# frozen_string_literal: true

# Copyright 2020 Google LLC
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
#     https://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.

# Auto-generated by gapic-generator-ruby. DO NOT EDIT!


module Google
  module Cloud
    module Dialogflow
      module V2
        # Hints for the speech recognizer to help with recognition in a specific
        # conversation state.
        # @!attribute [rw] phrases
        #   @return [::Array<::String>]
        #     Optional. A list of strings containing words and phrases that the speech
        #     recognizer should recognize with higher likelihood.
        #
        #     This list can be used to:
        #
        #     * improve accuracy for words and phrases you expect the user to say,
        #       e.g. typical commands for your Dialogflow agent
        #     * add additional words to the speech recognizer vocabulary
        #     * ...
        #
        #     See the [Cloud Speech
        #     documentation](https://cloud.google.com/speech-to-text/quotas) for usage
        #     limits.
        # @!attribute [rw] boost
        #   @return [::Float]
        #     Optional. Boost for this context compared to other contexts:
        #
        #     * If the boost is positive, Dialogflow will increase the probability that
        #       the phrases in this context are recognized over similar sounding phrases.
        #     * If the boost is unspecified or non-positive, Dialogflow will not apply
        #       any boost.
        #
        #     Dialogflow recommends that you use boosts in the range (0, 20] and that you
        #     find a value that fits your use case with binary search.
        class SpeechContext
          include ::Google::Protobuf::MessageExts
          extend ::Google::Protobuf::MessageExts::ClassMethods
        end

        # Information for a word recognized by the speech recognizer.
        # @!attribute [rw] word
        #   @return [::String]
        #     The word this info is for.
        # @!attribute [rw] start_offset
        #   @return [::Google::Protobuf::Duration]
        #     Time offset relative to the beginning of the audio that corresponds to the
        #     start of the spoken word. This is an experimental feature and the accuracy
        #     of the time offset can vary.
        # @!attribute [rw] end_offset
        #   @return [::Google::Protobuf::Duration]
        #     Time offset relative to the beginning of the audio that corresponds to the
        #     end of the spoken word. This is an experimental feature and the accuracy of
        #     the time offset can vary.
        # @!attribute [rw] confidence
        #   @return [::Float]
        #     The Speech confidence between 0.0 and 1.0 for this word. A higher number
        #     indicates an estimated greater likelihood that the recognized word is
        #     correct. The default of 0.0 is a sentinel value indicating that confidence
        #     was not set.
        #
        #     This field is not guaranteed to be fully stable over time for the same
        #     audio input. Users should also not rely on it to always be provided.
        class SpeechWordInfo
          include ::Google::Protobuf::MessageExts
          extend ::Google::Protobuf::MessageExts::ClassMethods
        end

