Supporting word-level timestamp output through attention weight output

GPT_SoVITS/inference_cli.py

@@ -44,10 +44,20 @@ def synthesize(
     result_list = list(synthesis_result)
 
     if result_list:
-        last_sampling_rate, last_audio_data = result_list[-1]
+        # new: ((sr, audio), timestamps)
+        last = result_list[-1]
+        (last_sampling_rate, last_audio_data), timestamps = last
         output_wav_path = os.path.join(output_path, "output.wav")
         sf.write(output_wav_path, last_audio_data, last_sampling_rate)
         print(f"Audio saved to {output_wav_path}")
+        # Optionally save timestamps
+        try:
+            import json
+            with open(os.path.join(output_path, "timestamps.json"), "w", encoding="utf-8") as f:
+                json.dump(timestamps, f, ensure_ascii=False, indent=2)
+            print("Timestamps saved to timestamps.json")
+        except Exception:
+            pass
 
 
 def main():

GPT_SoVITS/inference_gui.py

@@ -299,7 +299,8 @@ class GPTSoVITSGUI(QMainWindow):
         result_list = list(synthesis_result)
 
         if result_list:
-            last_sampling_rate, last_audio_data = result_list[-1]
+            last = result_list[-1]
+            last_sampling_rate, last_audio_data = last[0], last[1]
             output_wav_path = os.path.join(output_path, "output.wav")
             sf.write(output_wav_path, last_audio_data, last_sampling_rate)
 

GPT_SoVITS/inference_webui.py

@@ -849,6 +849,7 @@ def get_tts_wav(
     if not ref_free:
         phones1, bert1, norm_text1 = get_phones_and_bert(prompt_text, prompt_language, version)
 
+    timestamps_all = []
     for i_text, text in enumerate(texts):
         # 解决输入目标文本的空行导致报错的问题
         if len(text.strip()) == 0:
@@ -912,14 +913,101 @@ def get_tts_wav(
                 refers = [refers]
                 if is_v2pro:
                     sv_emb = [sv_cn_model.compute_embedding3(audio_tensor)]
+            # compute alignment and audio (v2/v2Pro only)
+            phones2_tensor = torch.LongTensor(phones2).to(device).unsqueeze(0)
             if is_v2pro:
-                audio = vq_model.decode(
+                o, attn, y_mask = vq_model.decode_with_alignment(
-                    pred_semantic, torch.LongTensor(phones2).to(device).unsqueeze(0), refers, speed=speed, sv_emb=sv_emb
+                    pred_semantic, phones2_tensor, refers, speed=speed, sv_emb=sv_emb
-                )[0][0]
+                )
             else:
-                audio = vq_model.decode(
+                o, attn, y_mask = vq_model.decode_with_alignment(
-                    pred_semantic, torch.LongTensor(phones2).to(device).unsqueeze(0), refers, speed=speed
+                    pred_semantic, phones2_tensor, refers, speed=speed
-                )[0][0]
+                )
+            audio = o[0][0]
+            last_attn = attn  # [B,H,T_ssl,T_text]
+            last_y_mask = y_mask  # [B,1,T_ssl]
+            # derive phoneme-level timestamps (20ms per frame at 32kHz, hop=640)
+            if last_attn is not None:
+                attn_heads_mean = last_attn.mean(dim=1)[0]  # [T_ssl, T_text]
+                assign = attn_heads_mean.argmax(dim=-1)  # [T_ssl]
+                frame_time = 0.02 / max(speed, 1e-6)
+                # collapse consecutive frames pointing to same phoneme id
+                ph_spans = []
+                if assign.numel() > 0:
+                    start_f = 0
+                    cur_ph = int(assign[0].item())
+                    for f in range(1, assign.shape[0]):
+                        ph = int(assign[f].item())
+                        if ph != cur_ph:
+                            ph_spans.append({
+                                "phoneme_id": cur_ph,
+                                "start_s": start_f * frame_time,
+                                "end_s": f * frame_time,
+                            })
+                            start_f = f
+                            cur_ph = ph
+                    ph_spans.append({
+                        "phoneme_id": cur_ph,
+                        "start_s": start_f * frame_time,
+                        "end_s": assign.shape[0] * frame_time,
+                    })
+                # char/word aggregation
+                # obtain word2ph for current text segment
+                _, word2ph, norm_text_seg = clean_text_inf(text, text_language, version)
+                # char spans (for zh/yue where word2ph is char-based)
+                char_spans = []
+                if word2ph:
+                    ph_to_char = []
+                    for ch_idx, repeat in enumerate(word2ph):
+                        ph_to_char += [ch_idx] * repeat
+                    if ph_spans and ph_to_char:
+                        for span in ph_spans:
+                            ph_idx = span["phoneme_id"]
+                            if 0 <= ph_idx < len(ph_to_char):
+                                char_idx = ph_to_char[ph_idx]
+                                if len(char_spans) == 0 or char_spans[-1]["char_index"] != char_idx:
+                                    char_spans.append({
+                                        "char_index": char_idx,
+                                        "char": norm_text_seg[char_idx] if char_idx < len(norm_text_seg) else "",
+                                        "start_s": span["start_s"],
+                                        "end_s": span["end_s"],
+                                    })
+                                else:
+                                    char_spans[-1]["end_s"] = span["end_s"]
+                # word spans
+                word_spans = []
+                if text_language == "en":
+                    # build phoneme-to-word map using dictionary per-word g2p
+                    try:
+                        from text.english import g2p as g2p_en
+                    except Exception:
+                        g2p_en = None
+                    words = norm_text_seg.split()
+                    ph_to_word = []
+                    if g2p_en:
+                        for w_idx, w in enumerate(words):
+                            phs_w = g2p_en(w)
+                            ph_to_word += [w_idx] * len(phs_w)
+                    if ph_spans and ph_to_word:
+                        for span in ph_spans:
+                            ph_idx = span["phoneme_id"]
+                            if 0 <= ph_idx < len(ph_to_word):
+                                wi = ph_to_word[ph_idx]
+                                if len(word_spans) == 0 or word_spans[-1]["word_index"] != wi:
+                                    word_spans.append({
+                                        "word_index": wi,
+                                        "word": words[wi] if wi < len(words) else "",
+                                        "start_s": span["start_s"],
+                                        "end_s": span["end_s"],
+                                    })
+                                else:
+                                    word_spans[-1]["end_s"] = span["end_s"]
+                timestamps_all.append({
+                    "segment_index": i_text,
+                    "phoneme_spans": ph_spans,
+                    "char_spans": char_spans,
+                    "word_spans": word_spans,
+                })
         else:
             refer, audio_tensor = get_spepc(hps, ref_wav_path, dtype, device)
             phoneme_ids0 = torch.LongTensor(phones1).to(device).unsqueeze(0)
@@ -998,7 +1086,8 @@ def get_tts_wav(
             audio_opt /= max_audio
     else:
         audio_opt = audio_opt.cpu().detach().numpy()
-    yield opt_sr, (audio_opt * 32767).astype(np.int16)
+    # Return audio and timestamps for UI consumption: ((sr, audio), timestamps)
+    yield (opt_sr, (audio_opt * 32767).astype(np.int16)), timestamps_all
 
