diff --git a/main/manager-api/src/main/resources/db/changelog/202506051538.sql b/main/manager-api/src/main/resources/db/changelog/202506051538.sql
new file mode 100644
index 00000000..cfe65da7
--- /dev/null
+++ b/main/manager-api/src/main/resources/db/changelog/202506051538.sql
@@ -0,0 +1,20 @@
+-- 增加LinkeraiTTS供应器和模型配置
+delete from `ai_model_provider` where id = 'SYSTEM_TTS_LinkeraiTTS';
+INSERT INTO `ai_model_provider` (`id`, `model_type`, `provider_code`, `name`, `fields`, `sort`, `creator`, `create_date`, `updater`, `update_date`) VALUES
+('SYSTEM_TTS_LinkeraiTTS', 'TTS', 'linkerai', 'Linkerai语音合成', '[{"key":"api_url","label":"API地址","type":"string"},{"key":"audio_format","label":"音频格式","type":"string"},{"key":"access_token","label":"访问令牌","type":"string"},{"key":"voice","label":"默认音色","type":"string"}]', 14, 1, NOW(), 1, NOW());
+
+delete from `ai_model_config` where id = 'TTS_LinkeraiTTS';
+INSERT INTO `ai_model_config` VALUES ('TTS_LinkeraiTTS', 'TTS', 'LinkeraiTTS', 'Linkerai语音合成', 0, 1, '{\"type\": \"linkerai\", \"api_url\": \"https://tts.linkerai.cn/tts\", \"audio_format\": \"pcm\", \"access_token\": \"U4YdYXVfpwWnk2t5Gp822zWPCuORyeJL\", \"voice\": \"OUeAo1mhq6IBExi\"}', NULL, NULL, 17, NULL, NULL, NULL, NULL);
+
+-- LinkeraiTTS模型配置说明文档
+UPDATE `ai_model_config` SET 
+`doc_link` = 'https://tts.linkerai.cn/docs',
+`remark` = 'Linkerai语音合成服务配置说明：
+1. 访问 https://linkerai.cn 注册并获取访问令牌
+2. 默认的access_token供测试使用，请勿用于商业用途
+3. 支持声音克隆功能，可自行上传音频，填入voice参数
+4. 如果voice参数为空，将使用默认声音' WHERE `id` = 'TTS_LinkeraiTTS';
+
+
+delete from `ai_tts_voice` where tts_model_id = 'TTS_LinkeraiTTS';
+INSERT INTO `ai_tts_voice` VALUES ('TTS_LinkeraiTTS_0001', 'TTS_LinkeraiTTS', '芷若', 'OUeAo1mhq6IBExi', '中文', NULL, NULL, 1, NULL, NULL, NULL, NULL);
diff --git a/main/manager-api/src/main/resources/db/changelog/db.changelog-master.yaml b/main/manager-api/src/main/resources/db/changelog/db.changelog-master.yaml
index 0583fce9..cb3007aa 100755
--- a/main/manager-api/src/main/resources/db/changelog/db.changelog-master.yaml
+++ b/main/manager-api/src/main/resources/db/changelog/db.changelog-master.yaml
@@ -190,4 +190,11 @@ databaseChangeLog:
       changes:
         - sqlFile:
             encoding: utf8
-            path: classpath:db/changelog/202506032232.sql
\ No newline at end of file
+            path: classpath:db/changelog/202506032232.sql
+  - changeSet:
+      id: 202506051538
+      author: hrz
+      changes:
+        - sqlFile:
+            encoding: utf8
+            path: classpath:db/changelog/202506051538.sql
\ No newline at end of file
diff --git a/main/xiaozhi-server/config.yaml b/main/xiaozhi-server/config.yaml
index a7ff54f3..7721b132 100644
--- a/main/xiaozhi-server/config.yaml
+++ b/main/xiaozhi-server/config.yaml
@@ -324,7 +324,7 @@ VAD:
     type: silero
     threshold: 0.5
     model_dir: models/snakers4_silero-vad
-    min_silence_duration_ms: 700  # 如果说话停顿比较长，可以把这个值设置大一些
+    min_silence_duration_ms: 200  # 如果说话停顿比较长，可以把这个值设置大一些
 
