessor):
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def __init__(self, start_id, end_id):
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self.blocked_token_ids = list(range(start_id, end_id))
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def __call__(self, input_ids, scores):
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scores[:, self.blocked_token_ids] = -float("inf")
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return scores
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def load_audio_mono(filepath, sampling_rate=16000):
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audio, sr = torchaudio.load(filepath)
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# Convert to mono
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audio = torch.mean(audio, dim=0, keepdim=True)
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# Resample if needed
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if sr != sampling_rate:
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resampler = Resample(orig_freq=sr, new_freq=sampling_rate)
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audio = resampler(audio)
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return audio
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def split_lyrics(lyrics: str):
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pattern = r"\[(\w+)\](.*?)\n(?=\[|\Z)"
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segments = re.findall(pattern, lyrics, re.DOTALL)
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structured_lyrics = [f"[{seg[0]}]\n{seg[1].strip()}\n\n" for seg in segments]
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return structured_lyrics
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# Call the function and print the result
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stage1_output_set = []
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genres = genre_txt.strip()
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lyrics = split_lyrics(lyrics_txt + "\n")
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# intruction
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full_lyrics = "\n".join(lyrics)
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prompt_texts = [f"Generate music from the given lyrics segment by segment.\n[Genre] {genres}\n{full_lyrics}"]
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prompt_texts += lyrics
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random_id = uuid.uuid4()
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output_seq = None
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# Here is suggested decoding config
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top_p = 0.93
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temperature = 1.0
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repetition_penalty = 1.2
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# special tokens
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start_of_segment = mmtokenizer.tokenize('[start_of_segment]')
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end_of_segment = mmtokenizer.tokenize('[end_of_segment]')
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raw_output = None
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# Format text prompt
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run_n_segments = min(run_n_segments + 1, len(lyrics))
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print(list(enumerate(tqdm(prompt_texts[:run_n_segments]))))
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for i, p in enumerate(tqdm(prompt_texts[:run_n_segments])):
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section_text = p.replace('[start_of_segment]', '').replace('[end_of_segment]', '')
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guidance_scale = 1.5 if i <= 1 else 1.2
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if i == 0:
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continue
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if i == 1:
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if use_audio_prompt:
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audio_prompt = load_audio_mono(audio_prompt_path)
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audio_prompt.unsqueeze_(0)
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with torch.no_grad():
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raw_codes = codec_model.encode(audio_prompt.to(device), target_bw=0.5)
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raw_codes = raw_codes.transpose(0, 1)
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raw_codes = raw_codes.cpu().numpy().astype(np.int16)
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# Format audio prompt
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code_ids = codectool.npy2ids(raw_codes[0])
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audio_prompt_codec = code_ids[int(prompt_start_time * 50): int(prompt_end_time * 50)] # 50 is tps of xcodec
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audio_prompt_codec_ids = [mmtokenizer.soa] + codectool.sep_ids + audio_prompt_codec + [
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mmtokenizer.eoa]
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sentence_ids = mmtokenizer.tokenize("[start_of_reference]") + audio_prompt_codec_ids + mmtokenizer.tokenize(
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"[end_of_reference]")
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head_id = mmtokenizer.tokenize(prompt_texts[0]) + sentence_ids
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else:
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head_id = mmtokenizer.tokenize(prompt_texts[0])
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prompt_ids = head_id + start_of_segment + mmtokenizer.tokenize(section_text) + [mmtokenizer.soa] + codectool.sep_ids
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else:
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prompt_ids = end_of_segment + start_of_segment + mmtokenizer.tokenize(section_text) + [mmtokenizer.soa] + codectool.sep_ids
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prompt_ids = torch.as_tensor(prompt_ids).unsqueeze(0).to(device)
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input_ids = torch.cat([raw_output, prompt_ids], dim=1) if i > 1 else prompt_ids
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# Use window slicing in case output sequence exceeds the context of model
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max_context = 16384 - max_new_tokens - 1
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if input_ids.shape[-1] > max_context:
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print(
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f'Section {i}: output length {input_ids.shape[-1]} exceeding context length {max_context}, now using the last {max_context} tokens.')
