From a2af08c32d96b136d3d91d28a6da0ba6ea52e00f Mon Sep 17 00:00:00 2001
From: 游雁 <zhifu.gzf@alibaba-inc.com>
Date: 星期四, 15 六月 2023 17:10:12 +0800
Subject: [PATCH] Merge branch 'main' of github.com:alibaba-damo-academy/FunASR add
---
funasr/runtime/websocket/CMakeLists.txt | 13
/dev/null | 277 ----------------
funasr/runtime/websocket/funasr-ws-client.cpp | 366 +++++++++++++++++++++
funasr/runtime/websocket/websocket-server.cpp | 30
funasr/runtime/python/websocket/wss_client_asr.py | 273 ++++++++-------
funasr/runtime/websocket/websocket-server.h | 6
funasr/runtime/websocket/readme.md | 31 +
tests/test_asr_inference_pipeline.py | 6
tests/test_asr_vad_punc_inference_pipeline.py | 1
funasr/runtime/websocket/funasr-ws-server.cpp | 6
10 files changed, 571 insertions(+), 438 deletions(-)
diff --git a/funasr/runtime/python/websocket/wss_client_asr.py b/funasr/runtime/python/websocket/wss_client_asr.py
index 0dd236d..2ea8a16 100644
--- a/funasr/runtime/python/websocket/wss_client_asr.py
+++ b/funasr/runtime/python/websocket/wss_client_asr.py
@@ -1,7 +1,7 @@
# -*- encoding: utf-8 -*-
import os
import time
-import websockets,ssl
+import websockets, ssl
import asyncio
# import threading
import argparse
@@ -12,6 +12,7 @@
import logging
+SUPPORT_AUDIO_TYPE_SETS = ['.wav', '.pcm']
logging.basicConfig(level=logging.ERROR)
parser = argparse.ArgumentParser()
@@ -53,7 +54,7 @@
type=str,
default=None,
help="output_dir")
-
+
parser.add_argument("--ssl",
type=int,
default=1,
@@ -68,22 +69,25 @@
print(args)
# voices = asyncio.Queue()
from queue import Queue
-voices = Queue()
+voices = Queue()
+offline_msg_done=False
+
ibest_writer = None
if args.output_dir is not None:
writer = DatadirWriter(args.output_dir)
ibest_writer = writer[f"1best_recog"]
+
async def record_microphone():
is_finished = False
import pyaudio
- #print("2")
- global voices
+ # print("2")
+ global voices
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = 16000
- chunk_size = 60*args.chunk_size[1]/args.chunk_interval
+ chunk_size = 60 * args.chunk_size[1] / args.chunk_interval
CHUNK = int(RATE / 1000 * chunk_size)
p = pyaudio.PyAudio()
@@ -94,19 +98,16 @@
input=True,
frames_per_buffer=CHUNK)
- message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval, "wav_name": "microphone", "is_speaking": True})
+ message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval,
+ "wav_name": "microphone", "is_speaking": True})
voices.put(message)
while True:
-
data = stream.read(CHUNK)
- message = data
-
+ message = data
voices.put(message)
-
await asyncio.sleep(0.005)
-async def record_from_scp(chunk_begin,chunk_size):
- import wave
+async def record_from_scp(chunk_begin, chunk_size):
global voices
is_finished = False
if args.audio_in.endswith(".scp"):
@@ -114,91 +115,98 @@
wavs = f_scp.readlines()
else:
wavs = [args.audio_in]
- if chunk_size>0:
- wavs=wavs[chunk_begin:chunk_begin+chunk_size]
+ if chunk_size > 0:
+ wavs = wavs[chunk_begin:chunk_begin + chunk_size]
for wav in wavs:
wav_splits = wav.strip().split()
+
wav_name = wav_splits[0] if len(wav_splits) > 1 else "demo"
wav_path = wav_splits[1] if len(wav_splits) > 1 else wav_splits[0]
-
- # bytes_f = open(wav_path, "rb")
- # bytes_data = bytes_f.read()
- with wave.open(wav_path, "rb") as wav_file:
- params = wav_file.getparams()
- # header_length = wav_file.getheaders()[0][1]
- # wav_file.setpos(header_length)
- frames = wav_file.readframes(wav_file.getnframes())
+ if not len(wav_path.strip())>0:
+ continue
+ if wav_path.endswith(".pcm"):
+ with open(wav_path, "rb") as f:
+ audio_bytes = f.read()
+ elif wav_path.endswith(".wav"):
+ import wave
+ with wave.open(wav_path, "rb") as wav_file:
+ params = wav_file.getparams()
+ frames = wav_file.readframes(wav_file.getnframes())
+ audio_bytes = bytes(frames)
+ else:
+ raise NotImplementedError(
+ f'Not supported audio type')
- audio_bytes = bytes(frames)
# stride = int(args.chunk_size/1000*16000*2)
- stride = int(60*args.chunk_size[1]/args.chunk_interval/1000*16000*2)
- chunk_num = (len(audio_bytes)-1)//stride + 1
+ stride = int(60 * args.chunk_size[1] / args.chunk_interval / 1000 * 16000 * 2)
+ chunk_num = (len(audio_bytes) - 1) // stride + 1
# print(stride)
-
+
# send first time
- message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval, "wav_name": wav_name,"is_speaking": True})
- voices.put(message)
+ message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval,
+ "wav_name": wav_name, "is_speaking": True})
+ #voices.put(message)
+ await websocket.send(message)
is_speaking = True
for i in range(chunk_num):
- beg = i*stride
- data = audio_bytes[beg:beg+stride]
- message = data
- voices.put(message)
- if i == chunk_num-1:
+ beg = i * stride
+ data = audio_bytes[beg:beg + stride]
+ message = data
+ #voices.put(message)
+ await websocket.send(message)
+ if i == chunk_num - 1:
is_speaking = False
message = json.dumps({"is_speaking": is_speaking})
- voices.put(message)
- # print("data_chunk: ", len(data_chunk))
- # print(voices.qsize())
- sleep_duration = 0.001 if args.send_without_sleep else 60*args.chunk_size[1]/args.chunk_interval/1000
+ #voices.put(message)
+ await websocket.send(message)
+
+ sleep_duration = 0.001 if args.send_without_sleep else 60 * args.chunk_size[1] / args.chunk_interval / 1000
await asyncio.sleep(sleep_duration)
+ # when all data sent, we need to close websocket
+ while not voices.empty():
+ await asyncio.sleep(1)
+ await asyncio.sleep(3)
+ # offline model need to wait for message recved
+
+ if args.mode=="offline":
+ global offline_msg_done
+ while not offline_msg_done:
+ await asyncio.sleep(1)
+
+ await websocket.close()
+
+
+
-
-async def ws_send():
- global voices
- global websocket
- print("started to sending data!")
