# Copyright 2019 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.
module Google
module Cloud
module Speech
module V1p1beta1
# The top-level message sent by the client for the `Recognize` method.
# @!attribute [rw] config
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionConfig]
# *Required* Provides information to the recognizer that specifies how to
# process the request.
# @!attribute [rw] audio
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionAudio]
# *Required* The audio data to be recognized.
class RecognizeRequest; end
# The top-level message sent by the client for the `LongRunningRecognize`
# method.
# @!attribute [rw] config
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionConfig]
# *Required* Provides information to the recognizer that specifies how to
# process the request.
# @!attribute [rw] audio
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionAudio]
# *Required* The audio data to be recognized.
class LongRunningRecognizeRequest; end
# The top-level message sent by the client for the `StreamingRecognize` method.
# Multiple `StreamingRecognizeRequest` messages are sent. The first message
# must contain a `streaming_config` message and must not contain `audio` data.
# All subsequent messages must contain `audio` data and must not contain a
# `streaming_config` message.
# @!attribute [rw] streaming_config
# @return [Google::Cloud::Speech::V1p1beta1::StreamingRecognitionConfig]
# Provides information to the recognizer that specifies how to process the
# request. The first `StreamingRecognizeRequest` message must contain a
# `streaming_config` message.
# @!attribute [rw] audio_content
# @return [String]
# The audio data to be recognized. Sequential chunks of audio data are sent
# in sequential `StreamingRecognizeRequest` messages. The first
# `StreamingRecognizeRequest` message must not contain `audio_content` data
# and all subsequent `StreamingRecognizeRequest` messages must contain
# `audio_content` data. The audio bytes must be encoded as specified in
# `RecognitionConfig`. Note: as with all bytes fields, protobuffers use a
# pure binary representation (not base64). See
# [content limits](https://cloud.google.com/speech-to-text/quotas#content).
class StreamingRecognizeRequest; end
# Provides information to the recognizer that specifies how to process the
# request.
# @!attribute [rw] config
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionConfig]
