语音合成(webSocket版)

前提条件

• 确保已经按照配置好Android开发环境。
• 请参考SDK（websocket）获取最新版本SDK包。

初始化Client

初始化RttsClient，其中参数包含AuthInfo和SisHttpConfig和RttsResponseListener。其中AuthInfo和SisHttpConfig的参数如表1所示。

其中RttsResponseListener是用户自定义的，建立webSocket之后，接收服务端返回消息的Listener。

请求参数

请求类为RttsRequest，其中参数包含text,command,Config,详见表5。

建立连接，发送合成数据

rttsClient.connect();
rttsClient.sendData(rttsRequest);

发送关闭连接请求

rttsClient.close();

代码示例

如下示例仅供参考，最新代码请前往SDK（websocket）章节获取并运行。

/*
 * Copyright (c) Huawei Technologies Co., Ltd. 2022-2022. All rights reserved.
 */

package com.huaweicloud.sis.android.demo.tts;

import android.os.Bundle;
import android.util.Log;
import android.view.View;
import android.widget.Button;
import android.widget.EditText;
import android.widget.Toast;

import androidx.annotation.Nullable;
import androidx.appcompat.app.AppCompatActivity;

import com.cloud.sdk.util.StringUtils;
import com.huaweicloud.sdk.core.utils.JsonUtils;
import com.huaweicloud.sis.android.demo.R;
import com.huaweicloud.sis.android.demo.service.AudioTrackService;
import com.huaweicloud.sis.android.demo.Config;
import com.huaweicloud.sis.android.demo.service.AudioTrackServiceCallback;

import sis.android.sdk.RttsClient;
import sis.android.sdk.bean.AuthInfo;
import sis.android.sdk.bean.SisHttpConfig;
import sis.android.sdk.bean.request.RttsRequest;
import sis.android.sdk.bean.response.RttsResponse;
import sis.android.sdk.exception.SisException;
import sis.android.sdk.listeners.RttsResponseListener;

/**
 * 功能描述
 * 语音合成 websocket
 *
 * @since 2022-07-18
 */
public class RttsActivity extends AppCompatActivity {
	private EditText textView;
	private Button startPlay;

	private RttsClient rttsClient;

	private SisHttpConfig sisHttpConfig;

	private AudioTrackService audioTrackService;
	private AudioTrackServiceCallback audioPlayerCallback;

	private AuthInfo authInfo;

	private static int dataAcceptanceTime = 1000 * 60 * 5;

	private RttsResponseListener rttsResponseListener = new RttsResponseListener() {
		@Override
		public void onTranscriptionConnect() {
			Log.i("info", "建立连接后回调");
		}

		@Override
		public void onTranscriptionClose() {
			Log.i("info", "关闭连接后回调");
		}

		@Override
		public void onTranscriptionFail(RttsResponse response) {
			runOnUiThread(new Runnable() {
				@Override
				public void run() {
					Toast.makeText(getApplicationContext(), "出现异常" + JsonUtils.toJSON(response), Toast.LENGTH_SHORT).show();
				}
			});
			updateButtonState(startPlay, true);
			Log.i("info", "长连接失败后回调" + JsonUtils.toJSON(response));
		}

		@Override
		public void onTranscriptionBegin(RttsResponse response) {
			Log.i("info", "开始合成时的响应事件" + response.toString());
		}

		@Override
		public void onTranscriptionEnd(RttsResponse response) {
			rttsClient.close();
			audioTrackService.setPlayState(AudioTrackService.PlayState.playEnd);
		}

		@Override
		public void onTranscriptionError(RttsResponse response) {
			runOnUiThread(new Runnable() {
				@Override
				public void run() {
					Toast.makeText(getApplicationContext(), "出现异常" + JsonUtils.toJSON(response), Toast.LENGTH_SHORT).show();
				}
			});
			updateButtonState(startPlay, true);
			Log.i("info", "合成时发生错误的响应事件" + response.toString());
		}

		@Override
		public void onTranscriptionResponse(byte[] bytes) {
			Log.i("info", "合成过程中返回的二进制流");
			audioTrackService.setAudioData(bytes);
		}
	};

	@Override
	protected void onCreate(@Nullable Bundle savedInstanceState) {
		super.onCreate(savedInstanceState);
		setContentView(R.layout.rtts);
	}

	@Override
	protected void onStart() {
		super.onStart();
		initView();
		initResources();
	}


	@Override
	protected void onDestroy() {
		super.onDestroy();
		if (audioTrackService != null) {
			audioTrackService.stop();
			audioTrackService.releaseTrack();
		}
	}

	/**
	 * 初始化UI
	 */
	private void initView() {
		textView = findViewById(R.id.itext);
		startPlay = findViewById(R.id.startplay);
		startPlay.setOnClickListener(new View.OnClickListener() {
			@Override
			public void onClick(View view) {
				updateButtonState(startPlay, false);
				new Thread(new Runnable() {
					@Override
					public void run() {
						try {
							audioTrackService.play();
							rttsClient = new RttsClient(authInfo, sisHttpConfig, rttsResponseListener);
							RttsRequest rttsRequest = rttsRequets();
							rttsClient.connect();
							rttsClient.sendData(rttsRequest);
						} catch (SisException e) {
							Log.e("error", e.getErrorCode() + e.getErrorMsg());
						}
					}
				}).start();
			}
		});
	}

	// 用于设置按钮的状态
	private void updateButtonState(final Button btn, final boolean state) {
		runOnUiThread(new Runnable() {
			@Override
			public void run() {
				btn.setEnabled(state);
			}
		});
	}

	/**
	 * 初始化资源
	 */
	private void initResources() {
		authInfo = new AuthInfo(this.getString(R.string.HUAWEICLOUD_SDK_AK), this.getString(R.string.HUAWEICLOUD_SDK_SK), 
Config.REGION, Config.PROJECT_ID);
		sisHttpConfig = new SisHttpConfig();
		// 设置等待事件为5分钟
		sisHttpConfig.setWebsocketWaitTimeout(dataAcceptanceTime);
		audioPlayerCallback = new AudioTrackServiceCallback() {
			@Override
			public void playStart() {

			}

			@Override
			public void playOver() {
				updateButtonState(startPlay, true);
			}
		};
		audioTrackService = new AudioTrackService(audioPlayerCallback);
	}


	/**
	 * 功能描述
	 * 语音合成websocket 版
	 */
	private RttsRequest rttsRequets() {
		RttsRequest rttsRequest = new RttsRequest();
		RttsRequest.Config config = new RttsRequest.Config();
		config.setAudioFormat("pcm");
		config.setProperty("chinese_huaxiaomei_common");
		config.setSampleRate("16000");
		config.setPitch(0);
		config.setVolume(50);
		rttsRequest.setCommand("START");
		rttsRequest.setConfig(config);
		String text = textView.getText().toString().trim();
		if (!StringUtils.isNullOrEmpty(text)) {
			rttsRequest.setText(text);
		} else {
			rttsRequest.setText("请输入合成音频");
		}
		return rttsRequest;
	}
}