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

Inherits:

Object

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

show all

Defined in:: lib/google/cloud/speech/v1p1beta1/doc/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 collapse

#alternative_language_codes ⇒ Array<String>
Optional A list of up to 3 additional BCP-47 language tags, listing possible alternative languages of the supplied audio.
#audio_channel_count ⇒ Integer
Optional The number of channels in the input audio data.
#diarization_speaker_count ⇒ Integer
Optional If set, specifies the estimated number of speakers in the conversation.
#enable_automatic_punctuation ⇒ true, false
Optional If 'true', adds punctuation to recognition result hypotheses.
#enable_separate_recognition_per_channel ⇒ true, false
This needs to be set to ‘true’ explicitly and audio_channel_count > 1 to get each channel recognized separately.
#enable_speaker_diarization ⇒ true, false
Optional 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.
#enable_word_confidence ⇒ true, false
Optional If +true+, the top result includes a list of words and the confidence for those words.
#enable_word_time_offsets ⇒ true, false
Optional 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 The language of the supplied audio as a BCP-47 language tag.
#max_alternatives ⇒ Integer
Optional Maximum number of recognition hypotheses to be returned.
#metadata ⇒ Google::Cloud::Speech::V1p1beta1::RecognitionMetadata
Optional Metadata regarding this request.
#model ⇒ String
Optional Which model to select for the given request.
#profanity_filter ⇒ true, false
Optional 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>
Optional A means to provide context to assist the speech recognition.
#use_enhanced ⇒ true, false
Optional Set to true to use an enhanced model for speech recognition.

Instance Attribute Details

#alternative_language_codes ⇒ `Array<String>`

Returns Optional 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>) —
Optional 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 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

#audio_channel_count ⇒ `Integer`

Returns Optional The number of channels in the input audio data. ONLY set this for MULTI-CHANNEL recognition. Valid values for LINEAR16 and FLAC are +1+-+8+. Valid values for OGG_OPUS are '1'-'254'. 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) —
Optional The number of channels in the input audio data. ONLY set this for MULTI-CHANNEL recognition. Valid values for LINEAR16 and FLAC are +1+-+8+. Valid values for OGG_OPUS are '1'-'254'. 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 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

#diarization_speaker_count ⇒ `Integer`

Returns Optional If set, specifies the estimated number of speakers in the conversation. If not set, defaults to '2'. Ignored unless enable_speaker_diarization is set to true."

Returns:

(Integer) —
Optional If set, specifies the estimated number of speakers in the conversation. If not set, defaults to '2'. Ignored unless enable_speaker_diarization is set to true."

# File 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

#enable_automatic_punctuation ⇒ `true`, `false`

Returns Optional 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. NOTE: "This is currently offered as an experimental service, complimentary to all users. In the future this may be exclusively available as a premium feature."

Returns:

(true, false) —
Optional 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. NOTE: "This is currently offered as an experimental service, complimentary to all users. In the future this may be exclusively available as a premium feature."

# File 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

#enable_separate_recognition_per_channel ⇒ `true`, `false`

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. NOTE: The request is also billed cumulatively for all channels recognized: (audio_channel_count times the audio length)

Returns:

(true, false) —
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. NOTE: The request is also billed cumulatively for all channels recognized: (audio_channel_count times the audio length)

# File 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

#enable_speaker_diarization ⇒ `true`, `false`

Returns Optional 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: When this is true, we send all the words from the beginning of the audio for the top alternative in every consecutive responses. This is done in order to improve our speaker tags as our models learn to identify the speakers in the conversation over time.

Returns:

(true, false) —
Optional 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: When this is true, we send all the words from the beginning of the audio for the top alternative in every consecutive responses. This is done in order to improve our speaker tags as our models learn to identify the speakers in the conversation over time.

# File 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

#enable_word_confidence ⇒ `true`, `false`

Returns Optional 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:

(true, false) —
Optional 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 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

#enable_word_time_offsets ⇒ `true`, `false`

Returns Optional 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:

(true, false) —
Optional 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 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  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 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  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 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

#max_alternatives ⇒ `Integer`

Returns Optional 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) —
Optional 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 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

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

Returns Optional Metadata regarding this request.

Returns:

(Google::Cloud::Speech::V1p1beta1::RecognitionMetadata) —
Optional Metadata regarding this request.

# File 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

#model ⇒ `String`

Returns Optional 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
`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 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.

Returns:

(String) —

Optional 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
`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 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.

# File 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

#profanity_filter ⇒ `true`, `false`

Returns Optional 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:

(true, false) —
Optional 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 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  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 and 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 and required for all other audio formats. For details, see AudioEncoding.

# File 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

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

Returns Optional A means to provide context to assist the speech recognition.

Returns:

(Array<Google::Cloud::Speech::V1p1beta1::SpeechContext>) —
Optional A means to provide context to assist the speech recognition.

# File 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

#use_enhanced ⇒ `true`, `false`

Returns Optional Set to true to use an enhanced model for speech recognition. You must also set the +model+ field to a valid, enhanced model. If +use_enhanced+ is set to true and the +model+ field is not set, then +use_enhanced+ is ignored. 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.

Enhanced speech models require that you opt-in to the audio logging using instructions in the alpha documentation. If you set +use_enhanced+ to true and you have not enabled audio logging, then you will receive an error.

Returns:

(true, false) —
Optional Set to true to use an enhanced model for speech recognition. You must also set the +model+ field to a valid, enhanced model. If +use_enhanced+ is set to true and the +model+ field is not set, then +use_enhanced+ is ignored. 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.

Enhanced speech models require that you opt-in to the audio logging using instructions in the alpha documentation. If you set +use_enhanced+ to true and you have not enabled audio logging, then you will receive an error.

# File 'lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb', line 247

class RecognitionConfig
  # The encoding of the audio data sent in the request.
  #
  # All encodings support only 1 channel (mono) audio.
  #
  # 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+, and +SPEEX_WITH_HEADER_BYTE+.
  #
  # 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} 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
  end
end

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

Overview

Defined Under Namespace

Instance Attribute Summary collapse

Instance Attribute Details

#alternative_language_codes ⇒ Array<String>

#audio_channel_count ⇒ Integer

#diarization_speaker_count ⇒ Integer

#enable_automatic_punctuation ⇒ true, false

#enable_separate_recognition_per_channel ⇒ true, false

#enable_speaker_diarization ⇒ true, false

#enable_word_confidence ⇒ true, false

#enable_word_time_offsets ⇒ true, false

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

#language_code ⇒ String

#max_alternatives ⇒ Integer

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

#model ⇒ String

#profanity_filter ⇒ true, false

#sample_rate_hertz ⇒ Integer

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

#use_enhanced ⇒ true, false

#alternative_language_codes ⇒ `Array<String>`

#audio_channel_count ⇒ `Integer`

#diarization_speaker_count ⇒ `Integer`

#enable_automatic_punctuation ⇒ `true`, `false`

#enable_separate_recognition_per_channel ⇒ `true`, `false`

#enable_speaker_diarization ⇒ `true`, `false`

#enable_word_confidence ⇒ `true`, `false`

#enable_word_time_offsets ⇒ `true`, `false`

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

#language_code ⇒ `String`

#max_alternatives ⇒ `Integer`

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

#model ⇒ `String`

#profanity_filter ⇒ `true`, `false`

#sample_rate_hertz ⇒ `Integer`

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

#use_enhanced ⇒ `true`, `false`