diff --git a/paddlespeech/server/conf/tts_online_application.yaml b/paddlespeech/server/conf/tts_online_application.yaml
new file mode 100644
index 000000000..a80b3ecec
--- /dev/null
+++ b/paddlespeech/server/conf/tts_online_application.yaml
@@ -0,0 +1,46 @@
+# This is the parameter configuration file for PaddleSpeech Serving.
+
+#################################################################################
+#                             SERVER SETTING                                    #
+#################################################################################
+host: 127.0.0.1
+port: 8092
+
+# The task format in the engin_list is: <speech task>_<engine type>
+# task choices = ['asr_online', 'tts_online']
+# protocol = ['websocket', 'http'] (only one can be selected).
+protocol: 'http'
+engine_list: ['tts_online']
+
+
+#################################################################################
+#                                ENGINE CONFIG                                  #
+#################################################################################
+
+################################### TTS #########################################
+################### speech task: tts; engine_type: online #######################
+tts_online: 
+    # am (acoustic model) choices=['fastspeech2_csmsc']        
+    am: 'fastspeech2_csmsc'   
+    am_config: 
+    am_ckpt: 
+    am_stat: 
+    phones_dict: 
+    tones_dict: 
+    speaker_dict: 
+    spk_id: 0
+
+    # voc (vocoder) choices=['mb_melgan_csmsc']
+    voc: 'mb_melgan_csmsc'
+    voc_config: 
+    voc_ckpt: 
+    voc_stat: 
+
+    # others
+    lang: 'zh'
+    device:  # set 'gpu:id' or 'cpu'
+    am_block: 42
+    am_pad: 12
+    voc_block: 14
+    voc_pad: 14
+    
diff --git a/paddlespeech/server/engine/engine_factory.py b/paddlespeech/server/engine/engine_factory.py
index 2a39fb79b..e147a29a6 100644
--- a/paddlespeech/server/engine/engine_factory.py
+++ b/paddlespeech/server/engine/engine_factory.py
@@ -34,6 +34,9 @@ class EngineFactory(object):
         elif engine_name == 'tts' and engine_type == 'python':
             from paddlespeech.server.engine.tts.python.tts_engine import TTSEngine
             return TTSEngine()
+        elif engine_name == 'tts' and engine_type == 'online':
+            from paddlespeech.server.engine.tts.online.tts_engine import TTSEngine
+            return TTSEngine()
         elif engine_name == 'cls' and engine_type == 'inference':
             from paddlespeech.server.engine.cls.paddleinference.cls_engine import CLSEngine
             return CLSEngine()
diff --git a/paddlespeech/server/engine/tts/online/__init__.py b/paddlespeech/server/engine/tts/online/__init__.py
new file mode 100644
index 000000000..97043fd7b
--- /dev/null
+++ b/paddlespeech/server/engine/tts/online/__init__.py
@@ -0,0 +1,13 @@
+# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
diff --git a/paddlespeech/server/engine/tts/online/tts_engine.py b/paddlespeech/server/engine/tts/online/tts_engine.py
new file mode 100644
index 000000000..2f068b3b9
--- /dev/null
+++ b/paddlespeech/server/engine/tts/online/tts_engine.py
@@ -0,0 +1,305 @@
+# Copyright (c) 2021 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+import base64
+import io
+import time
+
+import librosa
+import numpy as np
+import paddle
+import soundfile as sf
+from scipy.io import wavfile
+
+from paddlespeech.cli.log import logger
+from paddlespeech.cli.tts.infer import TTSExecutor
+from paddlespeech.server.engine.base_engine import BaseEngine
+from paddlespeech.server.utils.audio_process import change_speed
+from paddlespeech.server.utils.errors import ErrorCode
+from paddlespeech.server.utils.exception import ServerBaseException
+from paddlespeech.server.utils.audio_process import float2pcm
+from paddlespeech.server.utils.config import get_config
+from paddlespeech.server.utils.util import denorm
+from paddlespeech.server.utils.util import get_chunks
+
+
+import math
+
+__all__ = ['TTSEngine']
+
+
+class TTSServerExecutor(TTSExecutor):
+    def __init__(self):
+        super().__init__()
+        pass
+
+    @paddle.no_grad()
+    def infer(self,
+              text: str,
+              lang: str='zh',
+              am: str='fastspeech2_csmsc',
+              spk_id: int=0,
+              am_block: int=42,
+              am_pad: int=12,
+              voc_block: int=14,
+              voc_pad: int=14,):
+        """
+        Model inference and result stored in self.output.