 LLM:
   # 所有openai类型均可以修改超参，以AliLLM为例
@@ -758,11 +758,12 @@ TTS:
     output_dir: tmp/
   LinkeraiTTS:
     type: linkerai
-    api_url: https://tts.linkerai.top/tts
-    audio_format: "opus"
+    api_url: https://tts.linkerai.cn/tts
+    audio_format: "pcm"
     # 默认的access_token供大家测试时免费使用的，此access_token请勿用于商业用途
-    # 如果效果不错，可自行申请token，申请地址：https://linkerai.top
-    # 各参数意义见开发文档：https://tts.linkerai.top/docs#/default/text_to_speech_tts_get
+    # 如果效果不错，可自行申请token，申请地址：https://linkerai.cn
+    # 各参数意义见开发文档：https://tts.linkerai.cn/docs
+    # 支持声音克隆，可自行上传音频，填入voice参数，voice参数为空时，使用默认声音
     access_token: "U4YdYXVfpwWnk2t5Gp822zWPCuORyeJL"
-    voice: "7OEPouTL46bS2Qe"
+    voice: "OUeAo1mhq6IBExi"
     output_dir: tmp/
\ No newline at end of file
diff --git a/main/xiaozhi-server/core/providers/tts/base.py b/main/xiaozhi-server/core/providers/tts/base.py
index e82a8725..c467e99a 100644
--- a/main/xiaozhi-server/core/providers/tts/base.py
+++ b/main/xiaozhi-server/core/providers/tts/base.py
@@ -37,6 +37,7 @@ class TTSProviderBase(ABC):
         self.tts_text_queue = queue.Queue()
         self.tts_audio_queue = queue.Queue()
         self.tts_audio_first_sentence = True
+        self.before_stop_play_files = []
 
         self.tts_text_buff = []
         self.punctuations = (
@@ -324,6 +325,14 @@ class TTSProviderBase(ABC):
             os.remove(tts_file)
         return audio_datas
 
+    def _process_before_stop_play_files(self):
+        for tts_file, text in self.before_stop_play_files:
+            if tts_file and os.path.exists(tts_file):
+                audio_datas = self._process_audio_file(tts_file)
+                self.tts_audio_queue.put((SentenceType.MIDDLE, audio_datas, text))
+        self.before_stop_play_files.clear()
+        self.tts_audio_queue.put((SentenceType.LAST, [], None))
+
     def _process_remaining_text(self):
         """处理剩余的文本并生成语音
 
diff --git a/main/xiaozhi-server/core/providers/tts/huoshan_double_stream.py b/main/xiaozhi-server/core/providers/tts/huoshan_double_stream.py
index b9dc399b..b248f5ce 100644
--- a/main/xiaozhi-server/core/providers/tts/huoshan_double_stream.py
+++ b/main/xiaozhi-server/core/providers/tts/huoshan_double_stream.py
@@ -154,7 +154,6 @@ class TTSProvider(TTSProviderBase):
         self.header = {"Authorization": f"{self.authorization}{self.access_token}"}
         self.enable_two_way = True
         self.tts_text = ""
-        self.before_stop_play_files = []
         self.opus_encoder = opus_encoder_utils.OpusEncoderUtils(
             sample_rate=16000, channels=1, frame_size_ms=60
         )
@@ -432,14 +431,7 @@ class TTSProvider(TTSProviderBase):
                         is_first_sentence = False
                     elif res.optional.event == EVENT_SessionFinished:
                         logger.bind(tag=TAG).debug(f"会话结束～～")
-                        for tts_file, text in self.before_stop_play_files:
-                            if tts_file and os.path.exists(tts_file):
-                                audio_datas = self._process_audio_file(tts_file)
-                                self.tts_audio_queue.put(
-                                    (SentenceType.MIDDLE, audio_datas, text)
-                                )
-                        self.before_stop_play_files.clear()
-                        self.tts_audio_queue.put((SentenceType.LAST, [], None))
+                        self._process_before_stop_play_files()
                         break
                 except websockets.ConnectionClosed:
                     logger.bind(tag=TAG).warning("WebSocket连接已关闭")
diff --git a/main/xiaozhi-server/core/providers/tts/linkerai.py b/main/xiaozhi-server/core/providers/tts/linkerai.py
index 767d2662..dd8d0e6b 100644
--- a/main/xiaozhi-server/core/providers/tts/linkerai.py
+++ b/main/xiaozhi-server/core/providers/tts/linkerai.py
@@ -1,8 +1,5 @@
-import os
-import time
 import queue
 import asyncio
-import requests
 import traceback
 import aiohttp
 from config.logger import setup_logging
@@ -24,6 +21,7 @@ class TTSProvider(TTSProviderBase):
         self.api_url = config.get("api_url")
         self.audio_format = "pcm"
         self.before_stop_play_files = []
+        self.segment_count = 0  # 添加片段计数器
 
