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2 Commits
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27e20fcf39 | ||
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dff57ebd2a |
@@ -89,6 +89,12 @@ fix_duration = [
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# parts_to_edit = [[0.84, 1.4], [1.92, 2.4], [4.26, 6.26], ]
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# fix_duration = None # use origin text duration
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# audio_to_edit = "src/f5_tts/infer/examples/basic/basic_ref_zh.wav"
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# origin_text = "对,这就是我,万人敬仰的太乙真人。"
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# target_text = "对,这就是你,万人敬仰的李白金星。"
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# parts_to_edit = [[1.500, 2.784], [4.083, 6.760]]
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# fix_duration = [1.284, 2.677]
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# -------------------------------------------------#
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@@ -138,28 +144,55 @@ if rms < target_rms:
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if sr != target_sample_rate:
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resampler = torchaudio.transforms.Resample(sr, target_sample_rate)
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audio = resampler(audio)
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offset = 0
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audio_ = torch.zeros(1, 0)
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edit_mask = torch.zeros(1, 0, dtype=torch.bool)
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# Convert to mel spectrogram FIRST (on clean original audio)
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# This avoids boundary artifacts from mel windows straddling zeros and real audio
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audio = audio.to(device)
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with torch.inference_mode():
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original_mel = model.mel_spec(audio) # (batch, n_mel, n_frames)
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original_mel = original_mel.permute(0, 2, 1) # (batch, n_frames, n_mel)
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# Build mel_cond and edit_mask at FRAME level
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# Insert zero frames in mel domain instead of zero samples in wav domain
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offset_frame = 0
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mel_cond = torch.zeros(1, 0, n_mel_channels, device=device)
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edit_mask = torch.zeros(1, 0, dtype=torch.bool, device=device)
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fix_dur_list = fix_duration.copy() if fix_duration is not None else None
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for part in parts_to_edit:
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start, end = part
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part_dur = end - start if fix_duration is None else fix_duration.pop(0)
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part_dur = part_dur * target_sample_rate
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start = start * target_sample_rate
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audio_ = torch.cat((audio_, audio[:, round(offset) : round(start)], torch.zeros(1, round(part_dur))), dim=-1)
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part_dur_sec = end - start if fix_dur_list is None else fix_dur_list.pop(0)
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# Convert to frames (this is the authoritative unit)
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start_frame = round(start * target_sample_rate / hop_length)
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end_frame = round(end * target_sample_rate / hop_length)
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part_dur_frames = round(part_dur_sec * target_sample_rate / hop_length)
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# Number of frames for the kept (non-edited) region
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keep_frames = start_frame - offset_frame
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# Build mel_cond: original mel frames + zero frames for edit region
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mel_cond = torch.cat(
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(
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mel_cond,
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original_mel[:, offset_frame:start_frame, :],
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torch.zeros(1, part_dur_frames, n_mel_channels, device=device),
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),
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dim=1,
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)
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edit_mask = torch.cat(
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(
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edit_mask,
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torch.ones(1, round((start - offset) / hop_length), dtype=torch.bool),
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torch.zeros(1, round(part_dur / hop_length), dtype=torch.bool),
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torch.ones(1, keep_frames, dtype=torch.bool, device=device),
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torch.zeros(1, part_dur_frames, dtype=torch.bool, device=device),
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),
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dim=-1,
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)
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offset = end * target_sample_rate
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audio = torch.cat((audio_, audio[:, round(offset) :]), dim=-1)
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edit_mask = F.pad(edit_mask, (0, audio.shape[-1] // hop_length - edit_mask.shape[-1] + 1), value=True)
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audio = audio.to(device)
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edit_mask = edit_mask.to(device)
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offset_frame = end_frame
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# Append remaining mel frames after last edit
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mel_cond = torch.cat((mel_cond, original_mel[:, offset_frame:, :]), dim=1)
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edit_mask = F.pad(edit_mask, (0, mel_cond.shape[1] - edit_mask.shape[-1]), value=True)
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# Text
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text_list = [target_text]
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@@ -170,14 +203,13 @@ else:
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print(f"text : {text_list}")
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print(f"pinyin: {final_text_list}")
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# Duration
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ref_audio_len = 0
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duration = audio.shape[-1] // hop_length
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# Duration - use mel_cond length (not raw audio length)
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duration = mel_cond.shape[1]
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# Inference
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# Inference - pass mel_cond directly (not wav)
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with torch.inference_mode():
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generated, trajectory = model.sample(
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cond=audio,
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cond=mel_cond, # Now passing mel directly, not wav
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text=final_text_list,
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duration=duration,
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steps=nfe_step,
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@@ -190,7 +222,6 @@ with torch.inference_mode():
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# Final result
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generated = generated.to(torch.float32)
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generated = generated[:, ref_audio_len:, :]
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gen_mel_spec = generated.permute(0, 2, 1)
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if mel_spec_type == "vocos":
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generated_wave = vocoder.decode(gen_mel_spec).cpu()
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