- while True:
- while not voices.empty():
- data = voices.get()
- voices.task_done()
- try:
- await websocket.send(data)
- except Exception as e:
- print('Exception occurred:', e)
- traceback.print_exc()
- exit(0)
- await asyncio.sleep(0.005)
- await asyncio.sleep(0.005)
-
-
-
+
+
async def message(id):
- global websocket
+ global websocket,voices,offline_msg_done
text_print = ""
text_print_2pass_online = ""
text_print_2pass_offline = ""
- while True:
- try:
+ try:
+ while True:
+
meg = await websocket.recv()
meg = json.loads(meg)
wav_name = meg.get("wav_name", "demo")
- # print(wav_name)
text = meg["text"]
if ibest_writer is not None:
ibest_writer["text"][wav_name] = text
-
+
if meg["mode"] == "online":
text_print += "{}".format(text)
text_print = text_print[-args.words_max_print:]
os.system('clear')
- print("\rpid"+str(id)+": "+text_print)
+ print("\rpid" + str(id) + ": " + text_print)
elif meg["mode"] == "offline":
text_print += "{}".format(text)
text_print = text_print[-args.words_max_print:]
os.system('clear')
- print("\rpid"+str(id)+": "+text_print)
+ print("\rpid" + str(id) + ": " + text_print)
+ offline_msg_done=True
else:
if meg["mode"] == "2pass-online":
text_print_2pass_online += "{}".format(text)
@@ -211,10 +219,12 @@
os.system('clear')
print("\rpid" + str(id) + ": " + text_print)
- except Exception as e:
+ except Exception as e:
print("Exception:", e)
- traceback.print_exc()
- exit(0)
+ #traceback.print_exc()
+ #await websocket.close()
+
+
async def print_messge():
global websocket
@@ -225,72 +235,87 @@
print(meg)
except Exception as e:
print("Exception:", e)
- traceback.print_exc()
+ #traceback.print_exc()
exit(0)
-async def ws_client(id,chunk_begin,chunk_size):
- global websocket
- if args.ssl==1:
- ssl_context = ssl.SSLContext()
- ssl_context.check_hostname = False
- ssl_context.verify_mode = ssl.CERT_NONE
- uri = "wss://{}:{}".format(args.host, args.port)
+async def ws_client(id, chunk_begin, chunk_size):
+ if args.audio_in is None:
+ chunk_begin=0
+ chunk_size=1
+ global websocket,voices,offline_msg_done
+
+ for i in range(chunk_begin,chunk_begin+chunk_size):
+ offline_msg_done=False
+ voices = Queue()
+ if args.ssl == 1:
+ ssl_context = ssl.SSLContext()
+ ssl_context.check_hostname = False
+ ssl_context.verify_mode = ssl.CERT_NONE
+ uri = "wss://{}:{}".format(args.host, args.port)
else:
- uri = "ws://{}:{}".format(args.host, args.port)
- ssl_context=None
- print("connect to",uri)
- async for websocket in websockets.connect(uri, subprotocols=["binary"], ping_interval=None,ssl=ssl_context):
+ uri = "ws://{}:{}".format(args.host, args.port)
+ ssl_context = None
+ print("connect to", uri)
+ async with websockets.connect(uri, subprotocols=["binary"], ping_interval=None, ssl=ssl_context) as websocket:
if args.audio_in is not None:
- task = asyncio.create_task(record_from_scp(chunk_begin,chunk_size))
+ task = asyncio.create_task(record_from_scp(i, 1))
else:
task = asyncio.create_task(record_microphone())
- task2 = asyncio.create_task(ws_send())
- task3 = asyncio.create_task(message(id))
- await asyncio.gather(task, task2, task3)
+ #task2 = asyncio.create_task(ws_send())
+ task3 = asyncio.create_task(message(str(id)+"_"+str(i))) #processid+fileid
+ await asyncio.gather(task, task3)
+ exit(0)
+
-def one_thread(id,chunk_begin,chunk_size):
- asyncio.get_event_loop().run_until_complete(ws_client(id,chunk_begin,chunk_size))
- asyncio.get_event_loop().run_forever()
-
+def one_thread(id, chunk_begin, chunk_size):
+ asyncio.get_event_loop().run_until_complete(ws_client(id, chunk_begin, chunk_size))
+ asyncio.get_event_loop().run_forever()
if __name__ == '__main__':
- # for microphone
- if args.audio_in is None:
- p = Process(target=one_thread,args=(0, 0, 0))
- p.start()
- p.join()
- print('end')
- else:
- # calculate the number of wavs for each preocess
- if args.audio_in.endswith(".scp"):
- f_scp = open(args.audio_in)
- wavs = f_scp.readlines()
- else:
- wavs = [args.audio_in]
- total_len=len(wavs)
- if total_len>=args.test_thread_num:
- chunk_size=int((total_len)/args.test_thread_num)
- remain_wavs=total_len-chunk_size*args.test_thread_num
- else:
- chunk_size=1
- remain_wavs=0
+ # for microphone
+ if args.audio_in is None:
+ p = Process(target=one_thread, args=(0, 0, 0))
+ p.start()
+ p.join()
+ print('end')
+ else:
+ # calculate the number of wavs for each preocess
