Merge pull request #1882 from lym0302/streaming_tts_server

[server] improve code

paddlespeech/server/bin/paddlespeech_client.py

@@ -18,6 +18,7 @@ import io
 import json
 import os
 import random
+import sys
 import time
 from typing import List
 
@@ -91,7 +92,7 @@ class TTSClientExecutor(BaseExecutor):
             temp_wav = str(random.getrandbits(128)) + ".wav"
             soundfile.write(temp_wav, samples, sample_rate)
             wav2pcm(temp_wav, outfile, data_type=np.int16)
-            os.system("rm %s" % (temp_wav))
+            os.remove(temp_wav)
         else:
             logger.error("The format for saving audio only supports wav or pcm")
 
@@ -128,6 +129,7 @@ class TTSClientExecutor(BaseExecutor):
             return True
         except Exception as e:
             logger.error("Failed to synthesized audio.")
+            logger.error(e)
             return False
 
     @stats_wrapper
@@ -236,6 +238,7 @@ class TTSOnlineClientExecutor(BaseExecutor):
             return True
         except Exception as e:
             logger.error("Failed to synthesized audio.")
+            logger.error(e)
             return False
 
     @stats_wrapper
@@ -275,7 +278,7 @@ class TTSOnlineClientExecutor(BaseExecutor):
 
         else:
             logger.error("Please set correct protocol, http or websocket")
-            return False
+            sys.exit(-1)
 
         logger.info(f"sentence: {input}")
         logger.info(f"duration: {duration} s")
@@ -503,6 +506,7 @@ class ASROnlineClientExecutor(BaseExecutor):
         Returns:
             str: the audio text
         """
+
         logger.info("asr websocket client start")
         handler = ASRWsAudioHandler(
             server_ip,
@@ -555,6 +559,7 @@ class CLSClientExecutor(BaseExecutor):
             return True
         except Exception as e:
             logger.error("Failed to speech classification.")
+            logger.error(e)
             return False
 
     @stats_wrapper
@@ -728,6 +733,7 @@ class VectorClientExecutor(BaseExecutor):
         Returns:
             str: the audio embedding or score between enroll and test audio
         """
+
         if task == "spk":
             from paddlespeech.server.utils.audio_handler import VectorHttpHandler
             logger.info("vector http client start")

paddlespeech/server/bin/paddlespeech_server.py

@@ -12,6 +12,7 @@
 # See the License for the specific language governing permissions and
 # limitations under the License.
 import argparse
+import sys
 from typing import List
 
 import uvicorn
@@ -79,10 +80,12 @@ class ServerExecutor(BaseExecutor):
 
     def execute(self, argv: List[str]) -> bool:
         args = self.parser.parse_args(argv)
-        config = get_config(args.config_file)
-
-        if self.init(config):
-            uvicorn.run(app, host=config.host, port=config.port, debug=True)
+        try:
+            self(args.config_file, args.log_file)
+        except Exception as e:
+            logger.error("Failed to start server.")
+            logger.error(e)
+            sys.exit(-1)
 
     @stats_wrapper
     def __call__(self,

paddlespeech/server/tests/tts/offline/http_client.py

@@ -61,7 +61,7 @@ def tts_client(args):
         temp_wav = str(random.getrandbits(128)) + ".wav"
         soundfile.write(temp_wav, samples, sample_rate)
         wav2pcm(temp_wav, outfile, data_type=np.int16)
-        os.system("rm %s" % (temp_wav))
+        os.remove(temp_wav)
     else:
         print("The format for saving audio only supports wav or pcm")
 

paddlespeech/server/utils/audio_handler.py

@@ -304,52 +304,80 @@ class TTSWsHandler:
         receive_time_list = []
         chunk_duration_list = []
 
-        # 1. Send websocket handshake protocal
+        # 1. Send websocket handshake request
         async with websockets.connect(self.url) as ws:
-            # 2. Server has already received handshake protocal
-            # send text to engine
+            # 2. Server has already received handshake response, send start request
+            start_request = json.dumps({"task": "tts", "signal": "start"})
+            await ws.send(start_request)
+            msg = await ws.recv()
+            logger.info(f"client receive msg={msg}")
+            msg = json.loads(msg)
+            session = msg["session"]
+
+            # 3. send speech synthesis request 
             text_base64 = str(base64.b64encode((text).encode('utf-8')), "UTF8")
-            d = {"text": text_base64}
-            d = json.dumps(d)
+            request = json.dumps({"text": text_base64})
             st = time.time()
-            await ws.send(d)
+            await ws.send(request)
             logging.info("send a message to the server")
 
