Class: Google::Cloud::Speech::V1p1beta1::RecognitionConfig

Inherits:

Object

• Object
• Google::Cloud::Speech::V1p1beta1::RecognitionConfig

Extended by:

Protobuf::MessageExts::ClassMethods

Includes:

Protobuf::MessageExts

Defined in:

proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb

Overview

Provides information to the recognizer that specifies how to process the request.

Defined Under Namespace

Modules: AudioEncoding

Instance Attribute Summary

• #adaptation ⇒ ::Google::Cloud::Speech::V1p1beta1::SpeechAdaptation

  Speech adaptation configuration improves the accuracy of speech recognition.

• #alternative_language_codes ⇒ ::Array<::String>

  A list of up to 3 additional BCP-47 language tags, listing possible alternative languages of the supplied audio.

• #audio_channel_count ⇒ ::Integer

  The number of channels in the input audio data.

• #diarization_config ⇒ ::Google::Cloud::Speech::V1p1beta1::SpeakerDiarizationConfig

  Config to enable speaker diarization and set additional parameters to make diarization better suited for your application.

• #diarization_speaker_count ⇒ ::Integer deprecated Deprecated.

  This field is deprecated and may be removed in the next major version update.

• #enable_automatic_punctuation ⇒ ::Boolean

  If 'true', adds punctuation to recognition result hypotheses.

• #enable_separate_recognition_per_channel ⇒ ::Boolean

  This needs to be set to true explicitly and audio_channel_count > 1 to get each channel recognized separately.

• #enable_speaker_diarization ⇒ ::Boolean deprecated Deprecated.

  This field is deprecated and may be removed in the next major version update.

• #enable_spoken_emojis ⇒ ::Google::Protobuf::BoolValue

  The spoken emoji behavior for the call If not set, uses default behavior based on model of choice If 'true', adds spoken emoji formatting for the request.

• #enable_spoken_punctuation ⇒ ::Google::Protobuf::BoolValue

  The spoken punctuation behavior for the call If not set, uses default behavior based on model of choice e.g.

• #enable_word_confidence ⇒ ::Boolean

  If true, the top result includes a list of words and the confidence for those words.

• #enable_word_time_offsets ⇒ ::Boolean

  If true, the top result includes a list of words and the start and end time offsets (timestamps) for those words.

• #encoding ⇒ ::Google::Cloud::Speech::V1p1beta1::RecognitionConfig::AudioEncoding

  Encoding of audio data sent in all RecognitionAudio messages.

• #language_code ⇒ ::String

  Required.

• #max_alternatives ⇒ ::Integer

  Maximum number of recognition hypotheses to be returned.

• #metadata ⇒ ::Google::Cloud::Speech::V1p1beta1::RecognitionMetadata

  Metadata regarding this request.

• #model ⇒ ::String

  Which model to select for the given request.

• #profanity_filter ⇒ ::Boolean

  If set to true, the server will attempt to filter out profanities, replacing all but the initial character in each filtered word with asterisks, e.g.

• #sample_rate_hertz ⇒ ::Integer

  Sample rate in Hertz of the audio data sent in all RecognitionAudio messages.

• #speech_contexts ⇒ ::Array<::Google::Cloud::Speech::V1p1beta1::SpeechContext>

  Array of SpeechContext.

• #transcript_normalization ⇒ ::Google::Cloud::Speech::V1p1beta1::TranscriptNormalization

  Use transcription normalization to automatically replace parts of the transcript with phrases of your choosing.

• #use_enhanced ⇒ ::Boolean

  Set to true to use an enhanced model for speech recognition.

Instance Attribute Details

#adaptation ⇒ ::Google::Cloud::Speech::V1p1beta1::SpeechAdaptation

Returns Speech adaptation configuration improves the accuracy of speech recognition. For more information, see the speech adaptation documentation. When speech adaptation is set it supersedes the speech_contexts field.

Returns:

• (::Google::Cloud::Speech::V1p1beta1::SpeechAdaptation) —

  Speech adaptation configuration improves the accuracy of speech recognition. For more information, see the speech adaptation documentation. When speech adaptation is set it supersedes the speech_contexts field.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#alternative_language_codes ⇒ ::Array<::String>

Returns A list of up to 3 additional BCP-47 language tags, listing possible alternative languages of the supplied audio. See Language Support for a list of the currently supported language codes. If alternative languages are listed, recognition result will contain recognition in the most likely language detected including the main language_code. The recognition result will include the language tag of the language detected in the audio. Note: This feature is only supported for Voice Command and Voice Search use cases and performance may vary for other use cases (e.g., phone call transcription).

