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nimbleape / dana-tsg-rtp-stt-audioserver

An AudioServer that takes audio from Asterisk via UDP and sends it to Google's Speech To Text Engine
MIT License
32 stars 10 forks source link

Error from Google #1

Open danjenkins opened 4 years ago

danjenkins commented 4 years ago 
[1575391935639] INFO  (Dana-AudioServer/57627 on Dans-MacBook-Pro-7.local): Is recognizeStream destroyed? false
[1575391935669] INFO  (Dana-AudioServer/57627 on Dans-MacBook-Pro-7.local): Is recognizeStream destroyed? false
[1575391935699] INFO  (Dana-AudioServer/57627 on Dans-MacBook-Pro-7.local): Is recognizeStream destroyed? false
[1575391935699] INFO  (Dana-AudioServer/57627 on Dans-MacBook-Pro-7.local): Is recognizeStream destroyed? false
[1575391935722] ERROR (Dana-AudioServer/57627 on Dans-MacBook-Pro-7.local): Google API request error
    err: {
      "type": "NodeError",
      "message": "Cannot call write after a stream was destroyed",
      "stack":
          Error: Cannot call write after a stream was destroyed
              at doWrite (/Users/danjenkins/git/Dana-RTP-STT-AudioServer/node_modules/pumpify/node_modules/readable-stream/lib/_stream_writable.js:405:38)
              at writeOrBuffer (/Users/danjenkins/git/Dana-RTP-STT-AudioServer/node_modules/pumpify/node_modules/readable-stream/lib/_stream_writable.js:394:5)
              at Pumpify.Writable.write (/Users/danjenkins/git/Dana-RTP-STT-AudioServer/node_modules/pumpify/node_modules/readable-stream/lib/_stream_writable.js:303:11)
              at Pumpify.Duplexify.end (/Users/danjenkins/git/Dana-RTP-STT-AudioServer/node_modules/pumpify/node_modules/duplexify/index.js:234:41)
              at PassThrough.<anonymous> (/Users/danjenkins/git/Dana-RTP-STT-AudioServer/node_modules/pumpify/node_modules/duplexify/index.js:22:40)
              at PassThrough.f (/Users/danjenkins/git/Dana-RTP-STT-AudioServer/node_modules/once/once.js:25:25)
              at PassThrough.onfinish (/Users/danjenkins/git/Dana-RTP-STT-AudioServer/node_modules/end-of-stream/index.js:31:27)
              at emitNone (events.js:111:20)
              at PassThrough.emit (events.js:208:7)
              at finishMaybe (_stream_writable.js:613:14)
      "name": "Error",
      "code": "ERR_STREAM_DESTROYED"
    }
events.js:183
      throw er; // Unhandled 'error' event
      ^

Error: 4 DEADLINE_EXCEEDED: Deadline exceeded
    at callErrorFromStatus (/Users/danjenkins/git/Dana-RTP-STT-AudioServer/node_modules/@grpc/grpc-js/build/src/call.js:30:26)
    at Http2CallStream.call.on (/Users/danjenkins/git/Dana-RTP-STT-AudioServer/node_modules/@grpc/grpc-js/build/src/call.js:79:34)
    at emitOne (events.js:121:20)
    at Http2CallStream.emit (events.js:211:7)
    at process.nextTick (/Users/danjenkins/git/Dana-RTP-STT-AudioServer/node_modules/@grpc/grpc-js/build/src/call-stream.js:75:22)
    at _combinedTickCallback (internal/process/next_tick.js:132:7)
    at process._tickCallback (internal/process/next_tick.js:181:9)
✨  Done in 82.19s.