Transcribe a streaming feed on a local file

Transcribe a streaming audio feed from a microphone on a local file.

Explore further

For detailed documentation that includes this code sample, see the following:

• Transcribe audio from streaming input

Code sample

Go

To learn how to install and use the client library for Cloud STT, see Cloud STT client libraries. For more information, see the Cloud STT Go API reference documentation.

To authenticate to Cloud STT, set up Application Default Credentials. For more information, see Set up authentication for a local development environment.

import (
	"context"
	"flag"
	"fmt"
	"io"
	"log"
	"os"
	"path/filepath"

	speech "cloud.google.com/go/speech/apiv1"
	"cloud.google.com/go/speech/apiv1/speechpb"
)

func main() {
	flag.Usage = func() {
		fmt.Fprintf(os.Stderr, "Usage: %s <AUDIOFILE>\n", filepath.Base(os.Args[0]))
		fmt.Fprintf(os.Stderr, "<AUDIOFILE> must be a path to a local audio file. Audio file must be a 16-bit signed little-endian encoded with a sample rate of 16000.\n")

	}
	flag.Parse()
	if len(flag.Args()) != 1 {
		log.Fatal("Please pass path to your local audio file as a command line argument")
	}
	audioFile := flag.Arg(0)

	ctx := context.Background()

	client, err := speech.NewClient(ctx)
	if err != nil {
		log.Fatal(err)
	}
	stream, err := client.StreamingRecognize(ctx)
	if err != nil {
		log.Fatal(err)
	}
	// Send the initial configuration message.
	if err := stream.Send(&speechpb.StreamingRecognizeRequest{
		StreamingRequest: &speechpb.StreamingRecognizeRequest_StreamingConfig{
			StreamingConfig: &speechpb.StreamingRecognitionConfig{
				Config: &speechpb.RecognitionConfig{
					Encoding:        speechpb.RecognitionConfig_LINEAR16,
					SampleRateHertz: 16000,
					LanguageCode:    "en-US",
				},
			},
		},
	}); err != nil {
		log.Fatal(err)
	}

	f, err := os.Open(audioFile)
	if err != nil {
		log.Fatal(err)
	}
	defer f.Close()

	go func() {
		buf := make([]byte, 1024)
		for {
			n, err := f.Read(buf)
			if n > 0 {
				if err := stream.Send(&speechpb.StreamingRecognizeRequest{
					StreamingRequest: &speechpb.StreamingRecognizeRequest_AudioContent{
						AudioContent: buf[:n],
					},
				}); err != nil {
					log.Printf("Could not send audio: %v", err)
				}
			}
			if err == io.EOF {
				// Nothing else to pipe, close the stream.
				if err := stream.CloseSend(); err != nil {
					log.Fatalf("Could not close stream: %v", err)
				}
				return
			}
			if err != nil {
				log.Printf("Could not read from %s: %v", audioFile, err)
				continue
			}
		}
	}()

	for {
		resp, err := stream.Recv()
		if err == io.EOF {
			break
		}
		if err != nil {
			log.Fatalf("Cannot stream results: %v", err)
		}
		if err := resp.Error; err != nil {
			log.Fatalf("Could not recognize: %v", err)
		}
		for _, result := range resp.Results {
			fmt.Printf("Result: %+v\n", result)
		}
	}
}

Java

To learn how to install and use the client library for Cloud STT, see Cloud STT client libraries. For more information, see the Cloud STT Java API reference documentation.

To authenticate to Cloud STT, set up Application Default Credentials. For more information, see Set up authentication for a local development environment.

/**
 * Performs streaming speech recognition on raw PCM audio data.
 *
 * @param fileName the path to a PCM audio file to transcribe.
 */
public static void streamingRecognizeFile(String fileName) throws Exception, IOException {
  Path path = Paths.get(fileName);
  byte[] data = Files.readAllBytes(path);

  // Instantiates a client with GOOGLE_APPLICATION_CREDENTIALS
  try (SpeechClient speech = SpeechClient.create()) {

    // Configure request with local raw PCM audio
    RecognitionConfig recConfig =
        RecognitionConfig.newBuilder()
            .setEncoding(AudioEncoding.LINEAR16)
            .setLanguageCode("en-US")
            .setSampleRateHertz(16000)
            .setModel("default")
            .build();
    StreamingRecognitionConfig config =
        StreamingRecognitionConfig.newBuilder().setConfig(recConfig).build();

    class ResponseApiStreamingObserver<T> implements ApiStreamObserver<T> {
      private final SettableFuture<List<T>> future = SettableFuture.create();
      private final List<T> messages = new java.util.ArrayList<T>();