+        """
+        am_name = am[:am.rindex('_')]
+        am_dataset = am[am.rindex('_') + 1:]
+        get_tone_ids = False
+        merge_sentences = False
+        frontend_st = time.time()
+        if am_name == 'speedyspeech':
+            get_tone_ids = True
+        if lang == 'zh':
+            input_ids = self.frontend.get_input_ids(
+                text,
+                merge_sentences=merge_sentences,
+                get_tone_ids=get_tone_ids)
+            phone_ids = input_ids["phone_ids"]
+            if get_tone_ids:
+                tone_ids = input_ids["tone_ids"]
+        elif lang == 'en':
+            input_ids = self.frontend.get_input_ids(
+                text, merge_sentences=merge_sentences)
+            phone_ids = input_ids["phone_ids"]
+        else:
+            print("lang should in {'zh', 'en'}!")
+        self.frontend_time = time.time() - frontend_st
+
+        for i in range(len(phone_ids)):
+            am_st = time.time()
+            part_phone_ids = phone_ids[i]
+            # am
+            if am_name == 'speedyspeech':
+                part_tone_ids = tone_ids[i]
+                mel = self.am_inference(part_phone_ids, part_tone_ids)
+            # fastspeech2
+            else:
+                # multi speaker
+                if am_dataset in {"aishell3", "vctk"}:
+                    mel = self.am_inference(
+                        part_phone_ids, spk_id=paddle.to_tensor(spk_id))
+                else:
+                    mel = self.am_inference(part_phone_ids)
+            am_et = time.time()
+            
+            # voc streaming
+            voc_upsample = self.voc_config.n_shift
+            mel_chunks = get_chunks(mel, voc_block, voc_pad, "voc")
+            chunk_num = len(mel_chunks)
+            voc_st = time.time()
+            for i, mel_chunk in enumerate(mel_chunks):
+                sub_wav = self.voc_inference(mel_chunk)
+                front_pad = min(i*voc_block, voc_pad)
+
+                if i == 0:
+                    sub_wav = sub_wav[: voc_block * voc_upsample]
+                elif i == chunk_num - 1:
+                    sub_wav = sub_wav[front_pad * voc_upsample : ]
+                else:
+                    sub_wav = sub_wav[front_pad * voc_upsample: (front_pad + voc_block) * voc_upsample]
+            
+                yield sub_wav
+
+class TTSEngine(BaseEngine):
+    """TTS server engine
+
+    Args:
+        metaclass: Defaults to Singleton.
+    """
+
+    def __init__(self, name=None):
+        """Initialize TTS server engine
+        """
+        super(TTSEngine, self).__init__()
+
+    def init(self, config: dict) -> bool:
+        self.executor = TTSServerExecutor()
+
+        try:
+            self.config = config
+            if self.config.device:
+                self.device = self.config.device
+            else:
+                self.device = paddle.get_device()
+            paddle.set_device(self.device)
+        except Exception as e:
+            logger.error(
+                "Set device failed, please check if device is already used and the parameter 'device' in the yaml file"
+            )
+            logger.error("Initialize TTS server engine Failed on device: %s." %
+                         (self.device))
+            return False
+
+        try:
+            self.executor._init_from_path(
+                am=self.config.am,
+                am_config=self.config.am_config,
+                am_ckpt=self.config.am_ckpt,
+                am_stat=self.config.am_stat,
+                phones_dict=self.config.phones_dict,
+                tones_dict=self.config.tones_dict,
+                speaker_dict=self.config.speaker_dict,
+                voc=self.config.voc,
+                voc_config=self.config.voc_config,
+                voc_ckpt=self.config.voc_ckpt,
+                voc_stat=self.config.voc_stat,
+                lang=self.config.lang)
+        except Exception as e:
+            logger.error("Failed to get model related files.")