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input_ids = input_ids[:, -(max_context):]
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with torch.inference_mode(), torch.autocast(device_type='cuda', dtype=torch.float16):
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output_seq = model.generate(
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input_ids=input_ids,
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max_new_tokens=max_new_tokens,
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min_new_tokens=100,
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do_sample=True,
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top_p=top_p,
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temperature=temperature,
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repetition_penalty=repetition_penalty,
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eos_token_id=mmtokenizer.eoa,
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pad_token_id=mmtokenizer.eoa,
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logits_processor=LogitsProcessorList([BlockTokenRangeProcessor(0, 32002), BlockTokenRangeProcessor(32016, 32016)]),
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guidance_scale=guidance_scale,
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use_cache=True
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)
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if output_seq[0][-1].item() != mmtokenizer.eoa:
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tensor_eoa = torch.as_tensor([[mmtokenizer.eoa]]).to(model.device)
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output_seq = torch.cat((output_seq, tensor_eoa), dim=1)
|
215 |
+
if i > 1:
|
216 |
+
raw_output = torch.cat([raw_output, prompt_ids, output_seq[:, input_ids.shape[-1]:]], dim=1)
|
217 |
+
else:
|
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+
raw_output = output_seq
|
219 |
+
print(len(raw_output))
|
220 |
+
|
221 |
+
# save raw output and check sanity
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+
ids = raw_output[0].cpu().numpy()
|
223 |
+
soa_idx = np.where(ids == mmtokenizer.soa)[0].tolist()
|
224 |
+
eoa_idx = np.where(ids == mmtokenizer.eoa)[0].tolist()
|
225 |
+
if len(soa_idx) != len(eoa_idx):
|
226 |
+
raise ValueError(f'invalid pairs of soa and eoa, Num of soa: {len(soa_idx)}, Num of eoa: {len(eoa_idx)}')
|
227 |
+
|
228 |
+
vocals = []
|
229 |
+
instrumentals = []
|
230 |
+
range_begin = 1 if use_audio_prompt else 0
|
231 |
+
for i in range(range_begin, len(soa_idx)):
|
232 |
+
codec_ids = ids[soa_idx[i] + 1:eoa_idx[i]]
|
233 |
+
if codec_ids[0] == 32016:
|
234 |
+
codec_ids = codec_ids[1:]
|
235 |
+
codec_ids = codec_ids[:2 * (codec_ids.shape[0] // 2)]
|
236 |
+
vocals_ids = codectool.ids2npy(rearrange(codec_ids, "(n b) -> b n", b=2)[0])
|
237 |
+
vocals.append(vocals_ids)
|
238 |
+
instrumentals_ids = codectool.ids2npy(rearrange(codec_ids, "(n b) -> b n", b=2)[1])
|
239 |
+
instrumentals.append(instrumentals_ids)
|
240 |
+
vocals = np.concatenate(vocals, axis=1)
|
241 |
+
instrumentals = np.concatenate(instrumentals, axis=1)
|
242 |
+
|
243 |
+
vocal_audio = None
|
244 |
+
instrumental_audio = None
|
245 |
+
mixed_audio = None
|
246 |
+
|
247 |
+
# convert audio tokens to audio
|
248 |
+
def convert_to_audio(codec_result, rescale):
|
249 |
+
with torch.no_grad():
|
250 |
+
decoded_waveform = codec_model.decode(
|
251 |
+
torch.as_tensor(codec_result.astype(np.int16), dtype=torch.long).unsqueeze(0).permute(1, 0, 2).to(
|
252 |
+
device))
|
253 |
+
decoded_waveform = decoded_waveform.cpu().squeeze(0)
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|
254 |
|
255 |
+
limit = 0.99
|
256 |
+
max_val = decoded_waveform.abs().max()
|
257 |
+
scaled_waveform = decoded_waveform * min(limit / max_val, 1) if rescale else decoded_waveform.clamp(-limit, limit)
|
258 |
+
return (16000, (scaled_waveform * 32767).astype(np.int16))
|
|
|
259 |
|
260 |
+
vocal_audio = convert_to_audio(vocals, rescale)
|
261 |
+
instrumental_audio = convert_to_audio(instrumentals, rescale)
|
262 |
+
|
263 |
+
mix_stem = (vocal_audio[1] + instrumental_audio[1]) / 1 # mixing by summing and dividing
|
264 |
+
mixed_audio = (vocal_audio[0], mix_stem) # same sample rate
|
265 |
+
|
266 |
+
return mixed_audio, vocal_audio, instrumental_audio
|
267 |
|
268 |
def infer(genre_txt_content, lyrics_txt_content, num_segments=2, max_new_tokens=15):
|
269 |
# Execute the command
|
270 |
try:
|
271 |
mixed_audio_data, vocal_audio_data, instrumental_audio_data = generate_music(genre_txt=genre_txt_content, lyrics_txt=lyrics_txt_content, run_n_segments=num_segments,
|
272 |
+
cuda_idx=0, max_new_tokens=max_new_tokens)
|
273 |
return mixed_audio_data, vocal_audio_data, instrumental_audio_data
|
274 |
except Exception as e:
|
275 |
gr.Warning("An Error Occured: " + str(e))
|
|
|
355 |
inputs=[genre_txt, lyrics_txt],
|
356 |
outputs=[music_out, vocal_out, instrumental_out],
|
357 |
cache_examples=True,
|
|
|
358 |
fn=infer
|
359 |
)
|
360 |
|