         # 创建Opus编码器
         self.opus_encoder = opus_encoder_utils.OpusEncoderUtils(
@@ -50,7 +48,7 @@ class TTSProvider(TTSProviderBase):
                     self.tts_stop_request = False
                     self.processed_chars = 0
                     self.tts_text_buff = []
-                    self.is_first_sentence = True
+                    self.segment_count = 0
                     self.tts_audio_first_sentence = True
                     self.before_stop_play_files.clear()
                 elif ContentType.TEXT == message.content_type:
@@ -60,7 +58,6 @@ class TTSProvider(TTSProviderBase):
                         self.to_tts(segment_text)
 
                 elif ContentType.FILE == message.content_type:
-                    self._process_remaining_text()
                     logger.bind(tag=TAG).info(
                         f"添加音频文件到待播放列表: {message.content_file}"
                     )
@@ -69,7 +66,8 @@ class TTSProvider(TTSProviderBase):
                     )
 
                 if message.sentence_type == SentenceType.LAST:
-                    self._process_remaining_text()
+                    # 处理剩余的文本
+                    self._process_remaining_text(True)
 
             except queue.Empty:
                 continue
@@ -78,7 +76,7 @@ class TTSProvider(TTSProviderBase):
                     f"处理TTS文本失败: {str(e)}, 类型: {type(e).__name__}, 堆栈: {traceback.format_exc()}"
                 )
 
-    def _process_remaining_text(self):
+    def _process_remaining_text(self, is_last=False):
         """处理剩余的文本并生成语音
 
         Returns:
@@ -89,15 +87,17 @@ class TTSProvider(TTSProviderBase):
         if remaining_text:
             segment_text = textUtils.get_string_no_punctuation_or_emoji(remaining_text)
             if segment_text:
-                self.to_tts(segment_text)
+                self.to_tts(segment_text, is_last)
                 self.processed_chars += len(full_text)
+            else:
+                self._process_before_stop_play_files()
 
-    def to_tts(self, text):
+    def to_tts(self, text, is_last=False):
         try:
             max_repeat_time = 5
             text = MarkdownCleaner.clean_markdown(text)
             try:
-                asyncio.run(self.text_to_speak(text, None))
+                asyncio.run(self.text_to_speak(text, is_last))
             except Exception as e:
                 logger.bind(tag=TAG).warning(
                     f"语音生成失败{5 - max_repeat_time + 1}次: {text}，错误: {e}"
@@ -121,9 +121,9 @@ class TTSProvider(TTSProviderBase):
     # linkerai单流式TTS重写父类的方法--结束
     ###################################################################################
 
-    async def text_to_speak(self, text, _):
+    async def text_to_speak(self, text, is_last):
         """流式处理TTS音频，每句只推送一次音频列表"""
-        await self._tts_request(text)
+        await self._tts_request(text, is_last)
 
     async def close(self):
         """资源清理"""
@@ -131,9 +131,7 @@ class TTSProvider(TTSProviderBase):
         if hasattr(self, "opus_encoder"):
             self.opus_encoder.close()
 
