# frozen_string_literal: true
# Copyright 2021 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 CX
module V3
# 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 on how to process the audio content.
# @!attribute [rw] audio_encoding
# @return [::Google::Cloud::Dialogflow::CX::V3::AudioEncoding]
# Required. 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] enable_word_info
# @return [::Boolean]
# Optional. If `true`, Dialogflow returns
# {::Google::Cloud::Dialogflow::CX::V3::SpeechWordInfo SpeechWordInfo} in
# {::Google::Cloud::Dialogflow::CX::V3::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
# @return [::Array<::String>]
# Optional. 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.
# @!attribute [rw] model
# @return [::String]
# Optional. Which Speech 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 InputAudioConfig.
# 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.
# @!attribute [rw] model_variant
# @return [::Google::Cloud::Dialogflow::CX::V3::SpeechModelVariant]
# Optional. Which variant of the [Speech
# model][google.cloud.dialogflow.cx.v3.InputAudioConfig.model] to use.
# @!attribute [rw] single_utterance
# @return [::Boolean]
# Optional. 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.
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::CX::V3::VoiceSelectionParams#ssml_gender ssml_gender}.
#
# For the list of available voices, please refer to [Supported voices and
# languages](https://cloud.google.com/text-to-speech/docs/voices).
# @!attribute [rw] ssml_gender
# @return [::Google::Cloud::Dialogflow::CX::V3::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::CX::V3::VoiceSelectionParams#name name}. Note that
# this is only a preference, not requirement. If a voice of the appropriate
# gender is not available, the synthesizer substitutes 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::CX::V3::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 how to generate the output audio content.
# @!attribute [rw] audio_encoding
# @return [::Google::Cloud::Dialogflow::CX::V3::OutputAudioEncoding]
# Required. Audio encoding of the synthesized audio content.
# @!attribute [rw] sample_rate_hertz
# @return [::Integer]
# Optional. 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::CX::V3::SynthesizeSpeechConfig]
# Optional. Configuration of how speech should be synthesized.
class OutputAudioConfig
include ::Google::Protobuf::MessageExts
extend ::Google::Protobuf::MessageExts::ClassMethods
end
# Settings related to speech generating.
# @!attribute [rw] synthesize_speech_configs
# @return [::Google::Protobuf::Map{::String => ::Google::Cloud::Dialogflow::CX::V3::SynthesizeSpeechConfig}]
# Configuration of how speech should be synthesized, mapping from
# language (https://dialogflow.com/docs/reference/language) to
# SynthesizeSpeechConfig.
class TextToSpeechSettings
include ::Google::Protobuf::MessageExts
extend ::Google::Protobuf::MessageExts::ClassMethods
# @!attribute [rw] key
# @return [::String]
# @!attribute [rw] value
# @return [::Google::Cloud::Dialogflow::CX::V3::SynthesizeSpeechConfig]
class SynthesizeSpeechConfigsEntry
include ::Google::Protobuf::MessageExts
extend ::Google::Protobuf::MessageExts::ClassMethods
end
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
end
# Variant of the specified [Speech
# model][google.cloud.dialogflow.cx.v3.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][InputAudioConfig.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::CX::V3::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
end
end
end
end
end
end