      @Override
      public void onNext(T message) {
        messages.add(message);
      }

      @Override
      public void onError(Throwable t) {
        future.setException(t);
      }

      @Override
      public void onCompleted() {
        future.set(messages);
      }

      // Returns the SettableFuture object to get received messages / exceptions.
      public SettableFuture<List<T>> future() {
        return future;
      }
    }

    ResponseApiStreamingObserver<StreamingRecognizeResponse> responseObserver =
        new ResponseApiStreamingObserver<>();

    BidiStreamingCallable<StreamingRecognizeRequest, StreamingRecognizeResponse> callable =
        speech.streamingRecognizeCallable();

    ApiStreamObserver<StreamingRecognizeRequest> requestObserver =
        callable.bidiStreamingCall(responseObserver);

    // The first request must **only** contain the audio configuration:
    requestObserver.onNext(
        StreamingRecognizeRequest.newBuilder().setStreamingConfig(config).build());

    // Subsequent requests must **only** contain the audio data.
    requestObserver.onNext(
        StreamingRecognizeRequest.newBuilder()
            .setAudioContent(ByteString.copyFrom(data))
            .build());

    // Mark transmission as completed after sending the data.
    requestObserver.onCompleted();

    List<StreamingRecognizeResponse> responses = responseObserver.future().get();

    for (StreamingRecognizeResponse response : responses) {
      // For streaming recognize, the results list has one is_final result (if available) followed
      // by a number of in-progress results (if iterim_results is true) for subsequent utterances.
      // Just print the first result here.
      StreamingRecognitionResult result = response.getResultsList().get(0);
      // There can be several alternative transcripts for a given chunk of speech. Just use the
      // first (most likely) one here.
      SpeechRecognitionAlternative alternative = result.getAlternativesList().get(0);
      System.out.printf("Transcript : %s\n", alternative.getTranscript());
    }
  }
}

Node.js

To learn how to install and use the client library for Cloud STT, see Cloud STT client libraries. For more information, see the Cloud STT Node.js API reference documentation.

To authenticate to Cloud STT, set up Application Default Credentials. For more information, see Set up authentication for a local development environment.

const fs = require('fs');

// Imports the Google Cloud client library
const speech = require('@google-cloud/speech');

// Creates a client
const client = new speech.SpeechClient();

/**
 * TODO(developer): Uncomment the following lines before running the sample.
 */
// const filename = 'Local path to audio file, e.g. /path/to/audio.raw';
// const encoding = 'Encoding of the audio file, e.g. LINEAR16';
// const sampleRateHertz = 16000;
// const languageCode = 'BCP-47 language code, e.g. en-US';

const request = {
  config: {
    encoding: encoding,
    sampleRateHertz: sampleRateHertz,
    languageCode: languageCode,
  },
  interimResults: false, // If you want interim results, set this to true
};

// Stream the audio to the Google Cloud Speech API
const recognizeStream = client
  .streamingRecognize(request)
  .on('error', console.error)
  .on('data', data => {
    console.log(
      `Transcription: ${data.results[0].alternatives[0].transcript}`
    );
  });

// Stream an audio file from disk to the Speech API, e.g. "./resources/audio.raw"
fs.createReadStream(filename).pipe(recognizeStream);

PHP

To learn how to install and use the client library for Cloud STT, see Cloud STT client libraries.

To authenticate to Cloud STT, set up Application Default Credentials. For more information, see Set up authentication for a local development environment.

use Google\Cloud\Speech\V2\Client\SpeechClient;
use Google\Cloud\Speech\V2\StreamingRecognizeRequest;
use Google\Cloud\Speech\V2\RecognitionFeatures;
use Google\Cloud\Speech\V2\AutoDetectDecodingConfig;
use Google\Cloud\Speech\V2\RecognitionConfig;
use Google\Cloud\Speech\V2\StreamingRecognitionConfig;

/**
 * @param string $projectId The Google Cloud project ID.
 * @param string $location The location of the recognizer.
 * @param string $recognizerId The ID of the recognizer to use.
 * @param string $audioFile path to an audio file.
 */
function streaming_recognize(string $projectId, string $location, string $recognizerId, string $audioFile)
{
    // create the speech client
    $apiEndpoint = $location === 'global' ? null : sprintf('%s-speech.googleapis.com', $location);
    $speech = new SpeechClient(['apiEndpoint' => $apiEndpoint]);