# *Required* Provides information to the recognizer that specifies how to
# process the request.
# @!attribute [rw] single_utterance
# @return [true, false]
# *Optional* If `false` or omitted, the recognizer will perform continuous
# recognition (continuing to wait for and process audio even if the user
# pauses speaking) until the client closes the input stream (gRPC API) or
# until the maximum time limit has been reached. May return multiple
# `StreamingRecognitionResult`s with the `is_final` flag set to `true`.
#
# If `true`, the recognizer will detect a single spoken utterance. When it
# detects that the user has paused or stopped speaking, it will return an
# `END_OF_SINGLE_UTTERANCE` event and cease recognition. It will return no
# more than one `StreamingRecognitionResult` with the `is_final` flag set to
# `true`.
# @!attribute [rw] interim_results
# @return [true, false]
# *Optional* If `true`, interim results (tentative hypotheses) may be
# returned as they become available (these interim results are indicated with
# the `is_final=false` flag).
# If `false` or omitted, only `is_final=true` result(s) are returned.
class StreamingRecognitionConfig; end
# Provides information to the recognizer that specifies how to process the
# request.
# @!attribute [rw] encoding
# @return [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
# {Google::Cloud::Speech::V1p1beta1::RecognitionConfig::AudioEncoding AudioEncoding}.
# @!attribute [rw] sample_rate_hertz
# @return [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
# {Google::Cloud::Speech::V1p1beta1::RecognitionConfig::AudioEncoding AudioEncoding}.
# @!attribute [rw] audio_channel_count
# @return [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'.
# @!attribute [rw] enable_separate_recognition_per_channel
# @return [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. The
# request is billed cumulatively for all channels recognized:
# `audio_channel_count` multiplied by the length of the audio.
# @!attribute [rw] language_code
# @return [String]
# *Required* The language of the supplied audio as a
# [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag.
# Example: "en-US".
# See [Language Support](https://cloud.google.com/speech-to-text/docs/languages)
# for a list of the currently supported language codes.
# @!attribute [rw] alternative_language_codes
# @return [Array<String>]
# *Optional* A list of up to 3 additional
# [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tags,
# listing possible alternative languages of the supplied audio.
# See [Language Support](https://cloud.google.com/speech-to-text/docs/languages)
# 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).
# @!attribute [rw] max_alternatives
# @return [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.
# @!attribute [rw] profanity_filter
# @return [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.
# @!attribute [rw] speech_contexts
# @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechContext>]
# *Optional* array of
# {Google::Cloud::Speech::V1p1beta1::SpeechContext SpeechContext}. A means to
# provide context to assist the speech recognition. For more information, see
# [Phrase Hints](https://cloud.google.com/speech-to-text/docs/basics#phrase-hints).
# @!attribute [rw] enable_word_time_offsets
# @return [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`.
# @!attribute [rw] enable_word_confidence
# @return [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`.
# @!attribute [rw] enable_automatic_punctuation
# @return [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.
# @!attribute [rw] enable_speaker_diarization
# @return [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: Use diarization_config instead.
# @!attribute [rw] diarization_speaker_count
# @return [Integer]
# *Optional*
# If set, specifies the estimated number of speakers in the conversation.
# Defaults to '2'. Ignored unless enable_speaker_diarization is set to true.
# Note: Use diarization_config instead.
# @!attribute [rw] diarization_config
# @return [Google::Cloud::Speech::V1p1beta1::SpeakerDiarizationConfig]
# *Optional* Config to enable speaker diarization and set additional
# parameters to make diarization better suited for your application.
# Note: When this is enabled, we send all the words from the beginning of the
# audio for the top alternative in every consecutive STREAMING responses.
# This is done in order to improve our speaker tags as our models learn to
# identify the speakers in the conversation over time.
# For non-streaming requests, the diarization results will be provided only
# in the top alternative of the FINAL SpeechRecognitionResult.
# @!attribute [rw] metadata
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata]
# *Optional* Metadata regarding this request.
# @!attribute [rw] model
# @return [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.
# <table>
# <tr>
# <td><b>Model</b></td>
# <td><b>Description</b></td>
# </tr>
# <tr>
# <td><code>command_and_search</code></td>
# <td>Best for short queries such as voice commands or voice search.</td>
# </tr>
# <tr>
# <td><code>phone_call</code></td>
# <td>Best for audio that originated from a phone call (typically
# recorded at an 8khz sampling rate).</td>
# </tr>
# <tr>
# <td><code>video</code></td>
# <td>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.</td>
# </tr>
# <tr>
# <td><code>default</code></td>
# <td>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.</td>
# </tr>
# </table>
# @!attribute [rw] use_enhanced
# @return [true, false]
# *Optional* Set to true to use an enhanced model for speech recognition.
# If `use_enhanced` is set to true and the `model` field is not set, then
# an appropriate enhanced model is chosen if an enhanced model exists for
# the audio.
#
# If `use_enhanced` is true and an enhanced version of the specified model
# does not exist, then the speech is recognized using the standard version
# of the specified model.
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
# MP3 audio. Support all standard MP3 bitrates (which range from 32-320
# kbps). When using this encoding, `sample_rate_hertz` can be optionally
# unset if not known.
MP3 = 8
end
end
# *Optional* Config to enable speaker diarization.
# @!attribute [rw] enable_speaker_diarization
# @return [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.
# @!attribute [rw] min_speaker_count
# @return [Integer]
# *Optional*
# Minimum number of speakers in the conversation. This range gives you more
# flexibility by allowing the system to automatically determine the correct
# number of speakers. If not set, the default value is 2.
# @!attribute [rw] max_speaker_count
# @return [Integer]
# *Optional*
# Maximum number of speakers in the conversation. This range gives you more
# flexibility by allowing the system to automatically determine the correct
# number of speakers. If not set, the default value is 6.
class SpeakerDiarizationConfig; end
# Description of audio data to be recognized.
# @!attribute [rw] interaction_type
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata::InteractionType]
# The use case most closely describing the audio content to be recognized.
# @!attribute [rw] industry_naics_code_of_audio
# @return [Integer]
# The industry vertical to which this speech recognition request most
# closely applies. This is most indicative of the topics contained
# in the audio. Use the 6-digit NAICS code to identify the industry
# vertical - see https://www.naics.com/search/.
# @!attribute [rw] microphone_distance
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata::MicrophoneDistance]
# The audio type that most closely describes the audio being recognized.
# @!attribute [rw] original_media_type
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata::OriginalMediaType]
# The original media the speech was recorded on.
# @!attribute [rw] recording_device_type
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata::RecordingDeviceType]
