From f03a604204bbe0c79e53b01237a37e88683938c6 Mon Sep 17 00:00:00 2001
From: zhifu gao <zhifu.gzf@alibaba-inc.com>
Date: 星期六, 13 五月 2023 00:17:32 +0800
Subject: [PATCH] Merge pull request #505 from zhaomingwork/cpp-python-websocket-compatible
---
funasr/runtime/websocket/websocketclient.cpp | 20 ++++++++++++++++++--
1 files changed, 18 insertions(+), 2 deletions(-)
diff --git a/funasr/runtime/websocket/websocketclient.cpp b/funasr/runtime/websocket/websocketclient.cpp
index 3ab4e99..078fc5a 100644
--- a/funasr/runtime/websocket/websocketclient.cpp
+++ b/funasr/runtime/websocket/websocketclient.cpp
@@ -13,6 +13,7 @@
#include <websocketpp/config/asio_no_tls_client.hpp>
#include "audio.h"
+#include "nlohmann/json.hpp"
/**
* Define a semi-cross platform helper method that waits/sleeps for a bit.
@@ -156,6 +157,19 @@
}
}
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];
@@ -181,8 +195,10 @@
wait_a_bit();
}
-
- m_client.send(m_hdl, "Done", websocketpp::frame::opcode::text, ec);
+ nlohmann::json jsonresult;
+ jsonresult["is_speaking"] = false;
+ m_client.send(m_hdl, jsonresult.dump(), websocketpp::frame::opcode::text,
+ ec);
wait_a_bit();
}
--
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