+            logger.error("Initialize TTS server engine Failed on device: %s." %
+                         (self.device))
+            return False
+
+        self.am_block = self.config.am_block
+        self.am_pad = self.config.am_pad
+        self.voc_block = self.config.voc_block
+        self.voc_pad = self.config.voc_pad
+
+        logger.info("Initialize TTS server engine successfully on device: %s." %
+                    (self.device))
+        return True
+
+    def preprocess(self, text_bese64: str=None, text_bytes: bytes=None):
+        # Convert byte to text
+        if text_bese64:
+            text_bytes = base64.b64decode(text_bese64)   # base64 to bytes
+        text = text_bytes.decode('utf-8')    # bytes to text
+
+        return text
+
+    def postprocess(self,
+                    wav,
+                    original_fs: int,
+                    target_fs: int=0,
+                    volume: float=1.0,
+                    speed: float=1.0,
+                    audio_path: str=None):
+        """Post-processing operations, including speech, volume, sample rate, save audio file
+
+        Args:
+            wav (numpy(float)): Synthesized audio sample points
+            original_fs (int): original audio sample rate
+            target_fs (int): target audio sample rate
+            volume (float): target volume
+            speed (float): target speed
+
+        Raises:
+            ServerBaseException: Throws an exception if the change speed unsuccessfully.
+
+        Returns:
+            target_fs: target sample rate for synthesized audio.
+            wav_base64: The base64 format of the synthesized audio.
+        """
+
+        # transform sample_rate
+        if target_fs == 0 or target_fs > original_fs:
+            target_fs = original_fs
+            wav_tar_fs = wav
+            logger.info(
+                "The sample rate of synthesized audio is the same as model, which is {}Hz".
+                format(original_fs))
+        else:
+            wav_tar_fs = librosa.resample(
+                np.squeeze(wav), original_fs, target_fs)
+            logger.info(
+                "The sample rate of model is {}Hz and the target sample rate is {}Hz. Converting the sample rate of the synthesized audio successfully.".
+                format(original_fs, target_fs))
+        # transform volume
+        wav_vol = wav_tar_fs * volume
+        logger.info("Transform the volume of the audio successfully.")
+
+        # transform speed
+        try:  # windows not support soxbindings
+            wav_speed = change_speed(wav_vol, speed, target_fs)
+            logger.info("Transform the speed of the audio successfully.")
+        except ServerBaseException:
+            raise ServerBaseException(
+                ErrorCode.SERVER_INTERNAL_ERR,
+                "Failed to transform speed. Can not install soxbindings on your system. \
+                 You need to set speed value 1.0.")
+        except BaseException:
+            logger.error("Failed to transform speed.")
+
+        # wav to base64
+        buf = io.BytesIO()
+        wavfile.write(buf, target_fs, wav_speed)
+        base64_bytes = base64.b64encode(buf.read())
+        wav_base64 = base64_bytes.decode('utf-8')
+        logger.info("Audio to string successfully.")
+
+        # save audio
+        if audio_path is not None:
+            if audio_path.endswith(".wav"):
+                sf.write(audio_path, wav_speed, target_fs)
+            elif audio_path.endswith(".pcm"):
+                wav_norm = wav_speed * (32767 / max(0.001,
+                                                    np.max(np.abs(wav_speed))))
+                with open(audio_path, "wb") as f:
+                    f.write(wav_norm.astype(np.int16))
+            logger.info("Save audio to {} successfully.".format(audio_path))
+        else:
+            logger.info("There is no need to save audio.")
+
+        return target_fs, wav_base64
+
+    def run(self,
+            sentence: str,
+            spk_id: int=0,
+            speed: float=1.0,
+            volume: float=1.0,
+            sample_rate: int=0,
+            save_path: str=None):
+        """ run include inference and postprocess.
+
+        Args:
+            sentence (str): text to be synthesized
+            spk_id (int, optional): speaker id for multi-speaker speech synthesis. Defaults to 0.
+            speed (float, optional): speed. Defaults to 1.0.
+            volume (float, optional): volume. Defaults to 1.0.
+            sample_rate (int, optional): target sample rate for synthesized audio, 
+            0 means the same as the model sampling rate. Defaults to 0.
+            save_path (str, optional): The save path of the synthesized audio. 
+            None means do not save audio. Defaults to None.
+
+        Raises:
+            ServerBaseException: Throws an exception if tts inference unsuccessfully.
+            ServerBaseException: Throws an exception if postprocess unsuccessfully.
+
+        Returns:
+            lang: model language 
+            target_sample_rate: target sample rate for synthesized audio.
+            wav_base64: The base64 format of the synthesized audio.