Returns:

• (::Array<::String>) —

  A list of up to 3 additional BCP-47 language tags, listing possible alternative languages of the supplied audio. See Language Support for a list of the currently supported language codes. If alternative languages are listed, recognition result will contain recognition in the most likely language detected including the main language_code. The recognition result will include the language tag of the language detected in the audio. Note: This feature is only supported for Voice Command and Voice Search use cases and performance may vary for other use cases (e.g., phone call transcription).

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#audio_channel_count ⇒ ::Integer

Returns The number of channels in the input audio data. ONLY set this for MULTI-CHANNEL recognition. Valid values for LINEAR16, OGG_OPUS and FLAC are 1-8. Valid value for MULAW, AMR, AMR_WB and SPEEX_WITH_HEADER_BYTE is only 1. If 0 or omitted, defaults to one channel (mono). Note: We only recognize the first channel by default. To perform independent recognition on each channel set enable_separate_recognition_per_channel to 'true'.

Returns:

• (::Integer) —

  The number of channels in the input audio data. ONLY set this for MULTI-CHANNEL recognition. Valid values for LINEAR16, OGG_OPUS and FLAC are 1-8. Valid value for MULAW, AMR, AMR_WB and SPEEX_WITH_HEADER_BYTE is only 1. If 0 or omitted, defaults to one channel (mono). Note: We only recognize the first channel by default. To perform independent recognition on each channel set enable_separate_recognition_per_channel to 'true'.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#diarization_config ⇒ ::Google::Cloud::Speech::V1p1beta1::SpeakerDiarizationConfig

Returns Config to enable speaker diarization and set additional parameters to make diarization better suited for your application. Note: When this is enabled, we send all the words from the beginning of the audio for the top alternative in every consecutive STREAMING responses. This is done in order to improve our speaker tags as our models learn to identify the speakers in the conversation over time. For non-streaming requests, the diarization results will be provided only in the top alternative of the FINAL SpeechRecognitionResult.

Returns:

• (::Google::Cloud::Speech::V1p1beta1::SpeakerDiarizationConfig) —

  Config to enable speaker diarization and set additional parameters to make diarization better suited for your application. Note: When this is enabled, we send all the words from the beginning of the audio for the top alternative in every consecutive STREAMING responses. This is done in order to improve our speaker tags as our models learn to identify the speakers in the conversation over time. For non-streaming requests, the diarization results will be provided only in the top alternative of the FINAL SpeechRecognitionResult.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#diarization_speaker_count ⇒ ::Integer

Deprecated.

This field is deprecated and may be removed in the next major version update.

Returns If set, specifies the estimated number of speakers in the conversation. Defaults to '2'. Ignored unless enable_speaker_diarization is set to true. Note: Use diarization_config instead.

Returns:

• (::Integer) —

  If set, specifies the estimated number of speakers in the conversation. Defaults to '2'. Ignored unless enable_speaker_diarization is set to true. Note: Use diarization_config instead.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#enable_automatic_punctuation ⇒ ::Boolean

Returns If 'true', adds punctuation to recognition result hypotheses. This feature is only available in select languages. Setting this for requests in other languages has no effect at all. The default 'false' value does not add punctuation to result hypotheses.

Returns:

• (::Boolean) —

  If 'true', adds punctuation to recognition result hypotheses. This feature is only available in select languages. Setting this for requests in other languages has no effect at all. The default 'false' value does not add punctuation to result hypotheses.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#enable_separate_recognition_per_channel ⇒ ::Boolean

Returns This needs to be set to true explicitly and audio_channel_count > 1 to get each channel recognized separately. The recognition result will contain a channel_tag field to state which channel that result belongs to. If this is not true, we will only recognize the first channel. The request is billed cumulatively for all channels recognized: audio_channel_count multiplied by the length of the audio.