        # Instructs the speech recognizer how to process the audio content.
        # @!attribute [rw] audio_encoding
        #   @return [::Google::Cloud::Dialogflow::V2::AudioEncoding]
        #     Required. Audio encoding of the audio content to process.
        # @!attribute [rw] sample_rate_hertz
        #   @return [::Integer]
        #     Required. Sample rate (in Hertz) of the audio content sent in the query.
        #     Refer to [Cloud Speech API
        #     documentation](https://cloud.google.com/speech-to-text/docs/basics) for
        #     more details.
        # @!attribute [rw] language_code
        #   @return [::String]
        #     Required. The language of the supplied audio. Dialogflow does not do
        #     translations. See [Language
        #     Support](https://cloud.google.com/dialogflow/docs/reference/language)
        #     for a list of the currently supported language codes. Note that queries in
        #     the same session do not necessarily need to specify the same language.
        # @!attribute [rw] enable_word_info
        #   @return [::Boolean]
        #     If `true`, Dialogflow returns
        #     {::Google::Cloud::Dialogflow::V2::SpeechWordInfo SpeechWordInfo} in
        #     {::Google::Cloud::Dialogflow::V2::StreamingRecognitionResult StreamingRecognitionResult}
        #     with information about the recognized speech words, e.g. start and end time
        #     offsets. If false or unspecified, Speech doesn't return any word-level
        #     information.
        # @!attribute [rw] phrase_hints
        #   @deprecated This field is deprecated and may be removed in the next major version update.
        #   @return [::Array<::String>]
        #     A list of strings containing words and phrases that the speech
        #     recognizer should recognize with higher likelihood.
        #
        #     See [the Cloud Speech
        #     documentation](https://cloud.google.com/speech-to-text/docs/basics#phrase-hints)
        #     for more details.
        #
        #     This field is deprecated. Please use [`speech_contexts`]() instead. If you
        #     specify both [`phrase_hints`]() and [`speech_contexts`](), Dialogflow will
        #     treat the [`phrase_hints`]() as a single additional [`SpeechContext`]().
        # @!attribute [rw] speech_contexts
        #   @return [::Array<::Google::Cloud::Dialogflow::V2::SpeechContext>]
        #     Context information to assist speech recognition.
        #
        #     See [the Cloud Speech
        #     documentation](https://cloud.google.com/speech-to-text/docs/basics#phrase-hints)
        #     for more details.
        # @!attribute [rw] model
        #   @return [::String]
        #     Optional. Which Speech model to select for the given request.
        #     For more information, see
        #     [Speech models](https://cloud.google.com/dialogflow/es/docs/speech-models).
        # @!attribute [rw] model_variant
        #   @return [::Google::Cloud::Dialogflow::V2::SpeechModelVariant]
        #     Which variant of the [Speech
        #     model][google.cloud.dialogflow.v2.InputAudioConfig.model] to use.
        # @!attribute [rw] single_utterance
        #   @return [::Boolean]
        #     If `false` (default), recognition does not cease until the
        #     client closes the stream.
        #     If `true`, the recognizer will detect a single spoken utterance in input
        #     audio. Recognition ceases when it detects the audio's voice has
        #     stopped or paused. In this case, once a detected intent is received, the
        #     client should close the stream and start a new request with a new stream as
        #     needed.
        #     Note: This setting is relevant only for streaming methods.
        #     Note: When specified, InputAudioConfig.single_utterance takes precedence
        #     over StreamingDetectIntentRequest.single_utterance.
        # @!attribute [rw] disable_no_speech_recognized_event
        #   @return [::Boolean]
        #     Only used in
        #     {::Google::Cloud::Dialogflow::V2::Participants::Client#analyze_content Participants.AnalyzeContent}
        #     and
        #     {::Google::Cloud::Dialogflow::V2::Participants::Client#streaming_analyze_content Participants.StreamingAnalyzeContent}.
        #     If `false` and recognition doesn't return any result, trigger
        #     `NO_SPEECH_RECOGNIZED` event to Dialogflow agent.
        # @!attribute [rw] enable_automatic_punctuation
        #   @return [::Boolean]
        #     Enable automatic punctuation option at the speech backend.
        # @!attribute [rw] phrase_sets
        #   @return [::Array<::String>]
        #     A collection of phrase set resources to use for speech adaptation.
        # @!attribute [rw] opt_out_conformer_model_migration
        #   @return [::Boolean]
        #     If `true`, the request will opt out for STT conformer model migration.
        #     This field will be deprecated once force migration takes place in June
        #     2024. Please refer to [Dialogflow ES Speech model
        #     migration](https://cloud.google.com/dialogflow/es/docs/speech-model-migration).
        class InputAudioConfig
          include ::Google::Protobuf::MessageExts
          extend ::Google::Protobuf::MessageExts::ClassMethods
        end

        # Description of which voice to use for speech synthesis.
        # @!attribute [rw] name
        #   @return [::String]
        #     Optional. The name of the voice. If not set, the service will choose a
        #     voice based on the other parameters such as language_code and
        #     {::Google::Cloud::Dialogflow::V2::VoiceSelectionParams#ssml_gender ssml_gender}.
        # @!attribute [rw] ssml_gender
        #   @return [::Google::Cloud::Dialogflow::V2::SsmlVoiceGender]
        #     Optional. The preferred gender of the voice. If not set, the service will
        #     choose a voice based on the other parameters such as language_code and
        #     {::Google::Cloud::Dialogflow::V2::VoiceSelectionParams#name name}. Note that
        #     this is only a preference, not requirement. If a voice of the appropriate
        #     gender is not available, the synthesizer should substitute a voice with a
        #     different gender rather than failing the request.
        class VoiceSelectionParams
          include ::Google::Protobuf::MessageExts
          extend ::Google::Protobuf::MessageExts::ClassMethods
        end