+        """
+
+        lang = self.config.lang
+        wav_list = []
+
+        for wav in self.executor.infer(text=sentence, lang=lang, am=self.config.am, spk_id=spk_id, am_block=self.am_block, am_pad=self.am_pad, voc_block=self.voc_block, voc_pad=self.voc_pad):
+            # wav type: <class 'numpy.ndarray'>  float32, convert to pcm (base64)
+            wav = float2pcm(wav) # float32 to int16
+            wav_bytes = wav.tobytes()  # to bytes
+            wav_base64 = base64.b64encode(wav_bytes).decode('utf8')  # to base64
+            wav_list.append(wav)
+            
+            yield wav_base64
+        
+        wav_all = np.concatenate(wav_list, axis=0)
+        logger.info("The durations of audio is: {} s".format(len(wav_all)/self.executor.am_config.fs))
+
+
+
+ 
diff --git a/paddlespeech/server/restful/tts_api.py b/paddlespeech/server/restful/tts_api.py
index 4e9bbe23e..d1268428a 100644
--- a/paddlespeech/server/restful/tts_api.py
+++ b/paddlespeech/server/restful/tts_api.py
@@ -15,6 +15,7 @@ import traceback
 from typing import Union
 
 from fastapi import APIRouter
+from fastapi.responses import StreamingResponse
 
 from paddlespeech.cli.log import logger
 from paddlespeech.server.engine.engine_pool import get_engine_pool
@@ -125,3 +126,14 @@ def tts(request_body: TTSRequest):
         traceback.print_exc()
 
     return response
+
+
+@router.post("/paddlespeech/streaming/tts")
+async def stream_tts(request_body: TTSRequest):
+    text = request_body.text
+
+    engine_pool = get_engine_pool()
+    tts_engine = engine_pool['tts']
+    logger.info("Get tts engine successfully.")
+
+    return StreamingResponse(tts_engine.run(sentence=text))
diff --git a/paddlespeech/server/tests/tts/test_client.py b/paddlespeech/server/tests/tts/offline/http_client.py
similarity index 90%
rename from paddlespeech/server/tests/tts/test_client.py
rename to paddlespeech/server/tests/tts/offline/http_client.py
index e42c9bcfa..1bdee4c18 100644
--- a/paddlespeech/server/tests/tts/test_client.py
+++ b/paddlespeech/server/tests/tts/offline/http_client.py
@@ -33,7 +33,8 @@ def tts_client(args):
         text: A sentence to be synthesized
         outfile: Synthetic audio file
     """
-    url = 'http://127.0.0.1:8090/paddlespeech/tts'
+    url = "http://" + str(args.server) + ":" + str(
+        args.port) + "/paddlespeech/tts"
     request = {
         "text": args.text,
         "spk_id": args.spk_id,
@@ -72,7 +73,7 @@ if __name__ == "__main__":
     parser.add_argument(
         '--text',
         type=str,
-        default="你好，欢迎使用语音合成服务",
+        default="您好，欢迎使用语音合成服务。",
         help='A sentence to be synthesized')
     parser.add_argument('--spk_id', type=int, default=0, help='Speaker id')
     parser.add_argument('--speed', type=float, default=1.0, help='Audio speed')
@@ -88,6 +89,9 @@ if __name__ == "__main__":
         type=str,
         default="./out.wav",
         help='Synthesized audio file')
+    parser.add_argument(
+        "--server", type=str, help="server ip", default="127.0.0.1")
+    parser.add_argument("--port", type=int, help="server port", default=8090)
     args = parser.parse_args()
 
     st = time.time()
diff --git a/paddlespeech/server/tests/tts/online/http_client.py b/paddlespeech/server/tests/tts/online/http_client.py
new file mode 100644
index 000000000..cbc1f5c02
--- /dev/null
+++ b/paddlespeech/server/tests/tts/online/http_client.py
@@ -0,0 +1,100 @@
+# Copyright (c) 2021 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+import argparse
+import base64
+import json
+import os
+import time
+
+import requests
+
+from paddlespeech.server.utils.audio_process import pcm2wav
+
+
+def save_audio(buffer, audio_path) -> bool:
+    if args.save_path.endswith("pcm"):
+        with open(args.save_path, "wb") as f:
+            f.write(buffer)
+    elif args.save_path.endswith("wav"):
+        with open("./tmp.pcm", "wb") as f:
+            f.write(buffer)
+        pcm2wav("./tmp.pcm", audio_path, channels=1, bits=16, sample_rate=24000)
+        os.system("rm ./tmp.pcm")
+    else:
+        print("Only supports saved audio format is pcm or wav")
+        return False
+
+    return True
+
+
+def test(args):
+    params = {
+        "text": args.text,
+        "spk_id": args.spk_id,
+        "speed": args.speed,
+        "volume": args.volume,
