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Test the Bot Using Speech Input (AWS CLI) - Amazon Lex

    https://docs.aws.amazon.com/lex/latest/dg/gs-create-test-speech.html
    Amazon Lex saves the audio from the response in the specified output file. aws lex-runtime post-content \ --region region \ --bot-name OrderFlowersBot \ --bot-alias "\$LATEST" \ --user-id UserOne \ --content-type "audio/l16; rate=16000; channels=1" \ --input-stream IntentSpeech.mpg \ IntentOutputSpeech.mpg.

SpeechRecognizer 2.3 | Alexa Voice Service

    https://developer.amazon.com/en-US/docs/alexa/alexa-voice-service/speechrecognizer.html
    Accepted values: AUDIO_L16_RATE_16000_CHANNELS_1 (PCM), OPUS: string: initiator: Lets Alexa know how an interaction was initiated. Required when a user originates an interaction, such as wake word, tap, push-and-hold. If initiator is present in an ExpectSpeech directive then return the initiator in the following Recognize event.

Microphone Capability Interface | AVS for AWS IoT

    https://developer.amazon.com/en-US/docs/alexa/alexa-voice-service/avs-for-aws-iot-microphone.html
    AUDIO_L16_RATE_16000_CHANNELS_1: 16-bit linear PCM, 16-kHz sample rate, single channel, little-endian byte order string

Voice API > WebSockets | Vonage API Developer

    https://developer.vonage.com/voice/voice-api/guides/websockets
    To choose the sampling rate set the Content-Type property to audio/l16;rate=16000 or audio/l16;rate=8000 depending on if you need the data at 16kHz or 8kHz. Most real-time transcription services work best with audio at 8kHz. Each message will be a 20ms sample of the audio from the call. If you choose the 8kHz rate each message will be 320 bytes.

ASR API | WEB SOCKET | gRPC | HTTP - VoiceLab AI Sp. z o. o.

    https://voicelab.ai/asr-api-web-socket-grpc-http/
    Supported arguments:– audio/l16;rate=8000 – audio samples in PCM 8kHz encoding;– audio/l16;rate=16000 – audio samples in PCM 16kHz encoding; – audio/x-alaw-basic – alaw codec; – audio/basic – mulaw codec; – audio/flac – …

Vonage API Developer

    https://developer.vonage.com/api/voice.v2
    Connect to a Websocket. Must be one of: audio/l16;rate=8000 or audio/l16;rate=16000

Vonage API Developer

    https://developer.nexmo.com/api/voice
    Must be one of: audio/l16;rate=8000 or audio/l16;rate=16000. uri. string | Min: 1 | Max: 50 Connect to a Websocket. headers. object Details of the Websocket you want to connect to. customer_id. string This is an example header. You can …

How to sample audio with 16000 or 8000 Hz rate? · Issue ...

    https://github.com/muaz-khan/RecordRTC/issues/226
    var recorder = new StereoAudioRecorder (stream, {get16BitAudio: true}); // or more versatile to accept multiple values // default options for "sampleRate" are in between 22050 to 96000 // however we can support "16000" or "8000" according to stackoverflow solution var recorder = new StereoAudioRecorder (stream, {sampleRate: 16000});

detecting endianness in streaming speech data, swift ...

    https://intelligentsystemsmonitoring.com/community/ibm-community/detecting-endianness-in-streaming-speech-data-swift/
    dstFormat.mSampleRate = 16000 //Set sample rate dstFormat.mFormatID = kAudioFormatLinearPCM dstFormat.mChannelsPerFrame = 1 dstFormat.mBitsPerChannel = 16 dstFormat.mBytesPerPacket = 2 * dstFormat.mChannelsPerFrame dstFormat.mBytesPerFrame = 2 * dstFormat.mChannelsPerFrame dstFormat.mFramesPerPacket = 1

Capturing Voice Input in a Browser and Sending it to ...

    https://aws.amazon.com/blogs/machine-learning/capturing-voice-input-in-a-browser/
    You need to make sure that the audio buffer is sampled at 16 kHz (more on this later). In my experience the AudioContext sample rate in Chrome and Firefox is 44100Hz (the same sampling rate as used for compact discs (CDs)) but this can vary based on the recording device. The following function down samples the audio buffer to 16 kHz.

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