下载地址：/download/qq_39735878/12447473
unity写的讯飞语音识别合成与评测功能，走的是webapi,连接讯飞WebSocket服务器没有接入任何sdk，也没有多余插件，开发工具unity.3.6 ，真正实现了跨平台，不用在每个平台重新接入sdk这么麻烦
最近项目要实现语音识别合成与评测功能，看到了讯飞正好符合要求就打算介入讯飞sdk来实现，但是讯飞的sdk没有unity插件只有单个平台的sdk,看了开发文档之后发现可以用webapi实现跨平台这个可以再不同平台实现语音识别功能。讯飞语音合成与识别用的是WebSocket网络协议先要构建接口鉴权。在握手阶段，请求方需要对请求进行签名，服务端通过签名来校验请求的合法性。代码如下：
string GetUrl(string uriStr){Uri uri = new Uri(uriStr);string date = DateTime.Now.ToString("r");string signature_origin = string.Format("host: " + uri.Host + "\ndate: " + date + "\nGET " + uri.AbsolutePath + " HTTP/1.1");HMACSHA256 mac = new HMACSHA256(Encoding.UTF8.GetBytes(APISecret));string signature = Convert.ToBase64String(puteHash(Encoding.UTF8.GetBytes(signature_origin)));string authorization_origin = string.Format("api_key=\"{0}\",algorithm=\"hmac-sha256\",headers=\"host date request-line\",signature=\"{1}\"", APIKey, signature);string authorization = Convert.ToBase64String(Encoding.UTF8.GetBytes(authorization_origin));string url = string.Format("{0}?authorization={1}&date={2}&host={3}", uri, authorization, date, uri.Host);return url;}
获取到鉴权整个url后就需要连接讯飞WebSocket服务器了。连接成功之后需要获取unity录音的数据流AudioClip转化为byte[],需要每隔一段时间发送数据直到录音结束。经过研究这里提供一种思路来截取语音流：
public static byte[] 获取音频流片段(int star, int length, AudioClip recordedClip){float[] soundata = new float[length];recordedClip.GetData(soundata, star);int rescaleFactor = 32767;byte[] outData = new byte[soundata.Length * 2];for (int i = 0; i < soundata.Length; i++){short temshort = (short)(soundata[i] * rescaleFactor);byte[] temdata = BitConverter.GetBytes(temshort);outData[i * 2] = temdata[0];outData[i * 2 + 1] = temdata[1];}return outData;}
其中参数start是录音的位置与音频采样率有关，举个例子如果采样率是16000，那么start=16000的位置就是从获取一秒后语音流位置，以此类推。同样length也是采样长度。如果采样率是16000，start=16000x,length=16000y则是从x秒开始录音长度为y的一段语音流数据。明白这个道理后再研究文档就可以实现语音识别逻辑了：
string APPID = "5c81de59";string APISecret = "ea4d5e9b06f8cfb0deae4d5360e7f8a7";string APIKey = "94348d7a6d5f3807176cb1f4923efa5c";public event Action<string> 语音识别完成事件; //语音识别回调事件public AudioClip RecordedClip;ClientWebSocket 语音识别WebSocket; public void 开始语音识别(){if (语音识别WebSocket!=null && 语音识别WebSocket.State == WebSocketState.Open){Debug.LogWarning("开始语音识别失败！，等待上次识别连接结束");return;}连接语音识别WebSocket();RecordedClip = Microphone.Start(null, false, 60, 16000);}public IEnumerator 停止语音识别(){Microphone.End(null);yield return new WaitUntil(()=>语音识别WebSocket.State != WebSocketState.Open);Debug.Log("识别结束，停止录音");}async void 连接语音识别WebSocket(){using (语音识别WebSocket = new ClientWebSocket()){CancellationToken ct = new CancellationToken();Uri url = new Uri(GetUrl("wss://iat-/v2/iat"));await 语音识别WebSocket.ConnectAsync(url, ct);Debug.Log("连接成功");StartCoroutine(发送录音数据流(语音识别WebSocket));StringBuilder stringBuilder = new StringBuilder();while (语音识别WebSocket.State == WebSocketState.Open){var result = new byte[4096];await 语音识别WebSocket.ReceiveAsync(new ArraySegment<byte>(result), ct);//接受数据List<byte> list = new List<byte>(result);while (list[list.Count - 1] == 0x00) list.RemoveAt(list.Count - 1);//去除空字节string str = Encoding.UTF8.GetString(list.ToArray());Debug.Log("接收消息：" + str);stringBuilder.Append(获取识别单词(str));JSONNode js = JSONNode.Parse(str);JSONNode data = js["data"];if (data["status"] == 2){语音识别WebSocket.Abort();}}Debug.LogWarning("断开连接");string s = stringBuilder.ToString();if (!string.IsNullOrEmpty(s)){语音识别完成事件?.Invoke(s);Debug.LogWarning("识别到声音：" + s);}}}string 获取识别单词(string str){StringBuilder stringBuilder = new StringBuilder();if (!string.IsNullOrEmpty(str)){JSONNode recivejson = JSONNode.Parse(str);JSONNode ws = recivejson["data"]["result"]["ws"];foreach (JSONNode item in ws){JSONNode cw = item["cw"];foreach (JSONNode item1 in cw){stringBuilder.Append((string)item1["w"]);}}}return stringBuilder.ToString();}void 发送数据(byte[] audio, int status, ClientWebSocket socket){if (socket.State != WebSocketState.Open){return;}JSONNode jn = new JSONNode{{"common",new JSONNode{{"app_id",APPID}}},{"business",new JSONNode{{"language","zh_cn"},//识别语音{"domain","iat"},{"accent","mandarin"},{"vad_eos",2000}}},{"data",new JSONNode{{"status",0 },{"encoding","raw" },{"format","audio/L16;rate=16000"}}}};JSONNode data = jn["data"];if (status < 2){data["audio"] = Convert.ToBase64String(audio);}data["status"] = status;Debug.Log("发送消息:" + jn);socket.SendAsync(new ArraySegment<byte>(Encoding.UTF8.GetBytes(jn)), WebSocketMessageType.Binary, true, new CancellationToken()); //发送数据}IEnumerator 发送录音数据流(ClientWebSocket socket){yield return new WaitWhile(() => Microphone.GetPosition(null) <= 0);float t = 0;int position = Microphone.GetPosition(null);const float waitTime = 0.04f;//每隔40ms发送音频int status = 0;int lastPosition = 0;const int Maxlength = 640;//最大发送长度while (position < RecordedClip.samples && socket.State == WebSocketState.Open){t += waitTime;yield return new WaitForSecondsRealtime(waitTime);if (Microphone.IsRecording(null)) position = Microphone.GetPosition(null);Debug.Log("录音时长：" + t + "position=" + position + ",lastPosition=" + lastPosition);if (position <= lastPosition){Debug.LogWarning("字节流发送完毕！强制结束！");break;}int length = position - lastPosition > Maxlength ? Maxlength : position - lastPosition;byte[] date = 获取音频流片段(lastPosition, length, RecordedClip);发送数据(date, status, socket);lastPosition = lastPosition + length;status = 1;}发送数据(null, 2, socket);//WebSocket.CloseAsync(WebSocketCloseStatus.NormalClosure, "关闭WebSocket连接",new CancellationToken());Microphone.End(null);}
接下来再说说语音合成，主要的难点是语音数据流的播放。服务器发送给客户端的数据不是连续的，需要分段接受，并解析。并且需要把客户端接受的byte[]转化为AudioClip可读取的float[]类型。通过查阅资料可用以下方法实现：
public static float[] bytesToFloat(byte[] byteArray)//byte[]数组转化为AudioClip可读取的float[]类型{float[] sounddata = new float[byteArray.Length / 2];for (int i = 0; i < sounddata.Length; i++){sounddata[i] = bytesToFloat(byteArray[i * 2], byteArray[i * 2 + 1]);}return sounddata;}static float bytesToFloat(byte firstByte, byte secondByte){// convert two bytes to one short (little endian)//小端和大端顺序要调整short s;if (BitConverter.IsLittleEndian)s = (short)((secondByte << 8) | firstByte);elses = (short)((firstByte << 8) | secondByte);// convert to range from -1 to (just below) 1return s / 32768.0F;}