+ if args.audio_in.endswith(".scp"):
+ f_scp = open(args.audio_in)
+ wavs = f_scp.readlines()
+ else:
+ wavs = [args.audio_in]
+ for wav in wavs:
+ wav_splits = wav.strip().split()
+ wav_name = wav_splits[0] if len(wav_splits) > 1 else "demo"
+ wav_path = wav_splits[1] if len(wav_splits) > 1 else wav_splits[0]
+ audio_type = os.path.splitext(wav_path)[-1].lower()
+ if audio_type not in SUPPORT_AUDIO_TYPE_SETS:
+ raise NotImplementedError(
+ f'Not supported audio type: {audio_type}')
- process_list = []
- chunk_begin=0
- for i in range(args.test_thread_num):
- now_chunk_size= chunk_size
- if remain_wavs>0:
- now_chunk_size=chunk_size+1
- remain_wavs=remain_wavs-1
- # process i handle wavs at chunk_begin and size of now_chunk_size
- p = Process(target=one_thread,args=(i,chunk_begin,now_chunk_size))
- chunk_begin=chunk_begin+now_chunk_size
- p.start()
- process_list.append(p)
+ total_len = len(wavs)
+ if total_len >= args.test_thread_num:
+ chunk_size = int(total_len / args.test_thread_num)
+ remain_wavs = total_len - chunk_size * args.test_thread_num
+ else:
+ chunk_size = 1
+ remain_wavs = 0
- for i in process_list:
- p.join()
+ process_list = []
+ chunk_begin = 0
+ for i in range(args.test_thread_num):
+ now_chunk_size = chunk_size
+ if remain_wavs > 0:
+ now_chunk_size = chunk_size + 1
+ remain_wavs = remain_wavs - 1
+ # process i handle wavs at chunk_begin and size of now_chunk_size
+ p = Process(target=one_thread, args=(i, chunk_begin, now_chunk_size))
+ chunk_begin = chunk_begin + now_chunk_size
+ p.start()
+ process_list.append(p)
- print('end')
+ for i in process_list:
+ p.join()
-
+ print('end')
diff --git a/funasr/runtime/websocket/CMakeLists.txt b/funasr/runtime/websocket/CMakeLists.txt
index 58ca972..c1715d8 100644
--- a/funasr/runtime/websocket/CMakeLists.txt
+++ b/funasr/runtime/websocket/CMakeLists.txt
@@ -6,12 +6,10 @@
set(CMAKE_POSITION_INDEPENDENT_CODE ON)
set(CMAKE_RUNTIME_OUTPUT_DIRECTORY ${CMAKE_BINARY_DIR}/bin)
-
option(ENABLE_WEBSOCKET "Whether to build websocket server" ON)
if(ENABLE_WEBSOCKET)
# cmake_policy(SET CMP0135 NEW)
-
include(FetchContent)
FetchContent_Declare(websocketpp
GIT_REPOSITORY https://github.com/zaphoyd/websocketpp.git
@@ -22,7 +20,6 @@
FetchContent_MakeAvailable(websocketpp)
include_directories(${PROJECT_SOURCE_DIR}/third_party/websocket)
-
FetchContent_Declare(asio
URL https://github.com/chriskohlhoff/asio/archive/refs/tags/asio-1-24-0.tar.gz
SOURCE_DIR ${PROJECT_SOURCE_DIR}/third_party/asio
@@ -38,8 +35,6 @@
FetchContent_MakeAvailable(json)
include_directories(${PROJECT_SOURCE_DIR}/third_party/json/include)
-
-
endif()
@@ -61,8 +56,8 @@
# install openssl first apt-get install libssl-dev
find_package(OpenSSL REQUIRED)
-add_executable(websocketmain "websocketmain.cpp" "websocketsrv.cpp")
-add_executable(websocketclient "websocketclient.cpp")
+add_executable(funasr-ws-server "funasr-ws-server.cpp" "websocket-server.cpp")
+add_executable(funasr-ws-client "funasr-ws-client.cpp")
-target_link_libraries(websocketclient PUBLIC funasr ssl crypto)
-target_link_libraries(websocketmain PUBLIC funasr ssl crypto)
+target_link_libraries(funasr-ws-client PUBLIC funasr ssl crypto)
+target_link_libraries(funasr-ws-server PUBLIC funasr ssl crypto)
diff --git a/funasr/runtime/websocket/funasr-ws-client.cpp b/funasr/runtime/websocket/funasr-ws-client.cpp
new file mode 100644
index 0000000..4a3c751
--- /dev/null
+++ b/funasr/runtime/websocket/funasr-ws-client.cpp
@@ -0,0 +1,366 @@
+/**
+ * Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights
+ * Reserved. MIT License (https://opensource.org/licenses/MIT)
+ */
+/* 2022-2023 by zhaomingwork */
+
+// client for websocket, support multiple threads
+// ./funasr-ws-client --server-ip <string>
+// --port <string>
+// --wav-path <string>
+// [--thread-num <int>]
+// [--is-ssl <int>] [--]
+// [--version] [-h]
+// example:
+// ./funasr-ws-client --server-ip 127.0.0.1 --port 8889 --wav-path test.wav --thread-num 1 --is-ssl 0
+
+#define ASIO_STANDALONE 1
+#include <websocketpp/client.hpp>
+#include <websocketpp/common/thread.hpp>
+#include <websocketpp/config/asio_client.hpp>
+#include <fstream>
+#include <atomic>
+#include <glog/logging.h>
+
+#include "audio.h"
+#include "nlohmann/json.hpp"
+#include "tclap/CmdLine.h"
+
+/**
+ * Define a semi-cross platform helper method that waits/sleeps for a bit.