+        "sample_rate": args.sample_rate,
+        "save_path": ''
+    }
+
+    buffer = b''
+    flag = 1
+    url = "http://" + str(args.server) + ":" + str(
+        args.port) + "/paddlespeech/streaming/tts"
+    st = time.time()
+    html = requests.post(url, json.dumps(params), stream=True)
+    for chunk in html.iter_content(chunk_size=1024):
+        chunk = base64.b64decode(chunk)  # bytes
+        if flag:
+            first_response = time.time() - st
+            print(f"首包响应：{first_response} s")
+            flag = 0
+        buffer += chunk
+
+    final_response = time.time() - st
+    duration = len(buffer) / 2.0 / 24000
+
+    print(f"尾包响应：{final_response} s")
+    print(f"音频时长：{duration} s")
+    print(f"RTF: {final_response / duration}")
+
+    if args.save_path is not None:
+        if save_audio(buffer, args.save_path):
+            print("音频保存至：", args.save_path)
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument(
+        '--text',
+        type=str,
+        default="您好，欢迎使用语音合成服务。",
+        help='A sentence to be synthesized')
+    parser.add_argument('--spk_id', type=int, default=0, help='Speaker id')
+    parser.add_argument('--speed', type=float, default=1.0, help='Audio speed')
+    parser.add_argument(
+        '--volume', type=float, default=1.0, help='Audio volume')
+    parser.add_argument(
+        '--sample_rate',
+        type=int,
+        default=0,
+        help='Sampling rate, the default is the same as the model')
+    parser.add_argument(
+        "--server", type=str, help="server ip", default="127.0.0.1")
+    parser.add_argument("--port", type=int, help="server port", default=8092)
+    parser.add_argument(
+        "--save_path", type=str, help="save audio path", default=None)
+
+    args = parser.parse_args()
+    test(args)
diff --git a/paddlespeech/server/tests/tts/online/http_client_playaudio.py b/paddlespeech/server/tests/tts/online/http_client_playaudio.py
new file mode 100644
index 000000000..1e7e8064e
--- /dev/null
+++ b/paddlespeech/server/tests/tts/online/http_client_playaudio.py
@@ -0,0 +1,112 @@
+# Copyright (c) 2021 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+import argparse
+import base64
+import json
+import threading
+import time
+
+import pyaudio
+import requests
+
+mutex = threading.Lock()
+buffer = b''
+p = pyaudio.PyAudio()
+stream = p.open(
+    format=p.get_format_from_width(2), channels=1, rate=24000, output=True)
+max_fail = 50
+
+
+def play_audio():
+    global stream
+    global buffer
+    global max_fail
+    while True:
+        if not buffer:
+            max_fail -= 1
+            time.sleep(0.05)
+            if max_fail < 0:
+                break
+        mutex.acquire()
+        stream.write(buffer)
+        buffer = b''
+        mutex.release()
+
+
+def test(args):
+    global mutex
+    global buffer
+    params = {
+        "text": args.text,
+        "spk_id": args.spk_id,
+        "speed": args.speed,
+        "volume": args.volume,
+        "sample_rate": args.sample_rate,
+        "save_path": ''
+    }
+
+    all_bytes = 0.0
+    t = threading.Thread(target=play_audio)
+    flag = 1
+    url = "http://" + str(args.server) + ":" + str(
+        args.port) + "/paddlespeech/streaming/tts"
+    st = time.time()
+    html = requests.post(url, json.dumps(params), stream=True)
+    for chunk in html.iter_content(chunk_size=1024):
+        mutex.acquire()
+        chunk = base64.b64decode(chunk)  # bytes
+        buffer += chunk
+        mutex.release()
+        if flag:
+            first_response = time.time() - st
+            print(f"首包响应：{first_response} s")
+            flag = 0
+            t.start()
+        all_bytes += len(chunk)
+
+    final_response = time.time() - st
+    duration = all_bytes / 2 / 24000
+
+    print(f"尾包响应：{final_response} s")
+    print(f"音频时长：{duration} s")
+    print(f"RTF: {final_response / duration}")
+
+    t.join()
+    stream.stop_stream()
+    stream.close()
+    p.terminate()
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument(
+        '--text',
+        type=str,
+        default="您好，欢迎使用语音合成服务。",
+        help='A sentence to be synthesized')
+    parser.add_argument('--spk_id', type=int, default=0, help='Speaker id')
+    parser.add_argument('--speed', type=float, default=1.0, help='Audio speed')
+    parser.add_argument(