        # Configuration of how speech should be synthesized.
        # @!attribute [rw] speaking_rate
        #   @return [::Float]
        #     Optional. Speaking rate/speed, in the range [0.25, 4.0]. 1.0 is the normal
        #     native speed supported by the specific voice. 2.0 is twice as fast, and 0.5
        #     is half as fast. If unset(0.0), defaults to the native 1.0 speed. Any other
        #     values < 0.25 or > 4.0 will return an error.
        # @!attribute [rw] pitch
        #   @return [::Float]
        #     Optional. Speaking pitch, in the range [-20.0, 20.0]. 20 means increase 20
        #     semitones from the original pitch. -20 means decrease 20 semitones from the
        #     original pitch.
        # @!attribute [rw] volume_gain_db
        #   @return [::Float]
        #     Optional. Volume gain (in dB) of the normal native volume supported by the
        #     specific voice, in the range [-96.0, 16.0]. If unset, or set to a value of
        #     0.0 (dB), will play at normal native signal amplitude. A value of -6.0 (dB)
        #     will play at approximately half the amplitude of the normal native signal
        #     amplitude. A value of +6.0 (dB) will play at approximately twice the
        #     amplitude of the normal native signal amplitude. We strongly recommend not
        #     to exceed +10 (dB) as there's usually no effective increase in loudness for
        #     any value greater than that.
        # @!attribute [rw] effects_profile_id
        #   @return [::Array<::String>]
        #     Optional. An identifier which selects 'audio effects' profiles that are
        #     applied on (post synthesized) text to speech. Effects are applied on top of
        #     each other in the order they are given.
        # @!attribute [rw] voice
        #   @return [::Google::Cloud::Dialogflow::V2::VoiceSelectionParams]
        #     Optional. The desired voice of the synthesized audio.
        class SynthesizeSpeechConfig
          include ::Google::Protobuf::MessageExts
          extend ::Google::Protobuf::MessageExts::ClassMethods
        end

        # Instructs the speech synthesizer on how to generate the output audio content.
        # If this audio config is supplied in a request, it overrides all existing
        # text-to-speech settings applied to the agent.
        # @!attribute [rw] audio_encoding
        #   @return [::Google::Cloud::Dialogflow::V2::OutputAudioEncoding]
        #     Required. Audio encoding of the synthesized audio content.
        # @!attribute [rw] sample_rate_hertz
        #   @return [::Integer]
        #     The synthesis sample rate (in hertz) for this audio. If not
        #     provided, then the synthesizer will use the default sample rate based on
        #     the audio encoding. If this is different from the voice's natural sample
        #     rate, then the synthesizer will honor this request by converting to the
        #     desired sample rate (which might result in worse audio quality).
        # @!attribute [rw] synthesize_speech_config
        #   @return [::Google::Cloud::Dialogflow::V2::SynthesizeSpeechConfig]
        #     Configuration of how speech should be synthesized.
        class OutputAudioConfig
          include ::Google::Protobuf::MessageExts
          extend ::Google::Protobuf::MessageExts::ClassMethods
        end

        # A wrapper of repeated TelephonyDtmf digits.
        # @!attribute [rw] dtmf_events
        #   @return [::Array<::Google::Cloud::Dialogflow::V2::TelephonyDtmf>]
        #     A sequence of TelephonyDtmf digits.
        class TelephonyDtmfEvents
          include ::Google::Protobuf::MessageExts
          extend ::Google::Protobuf::MessageExts::ClassMethods
        end