Returns:

• (::Boolean) —

  This needs to be set to true explicitly and audio_channel_count > 1 to get each channel recognized separately. The recognition result will contain a channel_tag field to state which channel that result belongs to. If this is not true, we will only recognize the first channel. The request is billed cumulatively for all channels recognized: audio_channel_count multiplied by the length of the audio.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#enable_speaker_diarization ⇒ ::Boolean

Deprecated.

This field is deprecated and may be removed in the next major version update.

Returns If 'true', enables speaker detection for each recognized word in the top alternative of the recognition result using a speaker_tag provided in the WordInfo. Note: Use diarization_config instead.

Returns:

• (::Boolean) —

  If 'true', enables speaker detection for each recognized word in the top alternative of the recognition result using a speaker_tag provided in the WordInfo. Note: Use diarization_config instead.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#enable_spoken_emojis ⇒ ::Google::Protobuf::BoolValue

Returns The spoken emoji behavior for the call If not set, uses default behavior based on model of choice If 'true', adds spoken emoji formatting for the request. This will replace spoken emojis with the corresponding Unicode symbols in the final transcript. If 'false', spoken emojis are not replaced.

Returns:

• (::Google::Protobuf::BoolValue) —

  The spoken emoji behavior for the call If not set, uses default behavior based on model of choice If 'true', adds spoken emoji formatting for the request. This will replace spoken emojis with the corresponding Unicode symbols in the final transcript. If 'false', spoken emojis are not replaced.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#enable_spoken_punctuation ⇒ ::Google::Protobuf::BoolValue

Returns The spoken punctuation behavior for the call If not set, uses default behavior based on model of choice e.g. command_and_search will enable spoken punctuation by default If 'true', replaces spoken punctuation with the corresponding symbols in the request. For example, "how are you question mark" becomes "how are you?". See https://cloud.google.com/speech-to-text/docs/spoken-punctuation for support. If 'false', spoken punctuation is not replaced.

Returns:

• (::Google::Protobuf::BoolValue) —

  The spoken punctuation behavior for the call If not set, uses default behavior based on model of choice e.g. command_and_search will enable spoken punctuation by default If 'true', replaces spoken punctuation with the corresponding symbols in the request. For example, "how are you question mark" becomes "how are you?". See https://cloud.google.com/speech-to-text/docs/spoken-punctuation for support. If 'false', spoken punctuation is not replaced.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#enable_word_confidence ⇒ ::Boolean

Returns If true, the top result includes a list of words and the confidence for those words. If false, no word-level confidence information is returned. The default is false.

Returns:

• (::Boolean) —

  If true, the top result includes a list of words and the confidence for those words. If false, no word-level confidence information is returned. The default is false.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#enable_word_time_offsets ⇒ ::Boolean

Returns If true, the top result includes a list of words and the start and end time offsets (timestamps) for those words. If false, no word-level time offset information is returned. The default is false.

Returns:

• (::Boolean) —

  If true, the top result includes a list of words and the start and end time offsets (timestamps) for those words. If false, no word-level time offset information is returned. The default is false.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#encoding ⇒ ::Google::Cloud::Speech::V1p1beta1::RecognitionConfig::AudioEncoding

Returns Encoding of audio data sent in all RecognitionAudio messages. This field is optional for FLAC and WAV audio files and required for all other audio formats. For details, see AudioEncoding.

Returns:

• (::Google::Cloud::Speech::V1p1beta1::RecognitionConfig::AudioEncoding) —

  Encoding of audio data sent in all RecognitionAudio messages. This field is optional for FLAC and WAV audio files and required for all other audio formats. For details, see AudioEncoding.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#language_code ⇒ ::String

Returns Required. The language of the supplied audio as a BCP-47 language tag. Example: "en-US". See Language Support for a list of the currently supported language codes.

Returns:

• (::String) —

  Required. The language of the supplied audio as a BCP-47 language tag. Example: "en-US". See Language Support for a list of the currently supported language codes.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#max_alternatives ⇒ ::Integer

Returns Maximum number of recognition hypotheses to be returned. Specifically, the maximum number of SpeechRecognitionAlternative messages within each SpeechRecognitionResult. The server may return fewer than max_alternatives. Valid values are 0-30. A value of 0 or 1 will return a maximum of one. If omitted, will return a maximum of one.