+        '--volume', type=float, default=1.0, help='Audio volume')
+    parser.add_argument(
+        '--sample_rate',
+        type=int,
+        default=0,
+        help='Sampling rate, the default is the same as the model')
+    parser.add_argument(
+        "--server", type=str, help="server ip", default="127.0.0.1")
+    parser.add_argument("--port", type=int, help="server port", default=8092)
+
+    args = parser.parse_args()
+    test(args)
diff --git a/paddlespeech/server/tests/tts/online/out.pcm b/paddlespeech/server/tests/tts/online/out.pcm
new file mode 100644
index 000000000..a52377f82
Binary files /dev/null and b/paddlespeech/server/tests/tts/online/out.pcm differ
diff --git a/paddlespeech/server/tests/tts/online/ws_client.py b/paddlespeech/server/tests/tts/online/ws_client.py
new file mode 100644
index 000000000..e0f47b551
--- /dev/null
+++ b/paddlespeech/server/tests/tts/online/ws_client.py
@@ -0,0 +1,126 @@
+# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+import _thread as thread
+import argparse
+import base64
+import json
+import ssl
+import time
+
+import websocket
+
+flag = 1
+st = 0.0
+all_bytes = b''
+
+
+class Ws_Param(object):
+    # 初始化
+    def __init__(self, text, server="127.0.0.1", port=8090):
+        self.server = server
+        self.port = port
+        self.url = "ws://" + self.server + ":" + str(self.port) + "/ws/tts"
+        self.text = text
+
+    # 生成url
+    def create_url(self):
+        return self.url
+
+
+def on_message(ws, message):
+    global flag
+    global st
+    global all_bytes
+
+    try:
+        message = json.loads(message)
+        audio = message["audio"]
+        audio = base64.b64decode(audio)  # bytes
+        status = message["status"]
+        all_bytes += audio
+
+        if status == 0:
+            print("create successfully.")
+        elif status == 1:
+            if flag:
+                print(f"首包响应：{time.time() - st} s")
+                flag = 0
+        elif status == 2:
+            final_response = time.time() - st
+            duration = len(all_bytes) / 2.0 / 24000
+            print(f"尾包响应：{final_response} s")
+            print(f"音频时长：{duration} s")
+            print(f"RTF: {final_response / duration}")
+            with open("./out.pcm", "wb") as f:
+                f.write(all_bytes)
+            print("ws is closed")
+            ws.close()
+        else:
+            print("infer error")
+
+    except Exception as e:
+        print("receive msg,but parse exception:", e)
+
+
+# 收到websocket错误的处理
+def on_error(ws, error):
+    print("### error:", error)
+
+
+# 收到websocket关闭的处理
+def on_close(ws):
+    print("### closed ###")
+
+
+# 收到websocket连接建立的处理
+def on_open(ws):
+    def run(*args):
+        global st
+        text_base64 = str(
+            base64.b64encode((wsParam.text).encode('utf-8')), "UTF8")
+        d = {"text": text_base64}
+        d = json.dumps(d)
+        print("Start sending text data")
+        st = time.time()
+        ws.send(d)
+
+    thread.start_new_thread(run, ())
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument(
+        "--text",
+        type=str,
+        help="A sentence to be synthesized",
+        default="您好，欢迎使用语音合成服务。")
+    parser.add_argument(
+        "--server", type=str, help="server ip", default="127.0.0.1")
+    parser.add_argument("--port", type=int, help="server port", default=8092)
+    args = parser.parse_args()
+
+    print("***************************************")
+    print("Server ip: ", args.server)
+    print("Server port: ", args.port)
+    print("Sentence to be synthesized: ", args.text)
+    print("***************************************")
+
+    wsParam = Ws_Param(text=args.text, server=args.server, port=args.port)
+
+    websocket.enableTrace(False)
+    wsUrl = wsParam.create_url()
+    ws = websocket.WebSocketApp(
+        wsUrl, on_message=on_message, on_error=on_error, on_close=on_close)
+    ws.on_open = on_open
+    ws.run_forever(sslopt={"cert_reqs": ssl.CERT_NONE})
diff --git a/paddlespeech/server/tests/tts/online/ws_client_playaudio.py b/paddlespeech/server/tests/tts/online/ws_client_playaudio.py
new file mode 100644
index 000000000..4e1c538d1
--- /dev/null
+++ b/paddlespeech/server/tests/tts/online/ws_client_playaudio.py
@@ -0,0 +1,160 @@
+# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+import _thread as thread