+ */
+void WaitABit() {
+ #ifdef WIN32
+ Sleep(1000);
+ #else
+ sleep(1);
+ #endif
+}
+std::atomic<int> wav_index(0);
+
+bool IsTargetFile(const std::string& filename, const std::string target) {
+ std::size_t pos = filename.find_last_of(".");
+ if (pos == std::string::npos) {
+ return false;
+ }
+ std::string extension = filename.substr(pos + 1);
+ return (extension == target);
+}
+
+typedef websocketpp::config::asio_client::message_type::ptr message_ptr;
+typedef websocketpp::lib::shared_ptr<websocketpp::lib::asio::ssl::context> context_ptr;
+using websocketpp::lib::bind;
+using websocketpp::lib::placeholders::_1;
+using websocketpp::lib::placeholders::_2;
+context_ptr OnTlsInit(websocketpp::connection_hdl) {
+ context_ptr ctx = websocketpp::lib::make_shared<asio::ssl::context>(
+ asio::ssl::context::sslv23);
+
+ try {
+ ctx->set_options(
+ asio::ssl::context::default_workarounds | asio::ssl::context::no_sslv2 |
+ asio::ssl::context::no_sslv3 | asio::ssl::context::single_dh_use);
+
+ } catch (std::exception& e) {
+ LOG(ERROR) << e.what();
+ }
+ return ctx;
+}
+
+// template for tls or not config
+template <typename T>
+class WebsocketClient {
+ public:
+ // typedef websocketpp::client<T> client;
+ // typedef websocketpp::client<websocketpp::config::asio_tls_client>
+ // wss_client;
+ typedef websocketpp::lib::lock_guard<websocketpp::lib::mutex> scoped_lock;
+
+ WebsocketClient(int is_ssl) : m_open(false), m_done(false) {
+ // set up access channels to only log interesting things
+ m_client.clear_access_channels(websocketpp::log::alevel::all);
+ m_client.set_access_channels(websocketpp::log::alevel::connect);
+ m_client.set_access_channels(websocketpp::log::alevel::disconnect);
+ m_client.set_access_channels(websocketpp::log::alevel::app);
+
+ // Initialize the Asio transport policy
+ m_client.init_asio();
+
+ // Bind the handlers we are using
+ using websocketpp::lib::bind;
+ using websocketpp::lib::placeholders::_1;
+ m_client.set_open_handler(bind(&WebsocketClient::on_open, this, _1));
+ m_client.set_close_handler(bind(&WebsocketClient::on_close, this, _1));
+ // m_client.set_close_handler(bind(&WebsocketClient::on_close, this, _1));
+
+ m_client.set_message_handler(
+ [this](websocketpp::connection_hdl hdl, message_ptr msg) {
+ on_message(hdl, msg);
+ });
+
+ m_client.set_fail_handler(bind(&WebsocketClient::on_fail, this, _1));
+ m_client.clear_access_channels(websocketpp::log::alevel::all);
+ }
+
+ void on_message(websocketpp::connection_hdl hdl, message_ptr msg) {
+ const std::string& payload = msg->get_payload();
+ switch (msg->get_opcode()) {
+ case websocketpp::frame::opcode::text:
+ total_num=total_num+1;
+ LOG(INFO)<<total_num<<",on_message = " << payload;
+ if((total_num+1)==wav_index)
+ {
+ websocketpp::lib::error_code ec;
+ m_client.close(m_hdl, websocketpp::close::status::going_away, "", ec);
+ if (ec){
+ LOG(ERROR)<< "Error closing connection " << ec.message();
+ }
+ }
+ }
+ }
+
+ // This method will block until the connection is complete
+ void run(const std::string& uri, const std::vector<string>& wav_list, const std::vector<string>& wav_ids) {
+ // Create a new connection to the given URI
+ websocketpp::lib::error_code ec;
+ typename websocketpp::client<T>::connection_ptr con =
+ m_client.get_connection(uri, ec);
+ if (ec) {
+ m_client.get_alog().write(websocketpp::log::alevel::app,
+ "Get Connection Error: " + ec.message());
+ return;
+ }
+ // Grab a handle for this connection so we can talk to it in a thread
+ // safe manor after the event loop starts.
+ m_hdl = con->get_handle();
+
+ // Queue the connection. No DNS queries or network connections will be
+ // made until the io_service event loop is run.
+ m_client.connect(con);
+
+ // Create a thread to run the ASIO io_service event loop
+ websocketpp::lib::thread asio_thread(&websocketpp::client<T>::run,
+ &m_client);
+ while(true){
+ int i = wav_index.fetch_add(1);
+ if (i >= wav_list.size()) {
+ break;
+ }
+ send_wav_data(wav_list[i], wav_ids[i]);
+ }
+ WaitABit();
+
+ asio_thread.join();
+
+ }
+
+ // The open handler will signal that we are ready to start sending data
+ void on_open(websocketpp::connection_hdl) {
+ m_client.get_alog().write(websocketpp::log::alevel::app,
+ "Connection opened, starting data!");
+
+ scoped_lock guard(m_lock);
+ m_open = true;
+ }
+
+ // The close handler will signal that we should stop sending data
+ void on_close(websocketpp::connection_hdl) {
+ m_client.get_alog().write(websocketpp::log::alevel::app,
+ "Connection closed, stopping data!");
+
+ scoped_lock guard(m_lock);
+ m_done = true;
+ }
+
+ // The fail handler will signal that we should stop sending data
+ void on_fail(websocketpp::connection_hdl) {
+ m_client.get_alog().write(websocketpp::log::alevel::app,
+ "Connection failed, stopping data!");
+
+ scoped_lock guard(m_lock);
+ m_done = true;
+ }
+ // send wav to server
+ void send_wav_data(string wav_path, string wav_id) {
+ uint64_t count = 0;
+ std::stringstream val;
+
+ funasr::Audio audio(1);
+ int32_t sampling_rate = 16000;
+ if(IsTargetFile(wav_path.c_str(), "wav")){
+ int32_t sampling_rate = -1;
+ if(!audio.LoadWav(wav_path.c_str(), &sampling_rate))
+ return ;
+ }else if(IsTargetFile(wav_path.c_str(), "pcm")){
+ if (!audio.LoadPcmwav(wav_path.c_str(), &sampling_rate))
+ return ;
+ }else{
+ printf("Wrong wav extension");
+ exit(-1);
+ }
+
+ float* buff;
+ int len;
+ int flag = 0;
+ bool wait = false;
+ while (1) {
+ {
+ scoped_lock guard(m_lock);
+ // If the connection has been closed, stop generating data
+ if (m_done) {
+ break;
+ }
+ // If the connection hasn't been opened yet wait a bit and retry
+ if (!m_open) {
+ wait = true;
+ } else {
+ break;
+ }
+ }
+ if (wait) {
+ LOG(INFO) << "wait.." << m_open;
+ WaitABit();
+ continue;
+ }
+ }
+ websocketpp::lib::error_code ec;
+
+ nlohmann::json jsonbegin;
+ nlohmann::json chunk_size = nlohmann::json::array();
+ chunk_size.push_back(5);
+ chunk_size.push_back(0);
+ chunk_size.push_back(5);
+ jsonbegin["chunk_size"] = chunk_size;
+ jsonbegin["chunk_interval"] = 10;
+ jsonbegin["wav_name"] = wav_id;
+ jsonbegin["is_speaking"] = true;
+ m_client.send(m_hdl, jsonbegin.dump(), websocketpp::frame::opcode::text,
+ ec);
+
+ // fetch wav data use asr engine api
+ while (audio.Fetch(buff, len, flag) > 0) {
+ short iArray[len];
+
+ // convert float -1,1 to short -32768,32767
+ for (size_t i = 0; i < len; ++i) {
+ iArray[i] = (short)(buff[i] * 32767);
+ }
+ // send data to server
+ m_client.send(m_hdl, iArray, len * sizeof(short),
+ websocketpp::frame::opcode::binary, ec);
+ LOG(INFO) << "sended data len=" << len * sizeof(short);
+ // The most likely error that we will get is that the connection is
+ // not in the right state. Usually this means we tried to send a
+ // message to a connection that was closed or in the process of
+ // closing. While many errors here can be easily recovered from,
+ // in this simple example, we'll stop the data loop.