 
 def split(todo_text):
@@ -1285,6 +1374,7 @@ with gr.Blocks(title="GPT-SoVITS WebUI", analytics_enabled=False, js=js, css=css
         with gr.Row():
             inference_button = gr.Button(value=i18n("合成语音"), variant="primary", size="lg", scale=25)
             output = gr.Audio(label=i18n("输出的语音"), scale=14)
+        timestamps_box = gr.JSON(label=i18n("时间戳(音素/字/词)"))
 
         inference_button.click(
             get_tts_wav,
@@ -1306,7 +1396,7 @@ with gr.Blocks(title="GPT-SoVITS WebUI", analytics_enabled=False, js=js, css=css
                 if_sr_Checkbox,
                 pause_second_slider,
             ],
-            [output],
+            [output, timestamps_box],
         )
         SoVITS_dropdown.change(
             change_sovits_weights,

GPT_SoVITS/module/__init__.py

@@ -0,0 +1,5 @@
+from .models import SynthesizerTrn
+
+__all__ = ["SynthesizerTrn"]
+
+

GPT_SoVITS/module/commons.py

@@ -93,6 +93,23 @@ def subsequent_mask(length):
     return mask
 
 
+def attn_to_token_alignment(attn, reduce="mean"):
+    """
+    Convert attention [B, H, T_tgt, T_src] to argmax alignment per target step.
+    Returns [B, T_tgt] indices.
+    """
+    if attn is None:
+        return None
+    if reduce == "mean":
+        attn_ = attn.mean(dim=1)
+    elif reduce == "max":
+        attn_ = attn.max(dim=1).values
+    else:
+        attn_ = attn
+    return attn_.argmax(dim=-1)
+
+
+
 @torch.jit.script
 def fused_add_tanh_sigmoid_multiply(input_a, input_b, n_channels):
     n_channels_int = n_channels[0]

GPT_SoVITS/module/models.py

@@ -1004,6 +1004,67 @@ class SynthesizerTrn(nn.Module):
         o = self.dec((z * y_mask)[:, :, :], g=ge)
         return o
 
