文档建议反馈控制台
首页
学习
活动
专区
圈层
工具
MCP广场
文章/答案/技术大牛
发布
社区首页 >问答首页 >无法创建Sinch客户端

问无法创建Sinch客户端
EN

Stack Overflow用户
提问于 2019-09-19 19:53:30
回答 1查看 434关注 0票数 0

我只是想在我的应用程序中使用sinch进行通话，到目前为止，我已经关注了那里的官方documentation，所以按照这个链接，我正在构建Sinch客户端，如下所示：

代码语言：javascript
复制
 private var sinchClient: SinchClient? = null

 private fun initSinchClient() {
    sinchClient = Sinch.getSinchClientBuilder().context(this@CallNewActivity)
        .applicationKey(APP_KEY)
        .applicationSecret(APP_SECRET)
        .environmentHost(ENVIRONMENT)
        .userId("uName1234")
        .build()

    sinchClient!!.checkManifest()
}

然后什么都没有发生，问题是我的应用程序在执行完这段代码后就崩溃了！例外情况是：

代码语言：javascript
复制
     ----- class 'Lorg/webrtc/voiceengine/WebRtcAudioManager;' cl=0x134c27e0 -----
   objectSize=194 (172 from super)
   access=0x0008.0001
   super='java.lang.Class<java.lang.Object>' (cl=0x0)
   vtable (1 entries, 11 in super):
      0: boolean org.webrtc.voiceengine.WebRtcAudioManager.isLowLatencyInputSupported()
   direct methods (26 entries):
      0: void org.webrtc.voiceengine.WebRtcAudioManager.<clinit>()
      1: void org.webrtc.voiceengine.WebRtcAudioManager.<init>(long)
      2: void org.webrtc.voiceengine.WebRtcAudioManager.assertTrue(boolean)
      3: void org.webrtc.voiceengine.WebRtcAudioManager.dispose()
      4: int org.webrtc.voiceengine.WebRtcAudioManager.getLowLatencyInputFramesPerBuffer()
      5: int org.webrtc.voiceengine.WebRtcAudioManager.getLowLatencyOutputFramesPerBuffer()
      6: int org.webrtc.voiceengine.WebRtcAudioManager.getMinInputFrameSize(int, int)
      7: int org.webrtc.voiceengine.WebRtcAudioManager.getMinOutputFrameSize(int, int)
      8: int org.webrtc.voiceengine.WebRtcAudioManager.getNativeOutputSampleRate()
      9: int org.webrtc.voiceengine.WebRtcAudioManager.getSampleRateOnJellyBeanMR10OrHigher()
     10: boolean org.webrtc.voiceengine.WebRtcAudioManager.getStereoInput()
     11: boolean org.webrtc.voiceengine.WebRtcAudioManager.getStereoOutput()
     12: boolean org.webrtc.voiceengine.WebRtcAudioManager.hasEarpiece()
     13: boolean org.webrtc.voiceengine.WebRtcAudioManager.init()
     14: boolean org.webrtc.voiceengine.WebRtcAudioManager.isAAudioSupported()
     15: boolean org.webrtc.voiceengine.WebRtcAudioManager.isAcousticEchoCancelerSupported()
     16: boolean org.webrtc.voiceengine.WebRtcAudioManager.isCommunicationModeEnabled()
     17: boolean org.webrtc.voiceengine.WebRtcAudioManager.isDeviceBlacklistedForOpenSLESUsage()
     18: boolean org.webrtc.voiceengine.WebRtcAudioManager.isLowLatencyOutputSupported()
     19: boolean org.webrtc.voiceengine.WebRtcAudioManager.isNoiseSuppressorSupported()
     20: boolean org.webrtc.voiceengine.WebRtcAudioManager.isProAudioSupported()
     21: void org.webrtc.voiceengine.WebRtcAudioManager.nativeCacheAudioParameters(int, int, int, boolean, boolean, boolean, boolean, boolean, boolean, boolean, int, int, long)
     22: void org.webrtc.voiceengine.WebRtcAudioManager.setBlacklistDeviceForOpenSLESUsage(boolean)
     23: void org.webrtc.voiceengine.WebRtcAudioManager.setStereoInput(boolean)
     24: void org.webrtc.voiceengine.WebRtcAudioManager.setStereoOutput(boolean)
     25: void org.webrtc.voiceengine.WebRtcAudioManager.storeAudioParameters()
   static fields (9 entries):
      0: int org.webrtc.voiceengine.WebRtcAudioManager.BITS_PER_SAMPLE
      1: boolean org.webrtc.voiceengine.WebRtcAudioManager.DEBUG
      2: int org.webrtc.voiceengine.WebRtcAudioManager.DEFAULT_FRAME_PER_BUFFER
      3: java.lang.String org.webrtc.voiceengine.WebRtcAudioManager.TAG
      4: boolean org.webrtc.voiceengine.WebRtcAudioManager.blacklistDeviceForAAudioUsage
      5: boolean org.webrtc.voiceengine.WebRtcAudioManager.blacklistDeviceForOpenSLESUsage
      6: boolean org.webrtc.voiceengine.WebRtcAudioManager.blacklistDeviceForOpenSLESUsageIsOverridden
      7: boolean org.webrtc.voiceengine.WebRtcAudioManager.useStereoInput
      8: boolean org.webrtc.voiceengine.WebRtcAudioManager.useStereoOutput
   instance fields (18 entries):
      0: boolean org.webrtc.voiceengine.WebRtcAudioManager.aAudio
      1: android.media.AudioManager org.webrtc.voiceengine.WebRtcAudioManager.audioManager
      2: boolean org.webrtc.voiceengine.WebRtcAudioManager.hardwareAEC
      3: boolean org.webrtc.voiceengine.WebRtcAudioManager.hardwareAGC
      4: boolean org.webrtc.voiceengine.WebRtcAudioManager.hardwareNS
      5: boolean org.webrtc.voiceengine.WebRtcAudioManager.initialized
      6: int org.webrtc.voiceengine.WebRtcAudioManager.inputBufferSize
      7: int org.webrtc.voiceengine.WebRtcAudioManager.inputChannels
      8: boolean org.webrtc.voiceengine.WebRtcAudioManager.lowLatencyInput
     9: boolean org.webrtc.voiceengine.WebRtcAudioManager.lowLatencyOutput
    10: long org.webrtc.voiceengine.WebRtcAudioManager.nativeAudioManager
    11: int org.webrtc.voiceengine.WebRtcAudioManager.nativeChannels
    12: int org.webrtc.voiceengine.WebRtcAudioManager.nativeSampleRate
    13: int org.webrtc.voiceengine.WebRtcAudioManager.outputBufferSize
    14: int org.webrtc.voiceengine.WebRtcAudioManager.outputChannels
    15: boolean org.webrtc.voiceengine.WebRtcAudioManager.proAudio
    16: int org.webrtc.voiceengine.WebRtcAudioManager.sampleRate
    17: org.webrtc.voiceengine.WebRtcAudioManager$VolumeLogger org.webrtc.voiceengine.WebRtcAudioManager.volumeLogger
Failed to register native method org.webrtc.voiceengine.WebRtcAudioManager.nativeCacheAudioParameters(IIIZZZZZZIIJ)V in /data/app/com.meftii.doctor.e.visit-xRk9IdVgiR8jI85gaiv7CQ==/base.apk!classes3.dex
2019-09-19 15:41:44.987 9189-9308/com.meftii.doctor.e.visit E/rtc: #
    # Fatal error in ../../../modules/utility/source/jvm_android.cc, line 200
    # last system error: 0
    # Check failed: !jni_->ExceptionCheck()
    # Error during RegisterNatives
    #