        # Configures speech transcription for
        # {::Google::Cloud::Dialogflow::V2::ConversationProfile ConversationProfile}.
        # @!attribute [rw] speech_model_variant
        #   @return [::Google::Cloud::Dialogflow::V2::SpeechModelVariant]
        #     The speech model used in speech to text.
        #     `SPEECH_MODEL_VARIANT_UNSPECIFIED`, `USE_BEST_AVAILABLE` will be treated as
        #     `USE_ENHANCED`. It can be overridden in
        #     {::Google::Cloud::Dialogflow::V2::AnalyzeContentRequest AnalyzeContentRequest}
        #     and
        #     {::Google::Cloud::Dialogflow::V2::StreamingAnalyzeContentRequest StreamingAnalyzeContentRequest}
        #     request. If enhanced model variant is specified and an enhanced version of
        #     the specified model for the language does not exist, then it would emit an
        #     error.
        # @!attribute [rw] model
        #   @return [::String]
        #     Which Speech model to select. Select the
        #     model best suited to your domain to get best results. If a model is not
        #     explicitly specified, then Dialogflow auto-selects a model based on other
        #     parameters in the SpeechToTextConfig and Agent settings.
        #     If enhanced speech model is enabled for the agent and an enhanced
        #     version of the specified model for the language does not exist, then the
        #     speech is recognized using the standard version of the specified model.
        #     Refer to
        #     [Cloud Speech API
        #     documentation](https://cloud.google.com/speech-to-text/docs/basics#select-model)
        #     for more details.
        #     If you specify a model, the following models typically have the best
        #     performance:
        #
        #     - phone_call (best for Agent Assist and telephony)
        #     - latest_short (best for Dialogflow non-telephony)
        #     - command_and_search
        #
        #     Leave this field unspecified to use
        #     [Agent Speech
        #     settings](https://cloud.google.com/dialogflow/cx/docs/concept/agent#settings-speech)
        #     for model selection.
        # @!attribute [rw] phrase_sets
        #   @return [::Array<::String>]
        #     List of names of Cloud Speech phrase sets that are used for transcription.
        #     For phrase set limitations, please refer to [Cloud Speech API quotas and
        #     limits](https://cloud.google.com/speech-to-text/quotas#content).
        # @!attribute [rw] audio_encoding
        #   @return [::Google::Cloud::Dialogflow::V2::AudioEncoding]
        #     Audio encoding of the audio content to process.
        # @!attribute [rw] sample_rate_hertz
        #   @return [::Integer]
        #     Sample rate (in Hertz) of the audio content sent in the query.
        #     Refer to [Cloud Speech API
        #     documentation](https://cloud.google.com/speech-to-text/docs/basics) for
        #     more details.
        # @!attribute [rw] language_code
        #   @return [::String]
        #     The language of the supplied audio. Dialogflow does not do
        #     translations. See [Language
        #     Support](https://cloud.google.com/dialogflow/docs/reference/language)
        #     for a list of the currently supported language codes. Note that queries in
        #     the same session do not necessarily need to specify the same language.
        # @!attribute [rw] enable_word_info
        #   @return [::Boolean]
        #     If `true`, Dialogflow returns
        #     {::Google::Cloud::Dialogflow::V2::SpeechWordInfo SpeechWordInfo} in
        #     {::Google::Cloud::Dialogflow::V2::StreamingRecognitionResult StreamingRecognitionResult}
        #     with information about the recognized speech words, e.g. start and end time
        #     offsets. If false or unspecified, Speech doesn't return any word-level
        #     information.
        # @!attribute [rw] use_timeout_based_endpointing
        #   @return [::Boolean]
        #     Use timeout based endpointing, interpreting endpointer sensitivity as
        #     seconds of timeout value.
        class SpeechToTextConfig
          include ::Google::Protobuf::MessageExts
          extend ::Google::Protobuf::MessageExts::ClassMethods
        end

        # [DTMF](https://en.wikipedia.org/wiki/Dual-tone_multi-frequency_signaling)
        # digit in Telephony Gateway.
        module TelephonyDtmf
          # Not specified. This value may be used to indicate an absent digit.
          TELEPHONY_DTMF_UNSPECIFIED = 0

          # Number: '1'.
          DTMF_ONE = 1

          # Number: '2'.
          DTMF_TWO = 2

          # Number: '3'.
          DTMF_THREE = 3

          # Number: '4'.
          DTMF_FOUR = 4

          # Number: '5'.
          DTMF_FIVE = 5

          # Number: '6'.
          DTMF_SIX = 6

          # Number: '7'.
          DTMF_SEVEN = 7

          # Number: '8'.
          DTMF_EIGHT = 8

          # Number: '9'.
          DTMF_NINE = 9

          # Number: '0'.
          DTMF_ZERO = 10

          # Letter: 'A'.
          DTMF_A = 11

          # Letter: 'B'.
          DTMF_B = 12

          # Letter: 'C'.
          DTMF_C = 13

          # Letter: 'D'.
          DTMF_D = 14

          # Asterisk/star: '*'.
          DTMF_STAR = 15

          # Pound/diamond/hash/square/gate/octothorpe: '#'.
          DTMF_POUND = 16
        end

        # Audio encoding of the audio content sent in the conversational query request.
        # Refer to the
        # [Cloud Speech API
        # documentation](https://cloud.google.com/speech-to-text/docs/basics) for more
        # details.
        module AudioEncoding
          # Not specified.
          AUDIO_ENCODING_UNSPECIFIED = 0