-            # 3. Process the received response 
+            # 4. Process the received response
             message = await ws.recv()
             first_response = time.time() - st
             message = json.loads(message)
             status = message["status"]
+            while True:
+                # When throw an exception
+                if status == -1:
+                    # send end request
+                    end_request = json.dumps({
+                        "task": "tts",
+                        "signal": "end",
+                        "session": session
+                    })
+                    await ws.send(end_request)
+                    break
+
+                # Rerutn last packet normally, no audio information
+                elif status == 2:
+                    final_response = time.time() - st
+                    duration = len(all_bytes) / 2.0 / 24000
+
+                    if output is not None:
+                        save_audio_success = save_audio(all_bytes, output)
+                    else:
+                        save_audio_success = False
+
+                    # send end request
+                    end_request = json.dumps({
+                        "task": "tts",
+                        "signal": "end",
+                        "session": session
+                    })
+                    await ws.send(end_request)
+                    break
+
+                # Return the audio stream normally
+                elif status == 1:
+                    receive_time_list.append(time.time())
+                    audio = message["audio"]
+                    audio = base64.b64decode(audio)  # bytes
+                    chunk_duration_list.append(len(audio) / 2.0 / 24000)
+                    all_bytes += audio
+                    if self.play:
+                        self.mutex.acquire()
+                        self.buffer += audio
+                        self.mutex.release()
+                        if self.start_play:
+                            self.t.start()
+                            self.start_play = False
+
+                    message = await ws.recv()
+                    message = json.loads(message)
+                    status = message["status"]
 
-            while (status == 1):
-                receive_time_list.append(time.time())
-                audio = message["audio"]
-                audio = base64.b64decode(audio)  # bytes
-                chunk_duration_list.append(len(audio) / 2.0 / 24000)
-                all_bytes += audio
-                if self.play:
-                    self.mutex.acquire()
-                    self.buffer += audio
-                    self.mutex.release()
-                    if self.start_play:
-                        self.t.start()
-                        self.start_play = False
-
-                message = await ws.recv()
-                message = json.loads(message)
-                status = message["status"]
-
-            # 4. Last packet, no audio information
-            if status == 2:
-                final_response = time.time() - st
-                duration = len(all_bytes) / 2.0 / 24000
-
-                if output is not None:
-                    save_audio_success = save_audio(all_bytes, output)
                 else:
-                    save_audio_success = False
-            else:
-                logger.error("infer error")
+                    logger.error("infer error, return status is invalid.")
 
             if self.play:
                 self.t.join()
@@ -458,6 +486,7 @@ class TTSHttpHandler:
 
         final_response = time.time() - st
         duration = len(all_bytes) / 2.0 / 24000
+        html.close()  # when stream=True
 
         if output is not None:
             save_audio_success = save_audio(all_bytes, output)

paddlespeech/server/utils/audio_process.py

@@ -167,7 +167,7 @@ def save_audio(bytes_data, audio_path, sample_rate: int=24000) -> bool:
             channels=1,
             bits=16,
             sample_rate=sample_rate)
-        os.system("rm ./tmp.pcm")
+        os.remove("./tmp.pcm")
     else:
         print("Only supports saved audio format is pcm or wav")
         return False

paddlespeech/server/ws/tts_api.py

@@ -12,10 +12,10 @@
 # See the License for the specific language governing permissions and
 # limitations under the License.
 import json
+import uuid
 
 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
@@ -26,36 +26,79 @@ router = APIRouter()
 
 @router.websocket('/paddlespeech/tts/streaming')
 async def websocket_endpoint(websocket: WebSocket):
+    """PaddleSpeech Online TTS Server api
+
+    Args:
+        websocket (WebSocket): the websocket instance
+    """
+
+    #1. the interface wait to accept the websocket protocal header
+    #   and only we receive the header, it establish the connection with specific thread
     await websocket.accept()
 
+    #2. if we accept the websocket headers, we will get the online tts engine instance
+    engine_pool = get_engine_pool()
+    tts_engine = engine_pool['tts']
+
     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)
+        while True:
+            # careful here, changed the source code from starlette.websockets
+            assert websocket.application_state == WebSocketState.CONNECTED
+            message = await websocket.receive()
+            websocket._raise_on_disconnect(message)
+            message = json.loads(message["text"])
 
-        # get engine
-        engine_pool = get_engine_pool()
-        tts_engine = engine_pool['tts']
+            if 'signal' in message:
+                # start request
+                if message['signal'] == 'start':
+                    session = uuid.uuid1().hex
+                    resp = {
+                        "status": 0,
+                        "signal": "server ready",
+                        "session": session
+                    }
+                    await websocket.send_json(resp)
 
-        # 获取 message 并转文本
-        message = json.loads(message["text"])
-        text_bese64 = message["text"]
-        sentence = tts_engine.preprocess(text_bese64=text_bese64)
+                # end request
+                elif message['signal'] == 'end':
+                    resp = {
+                        "status": 0,
+                        "signal": "connection will be closed",
+                        "session": session
+                    }
+                    await websocket.send_json(resp)
+                    break
+                else:
+                    resp = {"status": 0, "signal": "no valid json data"}
+                    await websocket.send_json(resp)
 