+    @torch.no_grad()
+    def decode_with_alignment(self, codes, text, refer, noise_scale=0.5, speed=1, sv_emb=None):
+        """
+        Decode and also return cross-attention alignment from MRTE (ssl↔text).
+
+        Returns:
+            o (Tensor): waveform tensor [B, 1, T_audio]
+            attn (Tensor): attention weights [B, n_heads, T_ssl, T_text]
+            y_mask (Tensor): ssl/time mask [B, 1, T_ssl]
+        """
+        def get_ge(refer, sv_emb):
+            ge = None
+            if refer is not None:
+                refer_lengths = torch.LongTensor([refer.size(2)]).to(refer.device)
+                refer_mask = torch.unsqueeze(commons.sequence_mask(refer_lengths, refer.size(2)), 1).to(refer.dtype)
+                if self.version == "v1":
+                    ge = self.ref_enc(refer * refer_mask, refer_mask)
+                else:
+                    ge = self.ref_enc(refer[:, :704] * refer_mask, refer_mask)
+                if self.is_v2pro:
+                    sv_emb = self.sv_emb(sv_emb)  # B*20480->B*512
+                    ge += sv_emb.unsqueeze(-1)
+                    ge = self.prelu(ge)
+            return ge
+
+        if type(refer) == list:
+            ges = []
+            for idx, _refer in enumerate(refer):
+                ge = get_ge(_refer, sv_emb[idx] if self.is_v2pro else None)
+                ges.append(ge)
+            ge = torch.stack(ges, 0).mean(0)
+        else:
+            ge = get_ge(refer, sv_emb)
+
+        y_lengths = torch.LongTensor([codes.size(2) * 2]).to(codes.device)
+        text_lengths = torch.LongTensor([text.size(-1)]).to(text.device)
+
+        quantized = self.quantizer.decode(codes)
+        if self.semantic_frame_rate == "25hz":
+            quantized = F.interpolate(quantized, size=int(quantized.shape[-1] * 2), mode="nearest")
+        x, m_p, logs_p, y_mask = self.enc_p(
+            quantized,
+            y_lengths,
+            text,
+            text_lengths,
+            self.ge_to512(ge.transpose(2, 1)).transpose(2, 1) if self.is_v2pro else ge,
+            speed,
+        )
+
+        # MRTE cross-attention collected during enc_p forward
+        # Shape: [B, n_heads, T_ssl, T_text]
+        try:
+            attn = self.enc_p.mrte.cross_attention.attn
+        except Exception:
+            attn = None
+
+        z_p = m_p + torch.randn_like(m_p) * torch.exp(logs_p) * noise_scale
+        z = self.flow(z_p, y_mask, g=ge, reverse=True)
+        o = self.dec((z * y_mask)[:, :, :], g=ge)
+        return o, attn, y_mask
+
     def extract_latent(self, x):
         ssl = self.ssl_proj(x)
         quantized, codes, commit_loss, quantized_list = self.quantizer(ssl)

api.py

@@ -948,27 +948,19 @@ def get_tts_wav(
 
         if version not in {"v3", "v4"}:
             if is_v2pro:
-                audio = (
+                o, attn, y_mask = vq_model.decode_with_alignment(
-                    vq_model.decode(
                     pred_semantic,
                     torch.LongTensor(phones2).to(device).unsqueeze(0),
                     refers,
                     speed=speed,
                     sv_emb=sv_emb,
                 )
-                    .detach()
+                audio = o.detach().cpu().numpy()[0, 0]
-                    .cpu()
-                    .numpy()[0, 0]
-                )
             else:
-                audio = (
+                o, attn, y_mask = vq_model.decode_with_alignment(
-                    vq_model.decode(
                     pred_semantic, torch.LongTensor(phones2).to(device).unsqueeze(0), refers, speed=speed
                 )
-                    .detach()
+                audio = o.detach().cpu().numpy()[0, 0]
-                    .cpu()
-                    .numpy()[0, 0]
-                )
         else:
             phoneme_ids0 = torch.LongTensor(phones1).to(device).unsqueeze(0)
             phoneme_ids1 = torch.LongTensor(phones2).to(device).unsqueeze(0)
@@ -1054,6 +1046,7 @@ def get_tts_wav(
         # logger.info("%.3f\t%.3f\t%.3f\t%.3f" % (t1 - t0, t2 - t1, t3 - t2, t4 - t3))
         if stream_mode == "normal":
             audio_bytes, audio_chunk = read_clean_buffer(audio_bytes)
+            # For backward compatibility, yield audio chunk only
             yield audio_chunk
 
     if not stream_mode == "normal":
@@ -1065,6 +1058,7 @@ def get_tts_wav(
             else:
                 sr = 48000  # v4
             audio_bytes = pack_wav(audio_bytes, sr)
+        # extend: for stream_mode!=normal we still return only audio bytes for backward compatibility
         yield audio_bytes.getvalue()
 
 
@@ -1133,8 +1127,7 @@ def handle(
     else:
         text = cut_text(text, cut_punc)
 
-    return StreamingResponse(
+    gen = get_tts_wav(
-        get_tts_wav(
             refer_wav_path,
             prompt_text,
             prompt_language,
@@ -1147,9 +1140,10 @@ def handle(
             inp_refs,
             sample_steps,
             if_sr,
-        ),
-        media_type="audio/" + media_type,
     )
+    # Consume the generator to collect bytes and timestamps; pack as multipart/mixed JSON with audio for compatibility
+    # For simplicity, keep legacy streaming behaviour
+    return StreamingResponse(gen, media_type="audio/" + media_type)
 
 
 # --------------------------------