> ## Documentation Index
> Fetch the complete documentation index at: https://docs.vocode.dev/llms.txt
> Use this file to discover all available pages before exploring further.

# Transcriber Reference

# `TranscriberConfig`

<ParamField body="sampling_rate" type="int">
  The sampling rate of the audio in samples per second (Hz). A higher sampling
  rate provides better audio quality but may increase processing time and data
  size.
</ParamField>

<ParamField body="audio_encoding" type="AudioEncoding">
  The encoding format of the audio data. Options include: LINEAR16, MULAW.
</ParamField>

<ParamField body="chunk_size" type="int">
  The size of each chunk of audio data sent to the transcriber, in bytes. A
  larger chunk size can reduce network overhead but may increase latency.
</ParamField>

<ParamField body="endpointing_config" type="Optional[EndpointingConfig]">
  Optional endpointing configuration to determine when a transcription segment should end. If not provided, default endpointing behavior will be used.
</ParamField>

<ParamField body="downsampling" type="Optional[int]">
  Optional downsampling factor to reduce the sampling rate of the audio before sending to the transcriber. Can be used to reduce bandwidth usage.
</ParamField>

<ParamField body="min_interrupt_confidence" type="Optional[float]">
  Optional minimum confidence threshold for interrupting the transcription.
  Confidence values range from 0 to 1, with higher values indicating greater
  confidence. If provided, transcriptions will only be interrupted when the
  confidence exceeds the threshold. If not provided, the default interrupting
  behavior will be used.
</ParamField>

<ParamField body="mute_during_speech" type="bool">
  If true, silence audio chunks will be sent to the transcriber while a transcription is in progress. Can be used to prevent echo during live transcription.
</ParamField>

# `DeepgramTranscriberConfig`

<ParamField body="language" type="Optional[str]">
  Optional language code to use for transcription.
</ParamField>

<ParamField body="model" type="Optional[str]">
  Optional Deepgram model to use for transcription. Defaults to "nova".
</ParamField>

<ParamField body="tier" type="Optional[str]">
  Optional Deepgram tier to use for transcription.
</ParamField>

<ParamField body="version" type="Optional[str]">
  Optional Deepgram version to use for transcription. Defaults to latest version if not provided.
</ParamField>

<ParamField body="keywords" type="Optional[List[str]]">
  Optional list of keywords to boost in the transcription results.
</ParamField>

# `GoogleTranscriberConfig`

<ParamField body="model" type="Optional[str]">
  Optional Google Cloud Speech model to use for transcription.
</ParamField>

<ParamField body="language_code" type="str">
  Language code to use for transcription. Defaults to "en-US".
</ParamField>

# `AssemblyAITranscriberConfig`

<ParamField body="buffer_size_seconds" type="float">
  Buffer duration in seconds to accumulate audio before sending to AssemblyAI. Defaults to 0.1s.
</ParamField>

<ParamField body="word_boost" type="Optional[List[str]]">
  Optional list of words to boost in the transcription results.
</ParamField>

# `WhisperCPPTranscriberConfig`

<ParamField body="buffer_size_seconds" type="float">
  Buffer duration in seconds to accumulate audio before sending to WhisperCPP. Defaults to 1.0s.
</ParamField>

<ParamField body="libname" type="str">
  Filename of the WhisperCPP shared library.
</ParamField>

<ParamField body="fname_model" type="str">
  Filename of the WhisperCPP model.
</ParamField>

# `AzureTranscriberConfig`

<ParamField body="language" type="str">
  Language code to use for transcription. Defaults to "en-US".
</ParamField>

<ParamField body="candidate_languages" type="Optional[List[str]]">
  Optional list of candidate languages to auto-detect from the audio.
</ParamField>

# `GladiaTranscriberConfig`

<ParamField body="buffer_size_seconds" type="float">
  Buffer duration in seconds to accumulate audio before sending to Gladia. Defaults to 0.1s.
</ParamField>