Returns:

• (::Integer) —

  Maximum number of recognition hypotheses to be returned. Specifically, the maximum number of SpeechRecognitionAlternative messages within each SpeechRecognitionResult. The server may return fewer than max_alternatives. Valid values are 0-30. A value of 0 or 1 will return a maximum of one. If omitted, will return a maximum of one.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#metadata ⇒ ::Google::Cloud::Speech::V1p1beta1::RecognitionMetadata

Returns Metadata regarding this request.

Returns:

• (::Google::Cloud::Speech::V1p1beta1::RecognitionMetadata) —

  Metadata regarding this request.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#model ⇒ ::String

Returns Which model to select for the given request. Select the model best suited to your domain to get best results. If a model is not explicitly specified, then we auto-select a model based on the parameters in the RecognitionConfig.

Model	Description
latest_long	Best for long form content like media or conversation.
latest_short	Best for short form content like commands or single shot directed speech.
command_and_search	Best for short queries such as voice commands or voice search.
phone_call	Best for audio that originated from a phone call (typically recorded at an 8khz sampling rate).
video	Best for audio that originated from video or includes multiple speakers. Ideally the audio is recorded at a 16khz or greater sampling rate. This is a premium model that costs more than the standard rate.
default	Best for audio that is not one of the specific audio models. For example, long-form audio. Ideally the audio is high-fidelity, recorded at a 16khz or greater sampling rate.
medical_conversation	Best for audio that originated from a conversation between a medical provider and patient.
medical_dictation	Best for audio that originated from dictation notes by a medical provider.

.

Returns:

• (::String) —

  Which model to select for the given request. Select the model best suited to your domain to get best results. If a model is not explicitly specified, then we auto-select a model based on the parameters in the RecognitionConfig.

  Model	Description
  latest_long	Best for long form content like media or conversation.
  latest_short	Best for short form content like commands or single shot directed speech.
  command_and_search	Best for short queries such as voice commands or voice search.
  phone_call	Best for audio that originated from a phone call (typically recorded at an 8khz sampling rate).
  video	Best for audio that originated from video or includes multiple speakers. Ideally the audio is recorded at a 16khz or greater sampling rate. This is a premium model that costs more than the standard rate.
  default	Best for audio that is not one of the specific audio models. For example, long-form audio. Ideally the audio is high-fidelity, recorded at a 16khz or greater sampling rate.
  medical_conversation	Best for audio that originated from a conversation between a medical provider and patient.
  medical_dictation	Best for audio that originated from dictation notes by a medical provider.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#profanity_filter ⇒ ::Boolean

Returns If set to true, the server will attempt to filter out profanities, replacing all but the initial character in each filtered word with asterisks, e.g. "f***". If set to false or omitted, profanities won't be filtered out.

Returns:

• (::Boolean) —

  If set to true, the server will attempt to filter out profanities, replacing all but the initial character in each filtered word with asterisks, e.g. "f***". If set to false or omitted, profanities won't be filtered out.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#sample_rate_hertz ⇒ ::Integer

Returns Sample rate in Hertz of the audio data sent in all RecognitionAudio messages. Valid values are: 8000-48000. 16000 is optimal. For best results, set the sampling rate of the audio source to 16000 Hz. If that's not possible, use the native sample rate of the audio source (instead of re-sampling). This field is optional for FLAC and WAV audio files, but is required for all other audio formats. For details, see AudioEncoding.

Returns:

• (::Integer) —

  Sample rate in Hertz of the audio data sent in all RecognitionAudio messages. Valid values are: 8000-48000. 16000 is optimal. For best results, set the sampling rate of the audio source to 16000 Hz. If that's not possible, use the native sample rate of the audio source (instead of re-sampling). This field is optional for FLAC and WAV audio files, but is required for all other audio formats. For details, see AudioEncoding.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#speech_contexts ⇒ ::Array<::Google::Cloud::Speech::V1p1beta1::SpeechContext>

Returns Array of SpeechContext. A means to provide context to assist the speech recognition. For more information, see speech adaptation.

Returns:

• (::Array<::Google::Cloud::Speech::V1p1beta1::SpeechContext>) —

  Array of SpeechContext. A means to provide context to assist the speech recognition. For more information, see speech adaptation.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#transcript_normalization ⇒ ::Google::Cloud::Speech::V1p1beta1::TranscriptNormalization

Returns Use transcription normalization to automatically replace parts of the transcript with phrases of your choosing. For StreamingRecognize, this normalization only applies to stable partial transcripts (stability > 0.8) and final transcripts.