+import argparse
+import base64
+import json
+import ssl
+import threading
+import time
+
+import pyaudio
+import websocket
+
+mutex = threading.Lock()
+buffer = b''
+p = pyaudio.PyAudio()
+stream = p.open(
+    format=p.get_format_from_width(2), channels=1, rate=24000, output=True)
+flag = 1
+st = 0.0
+all_bytes = 0.0
+
+
+class Ws_Param(object):
+    # 初始化
+    def __init__(self, text, server="127.0.0.1", port=8090):
+        self.server = server
+        self.port = port
+        self.url = "ws://" + self.server + ":" + str(self.port) + "/ws/tts"
+        self.text = text
+
+    # 生成url
+    def create_url(self):
+        return self.url
+
+
+def play_audio():
+    global stream
+    global buffer
+    while True:
+        time.sleep(0.05)
+        if not buffer:  # buffer 为空
+            break
+        mutex.acquire()
+        stream.write(buffer)
+        buffer = b''
+        mutex.release()
+
+
+t = threading.Thread(target=play_audio)
+
+
+def on_message(ws, message):
+    global flag
+    global t
+    global buffer
+    global st
+    global all_bytes
+
+    try:
+        message = json.loads(message)
+        audio = message["audio"]
+        audio = base64.b64decode(audio)  # bytes
+        status = message["status"]
+        all_bytes += len(audio)
+
+        if status == 0:
+            print("create successfully.")
+        elif status == 1:
+            mutex.acquire()
+            buffer += audio
+            mutex.release()
+            if flag:
+                print(f"首包响应：{time.time() - st} s")
+                flag = 0
+                print("Start playing audio")
+                t.start()
+        elif status == 2:
+            final_response = time.time() - st
+            duration = all_bytes / 2 / 24000
+            print(f"尾包响应：{final_response} s")
+            print(f"音频时长：{duration} s")
+            print(f"RTF: {final_response / duration}")
+            print("ws is closed")
+            ws.close()
+        else:
+            print("infer error")
+
+    except Exception as e:
+        print("receive msg,but parse exception:", e)
+
+
+# 收到websocket错误的处理
+def on_error(ws, error):
+    print("### error:", error)
+
+
+# 收到websocket关闭的处理
+def on_close(ws):
+    print("### closed ###")
+
+
+# 收到websocket连接建立的处理
+def on_open(ws):
+    def run(*args):
+        global st
+        text_base64 = str(
+            base64.b64encode((wsParam.text).encode('utf-8')), "UTF8")
+        d = {"text": text_base64}
+        d = json.dumps(d)
+        print("Start sending text data")
+        st = time.time()
+        ws.send(d)
+
+    thread.start_new_thread(run, ())
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument(
+        "--text",
+        type=str,
+        help="A sentence to be synthesized",
+        default="您好，欢迎使用语音合成服务。")
+    parser.add_argument(
+        "--server", type=str, help="server ip", default="127.0.0.1")
+    parser.add_argument("--port", type=int, help="server port", default=8092)
+    args = parser.parse_args()
+
+    print("***************************************")
+    print("Server ip: ", args.server)
+    print("Server port: ", args.port)
+    print("Sentence to be synthesized: ", args.text)
+    print("***************************************")
+
+    wsParam = Ws_Param(text=args.text, server=args.server, port=args.port)
+
+    websocket.enableTrace(False)
+    wsUrl = wsParam.create_url()
+    ws = websocket.WebSocketApp(
+        wsUrl, on_message=on_message, on_error=on_error, on_close=on_close)
+    ws.on_open = on_open
+    ws.run_forever(sslopt={"cert_reqs": ssl.CERT_NONE})
+
+    t.join()
+    print("End of playing audio")
+    stream.stop_stream()
+    stream.close()
+    p.terminate()
diff --git a/paddlespeech/server/utils/audio_process.py b/paddlespeech/server/utils/audio_process.py
index 3cbb495a6..1d4b158c5 100644
--- a/paddlespeech/server/utils/audio_process.py
+++ b/paddlespeech/server/utils/audio_process.py
@@ -103,3 +103,26 @@ def change_speed(sample_raw, speed_rate, sample_rate):
         sample_rate_in=sample_rate).squeeze(-1).astype(np.float32).copy()
 
     return sample_speed
+
+
+def float2pcm(sig, dtype='int16'):
+    """Convert floating point signal with a range from -1 to 1 to PCM.
+
+    Args:
+        sig (array): Input array, must have floating point type.