当每次获取并解析成功一段语音流是可以把获取的float[]放入到队列中准备播放。这时还需要语音流播放完毕判断并停止播放。我这里的逻辑是用一个变量记录语音流总长度，当接受到新的语音流数据或播放空帧音频是增加这个变量长度，再实时与播放长度做对比就可以了。不废话了合成的所有逻辑代码如下：
public AudioSource 语音合成流播放器;ClientWebSocket 语音合成WebSocket;int 语音流总长度 = 0;Queue<float> 播放队列 = new Queue<float>();/// <summary>/// 开始语音合成/// </summary>/// <param name="s">需要合成的字符串</param>/// <param name="voice">发音人，默认是xiaoyan</param>/// <param name="speed">语速，范围是0~100，默认是50</param>/// <param name="volume">音量，范围是0~100，默认50</param>public IEnumerator 开始语音合成(String text, string voice = "xiaoyan", int speed = 50, int volume = 50){if (语音合成WebSocket != null){语音合成WebSocket.Abort();}if (语音合成流播放器==null){语音合成流播放器 = gameObject.AddComponent<AudioSource>();}语音合成流播放器.Stop();连接语音合成WebSocket(GetUrl("wss://tts-/v2/tts"),text, voice,speed,volume);语音合成流播放器.loop = true;语音合成流播放器.clip = AudioClip.Create("语音合成流", 16000*60 , 1, 16000, true, OnAudioRead);//播放器循环播放采样率16000，最长播放时长60秒，如果合成语音更长可以设置一个必60足够大的数语音合成流播放器.Play();//播放语音流while (true){yield return null;if (!语音合成流播放器.isPlaying || 语音合成WebSocket.State == WebSocketState.Aborted && 语音合成流播放器.timeSamples >= 语音流总长度){Debug.Log(text+"语音流播放完毕！");语音合成流播放器.Stop();break;}}}void OnAudioRead(float[] data){for (int i = 0; i < data.Length; i++){if (播放队列.Count > 0){data[i] = 播放队列.Dequeue();}else{if (语音合成WebSocket == null || 语音合成WebSocket.State != WebSocketState.Aborted) 语音流总长度++;data[i] = 0;}}}public async void 连接语音合成WebSocket(string urlStr, String text, string voice , int speed, int volume){//ServicePointManager.SecurityProtocol = SecurityProtocolType.Ssl3;using (语音合成WebSocket = new ClientWebSocket()){CancellationToken ct = new CancellationToken();Uri url = new Uri(urlStr);await 语音合成WebSocket.ConnectAsync(url, ct);text = Convert.ToBase64String(Encoding.UTF8.GetBytes(text));JSONNode sendJson = new JSONNode{{"common",new JSONNode{{"app_id", APPID } } },{"business",new JSONNode{{"vcn", voice },{"aue", "raw" },{"speed", speed },{"volume", volume },{"tte", "UTF8" } } },{"data",new JSONNode{{"status", 2 }, {"text", text } } }};Debug.Log("发送消息:" + sendJson);Debug.Log("连接成功");await 语音合成WebSocket.SendAsync(new ArraySegment<byte>(Encoding.UTF8.GetBytes(sendJson)), WebSocketMessageType.Binary, true, ct); //发送数据StringBuilder sb = new StringBuilder();播放队列.Clear();while (语音合成WebSocket.State == WebSocketState.Open){var result = new byte[4096];await 语音合成WebSocket.ReceiveAsync(new ArraySegment<byte>(result), ct);//接受数据List<byte> list = new List<byte>(result); while (list[list.Count - 1] == 0x00) list.RemoveAt(list.Count - 1);//去除空字节 var str = Encoding.UTF8.GetString(list.ToArray());Debug.Log(str);sb.Append(str);if (str.EndsWith("}}")){JSONNode json = JSONNode.Parse(sb.ToString());sb.Clear();Debug.Log("收到完整json数据：" + json);JSONNode data = json["data"];int status = data["status"];float[] fs = bytesToFloat(Convert.FromBase64String(data["audio"]));语音流总长度 += fs.Length;foreach (float f in fs) 播放队列.Enqueue(f);if (status == 2){语音合成WebSocket.Abort();break;}}}}}
原文链接：/chunyu90225/article/details/106172895