-    async def _tts_request(self, text: str) -> None:
-        """发送TTS请求"""
-        start_time = time.time()
+    async def _tts_request(self, text: str, is_last: bool) -> None:
         params = {
             "tts_text": text,
             "spk_id": self.voice,
@@ -141,41 +139,89 @@ class TTSProvider(TTSProviderBase):
             "stream": "true",
             "target_sr": 16000,
             "audio_format": "pcm",
-            "instruct_text": "",
+            "instruct_text": "请生成一段自然流畅的语音",
         }
         headers = {
             "Authorization": f"Bearer {self.access_token}",
             "Content-Type": "application/json",
         }
 
+        # 一帧 PCM 所需字节数：60 ms × 16 kHz × 1 ch × 2 B = 1 920
+        frame_bytes = int(
+            self.opus_encoder.sample_rate
+            * self.opus_encoder.channels  # 1
+            * self.opus_encoder.frame_size_ms
+            / 1000
+            * 2
+        )  # 16-bit = 2 bytes
+
         try:
             async with aiohttp.ClientSession() as session:
                 async with session.get(
                     self.api_url, params=params, headers=headers, timeout=10
-                ) as response:
-                    if response.status != 200:
-                        logger.error(
-                            f"TTS请求失败: {response.status}, {await response.text()}"
-                        )
-                        # 推送空LAST，防止播放端卡死
+                ) as resp:
+
+                    if resp.status != 200:
+                        logger.error(f"TTS请求失败: {resp.status}, {await resp.text()}")
                         self.tts_audio_queue.put((SentenceType.LAST, [], None))
                         return
 
-                    logger.info(
-                        f"TTS请求成功: {text}, 耗时: {time.time() - start_time}秒"
-                    )
+                    self.pcm_buffer.clear()
+                    opus_datas_cache = []
 
-                    # 流式处理音频数据
-                    async for chunk in response.content.iter_chunks():
-                        if chunk[0]:  # 确保数据不为空
-                            opus_data = self.opus_encoder.encode_pcm_to_opus(
-                                chunk[0], end_of_stream=True
+                    # 兼容 iter_chunked / iter_chunks / iter_any
+                    async for chunk in resp.content.iter_any():
+                        data = chunk[0] if isinstance(chunk, (list, tuple)) else chunk
+                        if not data:
+                            continue
+
+                        # 拼到 buffer
+                        self.pcm_buffer.extend(data)
+
+                        # 够一帧就编码
+                        while len(self.pcm_buffer) >= frame_bytes:
+                            frame = bytes(self.pcm_buffer[:frame_bytes])
+                            del self.pcm_buffer[:frame_bytes]
+
+                            opus = self.opus_encoder.encode_pcm_to_opus(
+                                frame, end_of_stream=False
                             )
-                            if opus_data:
+                            if opus:
+                                if self.segment_count < 10:  # 前10个片段直接发送
+                                    self.tts_audio_queue.put(
+                                        (SentenceType.MIDDLE, opus, text)
+                                    )
+                                    self.segment_count += 1
+                                else:
+                                    opus_datas_cache.extend(opus)
+
+                    # flush 剩余不足一帧的数据
+                    if self.pcm_buffer:
+                        opus = self.opus_encoder.encode_pcm_to_opus(
+                            bytes(self.pcm_buffer), end_of_stream=True
+                        )
+                        if opus:
+                            if self.segment_count < 10:  # 前10个片段直接发送
+                                # 直接发送
                                 self.tts_audio_queue.put(
-                                    (SentenceType.MIDDLE, opus_data, text)
+                                    (SentenceType.MIDDLE, opus, text)
                                 )
+                                self.segment_count += 1
+                            else:
+                                # 后续片段缓存
+                                opus_datas_cache.extend(opus)
+                        self.pcm_buffer.clear()
+
+                    # 如果不是前10个片段，发送缓存的数据
+                    if self.segment_count >= 10 and opus_datas_cache:
+                        self.tts_audio_queue.put(
+                            (SentenceType.MIDDLE, opus_datas_cache, text)
+                        )
+
+                    # 如果是最后一段，输出音频获取完毕
+                    if is_last:
+                        self._process_before_stop_play_files()
 
         except Exception as e:
-            logger.error(f"TTS请求异常: {str(e)}")
+            logger.error(f"TTS请求异常: {e}")
             self.tts_audio_queue.put((SentenceType.LAST, [], None))