Returns:

• (::Google::Cloud::Speech::V1p1beta1::TranscriptNormalization) —

  Use transcription normalization to automatically replace parts of the transcript with phrases of your choosing. For StreamingRecognize, this normalization only applies to stable partial transcripts (stability > 0.8) and final transcripts.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end

#use_enhanced ⇒ ::Boolean

Returns Set to true to use an enhanced model for speech recognition. If use_enhanced is set to true and the model field is not set, then an appropriate enhanced model is chosen if an enhanced model exists for the audio.

If use_enhanced is true and an enhanced version of the specified model does not exist, then the speech is recognized using the standard version of the specified model.

Returns:

• (::Boolean) —

  Set to true to use an enhanced model for speech recognition. If use_enhanced is set to true and the model field is not set, then an appropriate enhanced model is chosen if an enhanced model exists for the audio.

  If use_enhanced is true and an enhanced version of the specified model does not exist, then the speech is recognized using the standard version of the specified model.

# File 'proto_docs/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 365

class RecognitionConfig
  include ::Google::Protobuf::MessageExts
  extend ::Google::Protobuf::MessageExts::ClassMethods

  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio, unless the
  # `audio_channel_count` and `enable_separate_recognition_per_channel` fields
  # are set.
  #
  # For best results, the audio source should be captured and transmitted using
  # a lossless encoding (`FLAC` or `LINEAR16`). The accuracy of the speech
  # recognition can be reduced if lossy codecs are used to capture or transmit
  # audio, particularly if background noise is present. Lossy codecs include
  # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, `SPEEX_WITH_HEADER_BYTE`, `MP3`,
  # and `WEBM_OPUS`.
  #
  # The `FLAC` and `WAV` audio file formats include a header that describes the
  # included audio content. You can request recognition for `WAV` files that
  # contain either `LINEAR16` or `MULAW` encoded audio.
  # If you send `FLAC` or `WAV` audio file format in
  # your request, you do not need to specify an `AudioEncoding`; the audio
  # encoding format is determined from the file header. If you specify
  # an `AudioEncoding` when you send  send `FLAC` or `WAV` audio, the
  # encoding configuration must match the encoding described in the audio
  # header; otherwise the request returns an
  # [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT] error
  # code.
  module AudioEncoding
    # Not specified.
    ENCODING_UNSPECIFIED = 0

    # Uncompressed 16-bit signed little-endian samples (Linear PCM).
    LINEAR16 = 1

    # `FLAC` (Free Lossless Audio
    # Codec) is the recommended encoding because it is
    # lossless--therefore recognition is not compromised--and
    # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream
    # encoding supports 16-bit and 24-bit samples, however, not all fields in
    # `STREAMINFO` are supported.
    FLAC = 2

    # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
    MULAW = 3

    # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
    AMR = 4

    # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
    AMR_WB = 5

    # Opus encoded audio frames in Ogg container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)).
    # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000.
    OGG_OPUS = 6

    # Although the use of lossy encodings is not recommended, if a very low
    # bitrate encoding is required, `OGG_OPUS` is highly preferred over
    # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
    # Cloud Speech API has a header byte in each block, as in MIME type
    # `audio/x-speex-with-header-byte`.
    # It is a variant of the RTP Speex encoding defined in
    # [RFC 5574](https://tools.ietf.org/html/rfc5574).
    # The stream is a sequence of blocks, one block per RTP packet. Each block
    # starts with a byte containing the length of the block, in bytes, followed
    # by one or more frames of Speex data, padded to an integral number of
    # bytes (octets) as specified in RFC 5574. In other words, each RTP header
    # is replaced with a single byte containing the block length. Only Speex
    # wideband is supported. `sample_rate_hertz` must be 16000.
    SPEEX_WITH_HEADER_BYTE = 7

    # MP3 audio. MP3 encoding is a Beta feature and only available in
    # v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
    # kbps). When using this encoding, `sample_rate_hertz` has to match the
    # sample rate of the file being used.
    MP3 = 8

    # Opus encoded audio frames in WebM container
    # ([OggOpus](https://wiki.xiph.org/OggOpus)). `sample_rate_hertz` must be
    # one of 8000, 12000, 16000, 24000, or 48000.
    WEBM_OPUS = 9
  end
end