Merge pull request #1242 from acadarmeria/fix-speech-edit-mel-domain

Fix speech editing boundary artifacts by working in mel domain

pyproject.toml

@@ -4,7 +4,7 @@ build-backend = "setuptools.build_meta"
 
 [project]
 name = "f5-tts"
-version = "1.1.12"
+version = "1.1.15"
 description = "F5-TTS: A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching"
 readme = "README.md"
 license = {text = "MIT License"}

src/f5_tts/infer/infer_gradio.py

@@ -221,7 +221,7 @@ with gr.Blocks() as app_tts:
         )
         gen_text_file = gr.File(label="Load Text to Generate from File (.txt)", file_types=[".txt"], scale=1)
     generate_btn = gr.Button("Synthesize", variant="primary")
-    with gr.Accordion("Advanced Settings", open=False):
+    with gr.Accordion("Advanced Settings", open=True) as adv_settn:
         with gr.Row():
             ref_text_input = gr.Textbox(
                 label="Reference Text",
@@ -269,6 +269,17 @@ with gr.Blocks() as app_tts:
             info="Set the duration of the cross-fade between audio clips.",
         )
 
+    def collapse_accordion():
+        return gr.Accordion(open=False)
+
+    # Workaround for https://github.com/SWivid/F5-TTS/issues/1239#issuecomment-3677987413
+    # i.e. to set gr.Accordion(open=True) by default, then collapse manually Blocks loaded
+    app_tts.load(
+        fn=collapse_accordion,
+        inputs=None,
+        outputs=adv_settn,
+    )
+
     audio_output = gr.Audio(label="Synthesized Audio")
     spectrogram_output = gr.Image(label="Spectrogram")
 

src/f5_tts/infer/speech_edit.py

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