    $recognizerName = SpeechClient::recognizerName($projectId, $location, $recognizerId);

    // set streaming config
    $features = new RecognitionFeatures([
        'enable_automatic_punctuation' => true
    ]);
    $streamingConfig = (new StreamingRecognitionConfig())
        ->setConfig(new RecognitionConfig([
            // Can also use {@see Google\Cloud\Speech\V2\ExplicitDecodingConfig}
            'auto_decoding_config' => new AutoDetectDecodingConfig(),
            'features' => $features
        ]));
    $streamingRequest = (new StreamingRecognizeRequest())
        ->setRecognizer($recognizerName)
        ->setStreamingConfig($streamingConfig);

    // set the streaming request
    $stream = $speech->streamingRecognize();
    $stream->write($streamingRequest);

    // stream the audio file
    $handle = fopen($audioFile, 'r');
    while (!feof($handle)) {
        $chunk = fread($handle, 4096);
        $streamingRequest = (new StreamingRecognizeRequest())
            ->setAudio($chunk);
        $stream->write($streamingRequest);
    }
    fclose($handle);

    // read the responses
    foreach ($stream->closeWriteAndReadAll() as $response) {
        // an empty response indicates the end of the stream
        if (!$response->getResults()) {
            continue;
        }

        // process the results
        foreach ($response->getResults() as $result) {
            printf(
                'Transcript: "%s"' . PHP_EOL,
                $result->getAlternatives()[0]->getTranscript()
            );
        }
    }
}

Python

To learn how to install and use the client library for Cloud STT, see Cloud STT client libraries. For more information, see the Cloud STT Python API reference documentation.

To authenticate to Cloud STT, set up Application Default Credentials. For more information, see Set up authentication for a local development environment.

def transcribe_streaming(stream_file: str) -> speech.RecognitionConfig:
    """Streams transcription of the given audio file using Google Cloud Speech-to-Text API.
    Args:
        stream_file (str): Path to the local audio file to be transcribed.
            Example: "resources/audio.raw"
    """
    client = speech.SpeechClient()

    with open(stream_file, "rb") as audio_file:
        audio_content = audio_file.read()

    # In practice, stream should be a generator yielding chunks of audio data.
    stream = [audio_content]

    requests = (
        speech.StreamingRecognizeRequest(audio_content=chunk) for chunk in stream
    )

    config = speech.RecognitionConfig(
        encoding=speech.RecognitionConfig.AudioEncoding.LINEAR16,
        sample_rate_hertz=16000,
        language_code="en-US",
    )

    streaming_config = speech.StreamingRecognitionConfig(config=config)

    # streaming_recognize returns a generator.
    responses = client.streaming_recognize(
        config=streaming_config,
        requests=requests,
    )

    for response in responses:
        # Once the transcription has settled, the first result will contain the
        # is_final result. The other results will be for subsequent portions of
        # the audio.
        for result in response.results:
            print(f"Finished: {result.is_final}")
            print(f"Stability: {result.stability}")
            alternatives = result.alternatives
            # The alternatives are ordered from most likely to least.
            for alternative in alternatives:
                print(f"Confidence: {alternative.confidence}")
                print(f"Transcript: {alternative.transcript}")

Ruby

To learn how to install and use the client library for Cloud STT, see Cloud STT client libraries.

To authenticate to Cloud STT, set up Application Default Credentials. For more information, see Set up authentication for a local development environment.

# audio_file_path = "Path to file on which to perform speech recognition"

require "google/cloud/speech"

speech = Google::Cloud::Speech.speech version: :v1

audio_content = File.binread audio_file_path
bytes_total   = audio_content.size
bytes_sent    = 0
chunk_size    = 32_000

input_stream = Gapic::StreamInput.new
output_stream = speech.streaming_recognize input_stream

config = {
  config: {
    encoding:                 :LINEAR16,
    sample_rate_hertz:        16_000,
    language_code:            "en-US",
    enable_word_time_offsets: true
  }
}
input_stream.push streaming_config: config

# Simulated streaming from a microphone
# Stream bytes...
while bytes_sent < bytes_total
  input_stream.push audio_content: audio_content[bytes_sent, chunk_size]
  bytes_sent += chunk_size
  sleep 1
end

puts "Stopped passing"
input_stream.close

results = output_stream

results.each do |result|
  puts "Transcript: #{result}"
end

What's next

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