          # Uncompressed 16-bit signed little-endian samples (Linear PCM).
          AUDIO_ENCODING_LINEAR_16 = 1

          # [`FLAC`](https://xiph.org/flac/documentation.html) (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.
          AUDIO_ENCODING_FLAC = 2

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

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

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

          # Opus encoded audio frames in Ogg container
          # ([OggOpus](https://wiki.xiph.org/OggOpus)).
          # `sample_rate_hertz` must be 16000.
          AUDIO_ENCODING_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
          # Dialogflow 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.
          AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE = 7

          # 8-bit samples that compand 13-bit audio samples using G.711 PCMU/a-law.
          AUDIO_ENCODING_ALAW = 8
        end

        # Variant of the specified [Speech
        # model][google.cloud.dialogflow.v2.InputAudioConfig.model] to use.
        #
        # See the [Cloud Speech
        # documentation](https://cloud.google.com/speech-to-text/docs/enhanced-models)
        # for which models have different variants. For example, the "phone_call" model
        # has both a standard and an enhanced variant. When you use an enhanced model,
        # you will generally receive higher quality results than for a standard model.
        module SpeechModelVariant
          # No model variant specified. In this case Dialogflow defaults to
          # USE_BEST_AVAILABLE.
          SPEECH_MODEL_VARIANT_UNSPECIFIED = 0

          # Use the best available variant of the [Speech model][model] that the caller
          # is eligible for.
          #
          # Please see the [Dialogflow
          # docs](https://cloud.google.com/dialogflow/docs/data-logging) for
          # how to make your project eligible for enhanced models.
          USE_BEST_AVAILABLE = 1

          # Use standard model variant even if an enhanced model is available.  See the
          # [Cloud Speech
          # documentation](https://cloud.google.com/speech-to-text/docs/enhanced-models)
          # for details about enhanced models.
          USE_STANDARD = 2

          # Use an enhanced model variant:
          #
          # * If an enhanced variant does not exist for the given
          #   {::Google::Cloud::Dialogflow::V2::InputAudioConfig#model model} and request
          #   language, Dialogflow falls back to the standard variant.
          #
          #   The [Cloud Speech
          #   documentation](https://cloud.google.com/speech-to-text/docs/enhanced-models)
          #   describes which models have enhanced variants.
          #
          # * If the API caller isn't eligible for enhanced models, Dialogflow returns
          #   an error. Please see the [Dialogflow
          #   docs](https://cloud.google.com/dialogflow/docs/data-logging)
          #   for how to make your project eligible.
          USE_ENHANCED = 3
        end

        # Gender of the voice as described in
        # [SSML voice element](https://www.w3.org/TR/speech-synthesis11/#edef_voice).
        module SsmlVoiceGender
          # An unspecified gender, which means that the client doesn't care which
          # gender the selected voice will have.
          SSML_VOICE_GENDER_UNSPECIFIED = 0

          # A male voice.
          SSML_VOICE_GENDER_MALE = 1

          # A female voice.
          SSML_VOICE_GENDER_FEMALE = 2

          # A gender-neutral voice.
          SSML_VOICE_GENDER_NEUTRAL = 3
        end

        # Audio encoding of the output audio format in Text-To-Speech.
        module OutputAudioEncoding
          # Not specified.
          OUTPUT_AUDIO_ENCODING_UNSPECIFIED = 0

          # Uncompressed 16-bit signed little-endian samples (Linear PCM).
          # Audio content returned as LINEAR16 also contains a WAV header.
          OUTPUT_AUDIO_ENCODING_LINEAR_16 = 1

          # MP3 audio at 32kbps.
          OUTPUT_AUDIO_ENCODING_MP3 = 2

          # MP3 audio at 64kbps.
          OUTPUT_AUDIO_ENCODING_MP3_64_KBPS = 4

          # Opus encoded audio wrapped in an ogg container. The result will be a
          # file which can be played natively on Android, and in browsers (at least
          # Chrome and Firefox). The quality of the encoding is considerably higher
          # than MP3 while using approximately the same bitrate.
          OUTPUT_AUDIO_ENCODING_OGG_OPUS = 3

          # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
          OUTPUT_AUDIO_ENCODING_MULAW = 5

          # 8-bit samples that compand 13-bit audio samples using G.711 PCMU/a-law.
          OUTPUT_AUDIO_ENCODING_ALAW = 6
        end
      end
    end
  end
end