+ if (ec) {
+ m_client.get_alog().write(websocketpp::log::alevel::app,
+ "Send Error: " + ec.message());
+ break;
+ }
+ // WaitABit();
+ }
+ nlohmann::json jsonresult;
+ jsonresult["is_speaking"] = false;
+ m_client.send(m_hdl, jsonresult.dump(), websocketpp::frame::opcode::text,
+ ec);
+ // WaitABit();
+ }
+ websocketpp::client<T> m_client;
+
+ private:
+ websocketpp::connection_hdl m_hdl;
+ websocketpp::lib::mutex m_lock;
+ bool m_open;
+ bool m_done;
+ int total_num=0;
+};
+
+int main(int argc, char* argv[]) {
+ google::InitGoogleLogging(argv[0]);
+ FLAGS_logtostderr = true;
+
+ TCLAP::CmdLine cmd("funasr-ws-client", ' ', "1.0");
+ TCLAP::ValueArg<std::string> server_ip_("", "server-ip", "server-ip", true,
+ "127.0.0.1", "string");
+ TCLAP::ValueArg<std::string> port_("", "port", "port", true, "8889", "string");
+ TCLAP::ValueArg<std::string> wav_path_("", "wav-path",
+ "the input could be: wav_path, e.g.: asr_example.wav; pcm_path, e.g.: asr_example.pcm; wav.scp, kaldi style wav list (wav_id \t wav_path)",
+ true, "", "string");
+ TCLAP::ValueArg<int> thread_num_("", "thread-num", "thread-num",
+ false, 1, "int");
+ TCLAP::ValueArg<int> is_ssl_(
+ "", "is-ssl", "is-ssl is 1 means use wss connection, or use ws connection",
+ false, 0, "int");
+
+ cmd.add(server_ip_);
+ cmd.add(port_);
+ cmd.add(wav_path_);
+ cmd.add(thread_num_);
+ cmd.add(is_ssl_);
+ cmd.parse(argc, argv);
+
+ std::string server_ip = server_ip_.getValue();
+ std::string port = port_.getValue();
+ std::string wav_path = wav_path_.getValue();
+ int threads_num = thread_num_.getValue();
+ int is_ssl = is_ssl_.getValue();
+
+ std::vector<websocketpp::lib::thread> client_threads;
+ std::string uri = "";
+ if (is_ssl == 1) {
+ uri = "wss://" + server_ip + ":" + port;
+ } else {
+ uri = "ws://" + server_ip + ":" + port;
+ }
+
+ // read wav_path
+ std::vector<string> wav_list;
+ std::vector<string> wav_ids;
+ string default_id = "wav_default_id";
+ if(IsTargetFile(wav_path, "wav") || IsTargetFile(wav_path, "pcm")){
+ wav_list.emplace_back(wav_path);
+ wav_ids.emplace_back(default_id);
+ }
+ else if(IsTargetFile(wav_path, "scp")){
+ ifstream in(wav_path);
+ if (!in.is_open()) {
+ printf("Failed to open scp file");
+ return 0;
+ }
+ string line;
+ while(getline(in, line))
+ {
+ istringstream iss(line);
+ string column1, column2;
+ iss >> column1 >> column2;
+ wav_list.emplace_back(column2);
+ wav_ids.emplace_back(column1);
+ }
+ in.close();
+ }else{
+ printf("Please check the wav extension!");
+ exit(-1);
+ }
+
+ for (size_t i = 0; i < threads_num; i++) {
+ client_threads.emplace_back([uri, wav_list, wav_ids, is_ssl]() {
+ if (is_ssl == 1) {
+ WebsocketClient<websocketpp::config::asio_tls_client> c(is_ssl);
+
+ c.m_client.set_tls_init_handler(bind(&OnTlsInit, ::_1));
+
+ c.run(uri, wav_list, wav_ids);
+ } else {
+ WebsocketClient<websocketpp::config::asio_client> c(is_ssl);
+
+ c.run(uri, wav_list, wav_ids);
+ }
+ });
+ }
+
+ for (auto& t : client_threads) {
+ t.join();
+ }
+}
\ No newline at end of file
diff --git a/funasr/runtime/websocket/websocketmain.cpp b/funasr/runtime/websocket/funasr-ws-server.cpp
similarity index 97%
rename from funasr/runtime/websocket/websocketmain.cpp
rename to funasr/runtime/websocket/funasr-ws-server.cpp
index fabf6d8..872f6a1 100644
--- a/funasr/runtime/websocket/websocketmain.cpp
+++ b/funasr/runtime/websocket/funasr-ws-server.cpp
@@ -5,12 +5,12 @@
/* 2022-2023 by zhaomingwork */
// io server
-// Usage:websocketmain [--model_thread_num <int>] [--decoder_thread_num <int>]
+// Usage:funasr-ws-server [--model_thread_num <int>] [--decoder_thread_num <int>]
// [--io_thread_num <int>] [--port <int>] [--listen_ip
// <string>] [--punc-quant <string>] [--punc-dir <string>]
// [--vad-quant <string>] [--vad-dir <string>] [--quantize
// <string>] --model-dir <string> [--] [--version] [-h]
-#include "websocketsrv.h"
+#include "websocket-server.h"
using namespace std;
void GetValue(TCLAP::ValueArg<std::string>& value_arg, string key,
@@ -25,7 +25,7 @@
google::InitGoogleLogging(argv[0]);
FLAGS_logtostderr = true;
- TCLAP::CmdLine cmd("websocketmain", ' ', "1.0");
+ TCLAP::CmdLine cmd("funasr-ws-server", ' ', "1.0");
TCLAP::ValueArg<std::string> model_dir(
"", MODEL_DIR,
"the asr model path, which contains model.onnx, config.yaml, am.mvn",
diff --git a/funasr/runtime/websocket/readme.md b/funasr/runtime/websocket/readme.md
index 99255c8..4a1a9d4 100644
--- a/funasr/runtime/websocket/readme.md
+++ b/funasr/runtime/websocket/readme.md
@@ -51,7 +51,7 @@
```shell
cd bin
- ./websocketmain [--model_thread_num <int>] [--decoder_thread_num <int>]
+ ./funasr-ws-server [--model_thread_num <int>] [--decoder_thread_num <int>]
[--io_thread_num <int>] [--port <int>] [--listen_ip
<string>] [--punc-quant <string>] [--punc-dir <string>]
[--vad-quant <string>] [--vad-dir <string>] [--quantize
@@ -88,19 +88,38 @@
If use vad, please add: --vad-dir <string>
If use punc, please add: --punc-dir <string>
example:
- websocketmain --model-dir /FunASR/funasr/runtime/onnxruntime/export/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch
+ funasr-ws-server --model-dir /FunASR/funasr/runtime/onnxruntime/export/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch
```
## Run websocket client test
```shell
-Usage: ./websocketclient server_ip port wav_path threads_num is_ssl
+./funasr-ws-client --server-ip <string>
+ --port <string>
+ --wav-path <string>
+ [--thread-num <int>]
+ [--is-ssl <int>] [--]
+ [--version] [-h]
-is_ssl is 1 means use wss connection, or use ws connection
+Where:
+ --server-ip <string>
+ (required) server-ip
+
+ --port <string>
+ (required) port
+
+ --wav-path <string>
+ (required) the input could be: wav_path, e.g.: asr_example.wav;
+ pcm_path, e.g.: asr_example.pcm; wav.scp, kaldi style wav list (wav_id \t wav_path)
+
+ --thread-num <int>
+ thread-num
+
+ --is-ssl <int>
+ is-ssl is 1 means use wss connection, or use ws connection
example:
-
-websocketclient 127.0.0.1 8889 funasr/runtime/websocket/test.pcm.wav 64 0
+./funasr-ws-client --server-ip 127.0.0.1 --port 8889 --wav-path test.wav --thread-num 1 --is-ssl 0
result json, example like:
{"mode":"offline","text":"娆㈣繋澶у鏉ヤ綋楠岃揪鎽╅櫌鎺ㄥ嚭鐨勮闊宠瘑鍒ā鍨�","wav_name":"wav2"}
diff --git a/funasr/runtime/websocket/websocketsrv.cpp b/funasr/runtime/websocket/websocket-server.cpp
similarity index 90%
rename from funasr/runtime/websocket/websocketsrv.cpp
rename to funasr/runtime/websocket/websocket-server.cpp
index eb3c8db..a311c23 100644
--- a/funasr/runtime/websocket/websocketsrv.cpp
+++ b/funasr/runtime/websocket/websocket-server.cpp
@@ -10,7 +10,7 @@
// pools, one for handle network data and one for asr decoder.
// now only support offline engine.
-#include "websocketsrv.h"
+#include "websocket-server.h"
#include <thread>
#include <utility>
@@ -22,12 +22,11 @@
std::string& s_keyfile) {
namespace asio = websocketpp::lib::asio;
- std::cout << "on_tls_init called with hdl: " << hdl.lock().get() << std::endl;
- std::cout << "using TLS mode: "
+ LOG(INFO) << "on_tls_init called with hdl: " << hdl.lock().get();
+ LOG(INFO) << "using TLS mode: "
<< (mode == MOZILLA_MODERN ? "Mozilla Modern"
- : "Mozilla Intermediate")
- << std::endl;
-
+ : "Mozilla Intermediate");
+
context_ptr ctx = websocketpp::lib::make_shared<asio::ssl::context>(
asio::ssl::context::sslv23);
@@ -49,7 +48,7 @@
ctx->use_private_key_file(s_keyfile, asio::ssl::context::pem);
} catch (std::exception& e) {
- std::cout << "Exception: " << e.what() << std::endl;
+ LOG(INFO) << "Exception: " << e.what();
}
return ctx;
}
@@ -86,8 +85,7 @@
ec);
}
- std::cout << "buffer.size=" << buffer.size()
- << ",result json=" << jsonresult.dump() << std::endl;
+ LOG(INFO) << "buffer.size=" << buffer.size() << ",result json=" << jsonresult.dump();
if (!isonline) {
// close the client if it is not online asr
// server_->close(hdl, websocketpp::close::status::normal, "DONE", ec);
@@ -110,14 +108,14 @@
data_msg->samples = std::make_shared<std::vector<char>>();
data_msg->msg = nlohmann::json::parse("{}");
data_map.emplace(hdl, data_msg);
- std::cout << "on_open, active connections: " << data_map.size() << std::endl;
+ LOG(INFO) << "on_open, active connections: " << data_map.size();
}
void WebSocketServer::on_close(websocketpp::connection_hdl hdl) {
scoped_lock guard(m_lock);
data_map.erase(hdl); // remove data vector when connection is closed
- std::cout << "on_close, active connections: " << data_map.size() << std::endl;
+ LOG(INFO) << "on_close, active connections: " << data_map.size();
}
// remove closed connection
@@ -143,7 +141,7 @@
}
for (auto hdl : to_remove) {
data_map.erase(hdl);
- std::cout << "remove one connection " << std::endl;
+ LOG(INFO)<< "remove one connection ";
}
}
void WebSocketServer::on_message(websocketpp::connection_hdl hdl,
@@ -161,7 +159,7 @@
lock.unlock();
if (sample_data_p == nullptr) {
- std::cout << "error when fetch sample data vector" << std::endl;
+ LOG(INFO) << "error when fetch sample data vector";
return;
}
@@ -176,7 +174,7 @@
if (jsonresult["is_speaking"] == false ||
jsonresult["is_finished"] == true) {
- std::cout << "client done" << std::endl;
+ LOG(INFO) << "client done";
if (isonline) {
// do_close(ws);
@@ -225,9 +223,9 @@
// init model with api
asr_hanlde = FunOfflineInit(model_path, thread_num);
- std::cout << "model ready" << std::endl;
+ LOG(INFO) << "model successfully inited";
} catch (const std::exception& e) {
- std::cout << e.what() << std::endl;
+ LOG(INFO) << e.what();
}
}
diff --git a/funasr/runtime/websocket/websocketsrv.h b/funasr/runtime/websocket/websocket-server.h
similarity index 97%
rename from funasr/runtime/websocket/websocketsrv.h
rename to funasr/runtime/websocket/websocket-server.h
index 3cb8816..198af1c 100644
--- a/funasr/runtime/websocket/websocketsrv.h
+++ b/funasr/runtime/websocket/websocket-server.h
@@ -10,8 +10,8 @@
// pools, one for handle network data and one for asr decoder.
// now only support offline engine.