-        # run
-        wav_generator = tts_engine.run(sentence)
+            # speech synthesis request 
+            elif 'text' in message:
+                text_bese64 = message["text"]
+                sentence = tts_engine.preprocess(text_bese64=text_bese64)
 
-        while True:
-            try:
-                tts_results = next(wav_generator)
-                resp = {"status": 1, "audio": tts_results}
-                await websocket.send_json(resp)
-            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
\ No newline at end of file
+                # run
+                wav_generator = tts_engine.run(sentence)
+
+                while True:
+                    try:
+                        tts_results = next(wav_generator)
+                        resp = {"status": 1, "audio": tts_results}
+                        await websocket.send_json(resp)
+                    except StopIteration as e:
+                        resp = {"status": 2, "audio": ''}
+                        await websocket.send_json(resp)
+                        logger.info(
+                            "Complete the synthesis of the audio streams")
+                        break
+                    except Exception as e:
+                        resp = {"status": -1, "audio": ''}
+                        await websocket.send_json(resp)
+                        break
+
+            else:
+                logger.error(
+                    "Invalid request, please check if the request is correct.")
+
+    except Exception as e:
+        logger.error(e)

tests/unit/server/offline/change_yaml.py

 #!/usr/bin/python
 import argparse
 import os
+import shutil
 
 import yaml
 
@@ -14,7 +15,7 @@ def change_device(yamlfile: str, engine: str, device: str):
         model_type (dict): change model type
     """
     tmp_yamlfile = yamlfile.split(".yaml")[0] + "_tmp.yaml"
-    os.system("cp %s %s" % (yamlfile, tmp_yamlfile))
+    shutil.copyfile(yamlfile, tmp_yamlfile)
 
     if device == 'cpu':
         set_device = 'cpu'
@@ -41,7 +42,7 @@ def change_device(yamlfile: str, engine: str, device: str):
 
         print(yaml.dump(y, default_flow_style=False, sort_keys=False))
         yaml.dump(y, fw, allow_unicode=True)
-    os.system("rm %s" % (tmp_yamlfile))
+    os.remove(tmp_yamlfile)
     print("Change %s successfully." % (yamlfile))
 
 
@@ -52,7 +53,7 @@ def change_engine_type(yamlfile: str, engine_type):
         task (str):  asr or tts
     """
     tmp_yamlfile = yamlfile.split(".yaml")[0] + "_tmp.yaml"
-    os.system("cp %s %s" % (yamlfile, tmp_yamlfile))
+    shutil.copyfile(yamlfile, tmp_yamlfile)
     speech_task = engine_type.split("_")[0]
 
     with open(tmp_yamlfile) as f, open(yamlfile, "w+", encoding="utf-8") as fw:
@@ -65,7 +66,7 @@ def change_engine_type(yamlfile: str, engine_type):
         y['engine_list'] = engine_list
         print(yaml.dump(y, default_flow_style=False, sort_keys=False))
         yaml.dump(y, fw, allow_unicode=True)
-    os.system("rm %s" % (tmp_yamlfile))
+    os.remove(tmp_yamlfile)
     print("Change %s successfully." % (yamlfile))
 
 

tests/unit/server/online/tts/check_server/change_yaml.py

 #!/usr/bin/python
 import argparse
 import os
+import shutil
 
 import yaml
 
@@ -13,7 +14,7 @@ def change_value(args):
     target_value = args.target_value
 
     tmp_yamlfile = yamlfile.split(".yaml")[0] + "_tmp.yaml"
-    os.system("cp %s %s" % (yamlfile, tmp_yamlfile))
+    shutil.copyfile(yamlfile, tmp_yamlfile)
 
     with open(tmp_yamlfile) as f, open(yamlfile, "w+", encoding="utf-8") as fw:
         y = yaml.safe_load(f)
@@ -51,7 +52,7 @@ def change_value(args):
 
         print(yaml.dump(y, default_flow_style=False, sort_keys=False))
         yaml.dump(y, fw, allow_unicode=True)
-    os.system("rm %s" % (tmp_yamlfile))
+    os.remove(tmp_yamlfile)
     print(f"Change key: {target_key} to value: {target_value} successfully.")
 
 

tests/unit/server/online/tts/test_server/test_http_client.py

@@ -75,8 +75,8 @@ if __name__ == "__main__":
 
     args = parser.parse_args()
 
-    os.system("rm -rf %s" % (args.output_dir))
-    os.mkdir(args.output_dir)
+    if not os.path.exists(args.output_dir):
+        os.makedirs(args.output_dir)
 
     first_response_list = []
     final_response_list = []