    --------- beginning of crash
2019-09-19 15:41:44.988 9189-9308/com.meftii.doctor.e.visit A/libc: Fatal signal 6 (SIGABRT), code -6 (SI_TKILL) in tid 9308 (Thread-15), pid 9189 (.doctor.e.visit)

那么，对于这个问题，有人能找出我做错了什么吗?这个异常的目的是什么？我是第一次集成这个sdk，也浏览了官方资源；但我找不到任何导致这次崩溃的原因。提前感谢

java
android
kotlin
sinch
android-sinch-api
EN

回答 1

Stack Overflow用户

发布于 2019-09-28 05:54:45

您应该从运行Android示例Calling应用程序开始。在SDK包中包含的Samples文件夹中可用。

https://download.sinch.com/android/3.15.0/sinch-android-rtc-3.15.0.zip

下面是一个类似的示例代码，它工作得很好。

代码语言：javascript
复制
mSinchClient = Sinch.getSinchClientBuilder()
            .context(getApplicationContext())
            .userId("uName1234")
            .applicationKey(APP_KEY)
            .applicationSecret(APP_SECRET)
            .environmentHost(ENVIRONMENT).build();

Sinch语音和视频团队

票数 0
EN
页面原文内容由Stack Overflow提供。腾讯云小微IT领域专用引擎提供翻译支持
原文链接：

https://stackoverflow.com/questions/58010435

复制
相关文章

相似问题

领券

腾讯云开发者

扫码关注腾讯云开发者

扫码关注腾讯云开发者

领取腾讯云代金券

热门产品

热门推荐

更多推荐

Copyright © 2013 - 2025 Tencent Cloud. All Rights Reserved. 腾讯云 版权所有 

深圳市腾讯计算机系统有限公司 ICP备案/许可证号：粤B2-20090059 深公网安备号 44030502008569

腾讯云计算（北京）有限责任公司 京ICP证150476号 |  京ICP备11018762号 | 京公网安备号11010802020287

问题归档专栏文章快讯文章归档关键词归档开发者手册归档开发者手册 Section 归档