-#ifndef WEBSOCKETSRV_SERVER_H_
-#define WEBSOCKETSRV_SERVER_H_
+#ifndef WEBSOCKET_SERVER_H_
+#define WEBSOCKET_SERVER_H_
#include <iostream>
#include <map>
@@ -134,4 +134,4 @@
websocketpp::lib::mutex m_lock; // mutex for sample_map
};
-#endif // WEBSOCKETSRV_SERVER_H_
+#endif // WEBSOCKET_SERVER_H_
diff --git a/funasr/runtime/websocket/websocketclient.cpp b/funasr/runtime/websocket/websocketclient.cpp
deleted file mode 100644
index e9f8f1d..0000000
--- a/funasr/runtime/websocket/websocketclient.cpp
+++ /dev/null
@@ -1,277 +0,0 @@
-/**
- * Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights
- * Reserved. MIT License (https://opensource.org/licenses/MIT)
- */
-/* 2022-2023 by zhaomingwork */
-
-// client for websocket, support multiple threads
-// Usage: websocketclient server_ip port wav_path threads_num
-
-#define ASIO_STANDALONE 1
-#include <websocketpp/client.hpp>
-#include <websocketpp/common/thread.hpp>
-#include <websocketpp/config/asio_client.hpp>
-
-#include "audio.h"
-#include "nlohmann/json.hpp"
-
-/**
- * Define a semi-cross platform helper method that waits/sleeps for a bit.
- */
-void wait_a_bit() {
-#ifdef WIN32
- Sleep(1000);
-#else
- sleep(1);
-#endif
-}
-typedef websocketpp::config::asio_client::message_type::ptr message_ptr;
-typedef websocketpp::lib::shared_ptr<websocketpp::lib::asio::ssl::context>
- context_ptr;
-using websocketpp::lib::bind;
-using websocketpp::lib::placeholders::_1;
-using websocketpp::lib::placeholders::_2;
-context_ptr on_tls_init(websocketpp::connection_hdl) {
- context_ptr ctx = websocketpp::lib::make_shared<asio::ssl::context>(
- asio::ssl::context::sslv23);
-
- try {
- ctx->set_options(
- asio::ssl::context::default_workarounds | asio::ssl::context::no_sslv2 |
- asio::ssl::context::no_sslv3 | asio::ssl::context::single_dh_use);
-
- } catch (std::exception& e) {
- std::cout << e.what() << std::endl;
- }
- return ctx;
-}
-// template for tls or not config
-template <typename T>
-class websocket_client {
- public:
- // typedef websocketpp::client<T> client;
- // typedef websocketpp::client<websocketpp::config::asio_tls_client>
- // wss_client;
- typedef websocketpp::lib::lock_guard<websocketpp::lib::mutex> scoped_lock;
-
- websocket_client(int is_ssl) : m_open(false), m_done(false) {
- // set up access channels to only log interesting things
-
- m_client.clear_access_channels(websocketpp::log::alevel::all);
- m_client.set_access_channels(websocketpp::log::alevel::connect);
- m_client.set_access_channels(websocketpp::log::alevel::disconnect);
- m_client.set_access_channels(websocketpp::log::alevel::app);
-
- // Initialize the Asio transport policy
- m_client.init_asio();
-
- // Bind the handlers we are using
- using websocketpp::lib::bind;
- using websocketpp::lib::placeholders::_1;
- m_client.set_open_handler(bind(&websocket_client::on_open, this, _1));
- m_client.set_close_handler(bind(&websocket_client::on_close, this, _1));
- m_client.set_close_handler(bind(&websocket_client::on_close, this, _1));
-
- m_client.set_message_handler(
- [this](websocketpp::connection_hdl hdl, message_ptr msg) {
- on_message(hdl, msg);
- });
-
- m_client.set_fail_handler(bind(&websocket_client::on_fail, this, _1));
- m_client.clear_access_channels(websocketpp::log::alevel::all);
- }
- void on_message(websocketpp::connection_hdl hdl, message_ptr msg) {
- const std::string& payload = msg->get_payload();
- switch (msg->get_opcode()) {
- case websocketpp::frame::opcode::text:
- std::cout << "on_message=" << payload << std::endl;
- }
- }
- // This method will block until the connection is complete
-
- void run(const std::string& uri, const std::string& wav_path) {
- // Create a new connection to the given URI
- websocketpp::lib::error_code ec;
- typename websocketpp::client<T>::connection_ptr con =
- m_client.get_connection(uri, ec);
- if (ec) {
- m_client.get_alog().write(websocketpp::log::alevel::app,
- "Get Connection Error: " + ec.message());
- return;
- }
- this->wav_path = std::move(wav_path);
- // Grab a handle for this connection so we can talk to it in a thread
- // safe manor after the event loop starts.
- m_hdl = con->get_handle();
-
- // Queue the connection. No DNS queries or network connections will be
- // made until the io_service event loop is run.
- m_client.connect(con);
-
- // Create a thread to run the ASIO io_service event loop
- websocketpp::lib::thread asio_thread(&websocketpp::client<T>::run,
- &m_client);
-
- send_wav_data();
- asio_thread.join();
- }
-
- // The open handler will signal that we are ready to start sending data
- void on_open(websocketpp::connection_hdl) {
- m_client.get_alog().write(websocketpp::log::alevel::app,
- "Connection opened, starting data!");
-
- scoped_lock guard(m_lock);
- m_open = true;
- }
-
- // The close handler will signal that we should stop sending data
- void on_close(websocketpp::connection_hdl) {
- m_client.get_alog().write(websocketpp::log::alevel::app,
- "Connection closed, stopping data!");
-
- scoped_lock guard(m_lock);
- m_done = true;
- }
-
- // The fail handler will signal that we should stop sending data
- void on_fail(websocketpp::connection_hdl) {
- m_client.get_alog().write(websocketpp::log::alevel::app,
- "Connection failed, stopping data!");
-
- scoped_lock guard(m_lock);
- m_done = true;
- }
- // send wav to server
- void send_wav_data() {
- uint64_t count = 0;
- std::stringstream val;
-
- funasr::Audio audio(1);
- int32_t sampling_rate = 16000;
-
- if (!audio.LoadPcmwav(wav_path.c_str(), &sampling_rate)) {
- std::cout << "error in load wav" << std::endl;
- return;
- }
-
- float* buff;
- int len;
- int flag = 0;
- bool wait = false;
- while (1) {
- {
- scoped_lock guard(m_lock);
- // If the connection has been closed, stop generating data
- if (m_done) {
- break;
- }
-
- // If the connection hasn't been opened yet wait a bit and retry
- if (!m_open) {
- wait = true;
- } else {
- break;
- }
- }
-
- if (wait) {
- std::cout << "wait.." << m_open << std::endl;
- wait_a_bit();
-
- continue;
- }
- }
- websocketpp::lib::error_code ec;
-
- nlohmann::json jsonbegin;
- nlohmann::json chunk_size = nlohmann::json::array();
- chunk_size.push_back(5);
- chunk_size.push_back(0);
- chunk_size.push_back(5);
- jsonbegin["chunk_size"] = chunk_size;
- jsonbegin["chunk_interval"] = 10;
- jsonbegin["wav_name"] = "damo";
- jsonbegin["is_speaking"] = true;
- m_client.send(m_hdl, jsonbegin.dump(), websocketpp::frame::opcode::text,
- ec);
-
- // fetch wav data use asr engine api
- while (audio.Fetch(buff, len, flag) > 0) {
- short iArray[len];
-
- // convert float -1,1 to short -32768,32767
- for (size_t i = 0; i < len; ++i) {
- iArray[i] = (short)(buff[i] * 32767);
- }
- // send data to server
- m_client.send(m_hdl, iArray, len * sizeof(short),
- websocketpp::frame::opcode::binary, ec);
- std::cout << "sended data len=" << len * sizeof(short) << std::endl;
- // The most likely error that we will get is that the connection is
- // not in the right state. Usually this means we tried to send a
- // message to a connection that was closed or in the process of
- // closing. While many errors here can be easily recovered from,
- // in this simple example, we'll stop the data loop.