+        dtype (str, optional): Desired (integer) data type. Defaults to 'int16'.
+
+    Returns:
+        numpy.ndarray: Integer data, scaled and clipped to the range of the given
+    """
+    sig = np.asarray(sig)
+    if sig.dtype.kind != 'f':
+        raise TypeError("'sig' must be a float array")
+    dtype = np.dtype(dtype)
+    if dtype.kind not in 'iu':
+        raise TypeError("'dtype' must be an integer type")
+
+    i = np.iinfo(dtype)
+    abs_max = 2**(i.bits - 1)
+    offset = i.min + abs_max
+    return (sig * abs_max + offset).clip(i.min, i.max).astype(dtype)
diff --git a/paddlespeech/server/utils/util.py b/paddlespeech/server/utils/util.py
index e9104fa2d..c35939b74 100644
--- a/paddlespeech/server/utils/util.py
+++ b/paddlespeech/server/utils/util.py
@@ -11,6 +11,7 @@
 # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 # See the License for the 
 import base64
+import math
 
 
 def wav2base64(wav_file: str):
@@ -31,3 +32,29 @@ def self_check():
     """ self check resource
     """
     return True
+
+
+def denorm(data, mean, std):
+    return data * std + mean
+
+
+def get_chunks(data, block_size, pad_size, step):
+    if step == "am":
+        data_len = data.shape[1]
+    elif step == "voc":
+        data_len = data.shape[0]
+    else:
+        print("Please set correct type to get chunks, am or voc")
+
+    chunks = []
+    n = math.ceil(data_len / block_size)
+    for i in range(n):
+        start = max(0, i * block_size - pad_size)
+        end = min((i + 1) * block_size + pad_size, data_len)
+        if step == "am":
+            chunks.append(data[:, start:end, :])
+        elif step == "voc":
+            chunks.append(data[start:end, :])
+        else:
+            print("Please set correct type to get chunks, am or voc")
+    return chunks
diff --git a/paddlespeech/server/ws/api.py b/paddlespeech/server/ws/api.py
index 10664d114..313fd16f5 100644
--- a/paddlespeech/server/ws/api.py
+++ b/paddlespeech/server/ws/api.py
@@ -16,6 +16,7 @@ from typing import List
 from fastapi import APIRouter
 
 from paddlespeech.server.ws.asr_socket import router as asr_router
+from paddlespeech.server.ws.tts_socket import router as tts_router
 
 _router = APIRouter()
 
@@ -31,7 +32,7 @@ def setup_router(api_list: List):
         if api_name == 'asr':
             _router.include_router(asr_router)
         elif api_name == 'tts':
-            pass
+            _router.include_router(tts_router)
         else:
             pass
 
diff --git a/paddlespeech/server/ws/tts_socket.py b/paddlespeech/server/ws/tts_socket.py
new file mode 100644
index 000000000..4df2850af
--- /dev/null
+++ b/paddlespeech/server/ws/tts_socket.py
@@ -0,0 +1,62 @@
+# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+import json
+
+from fastapi import APIRouter
+from fastapi import WebSocket
+from fastapi import WebSocketDisconnect
+from starlette.websockets import WebSocketState as WebSocketState
+
+from paddlespeech.cli.log import logger
+from paddlespeech.server.engine.engine_pool import get_engine_pool
+
+router = APIRouter()
+
+
+@router.websocket('/ws/tts')
+async def websocket_endpoint(websocket: WebSocket):
+    await websocket.accept()
+
+    try:
+        # careful here, changed the source code from starlette.websockets
+        assert websocket.application_state == WebSocketState.CONNECTED
+        message = await websocket.receive()
+        websocket._raise_on_disconnect(message)
+
+        # get engine
+        engine_pool = get_engine_pool()
+        tts_engine = engine_pool['tts']
+
+        # 获取 message 并转文本
+        message = json.loads(message["text"])
+        text_bese64 = message["text"]
+        sentence = tts_engine.preprocess(text_bese64=text_bese64)
+
+        # run
+        wav = tts_engine.run(sentence)
+
+        while True:
+            try:
+                tts_results = next(wav)
+                resp = {"status": 1, "audio": tts_results}
+                await websocket.send_json(resp)
+                logger.info("streaming audio...")
+            except StopIteration as e:
+                resp = {"status": 2, "audio": ''}
+                await websocket.send_json(resp)
+                logger.info("Complete the transmission of audio streams")
+                break
+
+    except WebSocketDisconnect:
+        pass