- if (ec) {
- m_client.get_alog().write(websocketpp::log::alevel::app,
- "Send Error: " + ec.message());
- break;
- }
-
- wait_a_bit();
- }
- nlohmann::json jsonresult;
- jsonresult["is_speaking"] = false;
- m_client.send(m_hdl, jsonresult.dump(), websocketpp::frame::opcode::text,
- ec);
- wait_a_bit();
- }
- websocketpp::client<T> m_client;
-
- private:
- websocketpp::connection_hdl m_hdl;
- websocketpp::lib::mutex m_lock;
- std::string wav_path;
- bool m_open;
- bool m_done;
-};
-
-int main(int argc, char* argv[]) {
- if (argc < 6) {
- printf("Usage: %s server_ip port wav_path threads_num is_ssl\n", argv[0]);
- exit(-1);
- }
- std::string server_ip = argv[1];
- std::string port = argv[2];
- std::string wav_path = argv[3];
- int threads_num = atoi(argv[4]);
- int is_ssl = atoi(argv[5]);
- std::vector<websocketpp::lib::thread> client_threads;
- std::string uri = "";
- if (is_ssl == 1) {
- uri = "wss://" + server_ip + ":" + port;
- } else {
- uri = "ws://" + server_ip + ":" + port;
- }
-
- for (size_t i = 0; i < threads_num; i++) {
- client_threads.emplace_back([uri, wav_path, is_ssl]() {
- if (is_ssl == 1) {
- websocket_client<websocketpp::config::asio_tls_client> c(is_ssl);
-
- c.m_client.set_tls_init_handler(bind(&on_tls_init, ::_1));
-
- c.run(uri, wav_path);
- } else {
- websocket_client<websocketpp::config::asio_client> c(is_ssl);
-
- c.run(uri, wav_path);
- }
- });
- }
-
- for (auto& t : client_threads) {
- t.join();
- }
-}
\ No newline at end of file
diff --git a/tests/test_asr_inference_pipeline.py b/tests/test_asr_inference_pipeline.py
index 9098ea6..2b21acf 100644
--- a/tests/test_asr_inference_pipeline.py
+++ b/tests/test_asr_inference_pipeline.py
@@ -87,6 +87,7 @@
rec_result = inference_pipeline(
audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_hotword.wav')
logger.info("asr inference result: {0}".format(rec_result))
+ assert rec_result["text"] == "鍥藉姟闄㈠彂灞曠爺绌朵腑蹇冨競鍦虹粡娴庣爺绌舵墍鍓墍闀块倱閮佹澗璁や负"
def test_paraformer_large_aishell1(self):
inference_pipeline = pipeline(
@@ -95,6 +96,7 @@
rec_result = inference_pipeline(
audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav')
logger.info("asr inference result: {0}".format(rec_result))
+ assert rec_result["text"] == "娆㈣繋澶у鏉ヤ綋楠岃揪鎽╅櫌鎺ㄥ嚭鐨勮闊宠瘑鍒ā鍨�"
def test_paraformer_large_aishell2(self):
inference_pipeline = pipeline(
@@ -103,6 +105,7 @@
rec_result = inference_pipeline(
audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav')
logger.info("asr inference result: {0}".format(rec_result))
+ assert rec_result["text"] == "娆㈣繋澶у鏉ヤ綋楠岃揪鎽╅櫌鎺ㄥ嚭鐨勮闊宠瘑鍒ā鍨�"
def test_paraformer_large_common(self):
inference_pipeline = pipeline(
@@ -111,6 +114,7 @@
rec_result = inference_pipeline(
audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav')
logger.info("asr inference result: {0}".format(rec_result))
+ assert rec_result["text"] == "娆㈣繋澶у鏉ヤ綋楠岃揪鎽╅櫌鎺ㄥ嚭鐨勮闊宠瘑鍒ā鍨�"
def test_paraformer_large_online_common(self):
inference_pipeline = pipeline(
@@ -119,6 +123,7 @@
rec_result = inference_pipeline(
audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav')
logger.info("asr inference result: {0}".format(rec_result))
+ assert rec_result["text"] == "娆㈣繋澶� 瀹舵潵 浣撻獙杈� 鎽╅櫌鎺� 鍑虹殑 璇煶璇� 鍒ā 鍨�"
def test_paraformer_online_common(self):
inference_pipeline = pipeline(
@@ -127,6 +132,7 @@
rec_result = inference_pipeline(
audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav')
logger.info("asr inference result: {0}".format(rec_result))
+ assert rec_result["text"] == "娆㈣繋 澶у鏉� 浣撻獙杈� 鎽╅櫌鎺� 鍑虹殑 璇煶璇� 鍒ā 鍨�"
def test_paraformer_tiny_commandword(self):
inference_pipeline = pipeline(
diff --git a/tests/test_asr_vad_punc_inference_pipeline.py b/tests/test_asr_vad_punc_inference_pipeline.py
index 628b256..f86f23d 100644
--- a/tests/test_asr_vad_punc_inference_pipeline.py
+++ b/tests/test_asr_vad_punc_inference_pipeline.py
@@ -26,6 +26,7 @@
rec_result = inference_pipeline(
audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav')
logger.info("asr_vad_punc inference result: {0}".format(rec_result))
+ assert rec_result["text"] == "娆㈣繋澶у鏉ヤ綋楠岃揪鎽╅櫌鎺ㄥ嚭鐨勮闊宠瘑鍒ā鍨嬨��"
if __name__ == '__main__':
--
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