# The type of device the speech was recorded with.
# @!attribute [rw] recording_device_name
# @return [String]
# The device used to make the recording. Examples 'Nexus 5X' or
# 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
# 'Cardioid Microphone'.
# @!attribute [rw] original_mime_type
# @return [String]
# Mime type of the original audio file. For example `audio/m4a`,
# `audio/x-alaw-basic`, `audio/mp3`, `audio/3gpp`.
# A list of possible audio mime types is maintained at
# http://www.iana.org/assignments/media-types/media-types.xhtml#audio
# @!attribute [rw] obfuscated_id
# @return [Integer]
# Obfuscated (privacy-protected) ID of the user, to identify number of
# unique users using the service.
# @!attribute [rw] audio_topic
# @return [String]
# Description of the content. Eg. "Recordings of federal supreme court
# hearings from 2012".
class RecognitionMetadata
# Use case categories that the audio recognition request can be described
# by.
module InteractionType
# Use case is either unknown or is something other than one of the other
# values below.
INTERACTION_TYPE_UNSPECIFIED = 0
# Multiple people in a conversation or discussion. For example in a
# meeting with two or more people actively participating. Typically
# all the primary people speaking would be in the same room (if not,
# see PHONE_CALL)
DISCUSSION = 1
# One or more persons lecturing or presenting to others, mostly
# uninterrupted.
PRESENTATION = 2
# A phone-call or video-conference in which two or more people, who are
# not in the same room, are actively participating.
PHONE_CALL = 3
# A recorded message intended for another person to listen to.
VOICEMAIL = 4
# Professionally produced audio (eg. TV Show, Podcast).
PROFESSIONALLY_PRODUCED = 5
# Transcribe spoken questions and queries into text.
VOICE_SEARCH = 6
# Transcribe voice commands, such as for controlling a device.
VOICE_COMMAND = 7
# Transcribe speech to text to create a written document, such as a
# text-message, email or report.
DICTATION = 8
end
# Enumerates the types of capture settings describing an audio file.
module MicrophoneDistance
# Audio type is not known.
MICROPHONE_DISTANCE_UNSPECIFIED = 0
# The audio was captured from a closely placed microphone. Eg. phone,
# dictaphone, or handheld microphone. Generally if there speaker is within
# 1 meter of the microphone.
NEARFIELD = 1
# The speaker if within 3 meters of the microphone.
MIDFIELD = 2
# The speaker is more than 3 meters away from the microphone.
FARFIELD = 3
end
# The original media the speech was recorded on.
module OriginalMediaType
# Unknown original media type.
ORIGINAL_MEDIA_TYPE_UNSPECIFIED = 0
# The speech data is an audio recording.
AUDIO = 1
# The speech data originally recorded on a video.
VIDEO = 2
end
# The type of device the speech was recorded with.
module RecordingDeviceType
# The recording device is unknown.
RECORDING_DEVICE_TYPE_UNSPECIFIED = 0
# Speech was recorded on a smartphone.
SMARTPHONE = 1
# Speech was recorded using a personal computer or tablet.
PC = 2
# Speech was recorded over a phone line.
PHONE_LINE = 3
# Speech was recorded in a vehicle.
VEHICLE = 4
# Speech was recorded outdoors.
OTHER_OUTDOOR_DEVICE = 5
# Speech was recorded indoors.
OTHER_INDOOR_DEVICE = 6
end
end
# Provides "hints" to the speech recognizer to favor specific words and phrases
# in the results.
# @!attribute [rw] phrases
# @return [Array<String>]
# *Optional* A list of strings containing words and phrases "hints" so that
# the speech recognition is more likely to recognize them. This can be used
# to improve the accuracy for specific words and phrases, for example, if
# specific commands are typically spoken by the user. This can also be used
# to add additional words to the vocabulary of the recognizer. See
# [usage limits](https://cloud.google.com/speech-to-text/quotas#content).
#
# List items can also be set to classes for groups of words that represent
# common concepts that occur in natural language. For example, rather than
# providing phrase hints for every month of the year, using the $MONTH class
# improves the likelihood of correctly transcribing audio that includes
# months.
# @!attribute [rw] boost
# @return [Float]
# Hint Boost. Positive value will increase the probability that a specific
# phrase will be recognized over other similar sounding phrases. The higher
# the boost, the higher the chance of false positive recognition as well.
# Negative boost values would correspond to anti-biasing. Anti-biasing is not
# enabled, so negative boost will simply be ignored. Though `boost` can
# accept a wide range of positive values, most use cases are best served with
# values between 0 and 20. We recommend using a binary search approach to
# finding the optimal value for your use case.
class SpeechContext; end
# Contains audio data in the encoding specified in the `RecognitionConfig`.
# Either `content` or `uri` must be supplied. Supplying both or neither
# returns {Google::Rpc::Code::INVALID_ARGUMENT}.
# See [content limits](https://cloud.google.com/speech-to-text/quotas#content).
# @!attribute [rw] content
# @return [String]
# The audio data bytes encoded as specified in
# `RecognitionConfig`. Note: as with all bytes fields, protobuffers use a
# pure binary representation, whereas JSON representations use base64.
# @!attribute [rw] uri
# @return [String]
# URI that points to a file that contains audio data bytes as specified in
# `RecognitionConfig`. The file must not be compressed (for example, gzip).
# Currently, only Google Cloud Storage URIs are
# supported, which must be specified in the following format:
# `gs://bucket_name/object_name` (other URI formats return
# {Google::Rpc::Code::INVALID_ARGUMENT}).
# For more information, see [Request
# URIs](https://cloud.google.com/storage/docs/reference-uris).
class RecognitionAudio; end
# The only message returned to the client by the `Recognize` method. It
# contains the result as zero or more sequential `SpeechRecognitionResult`
# messages.
# @!attribute [rw] results
# @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechRecognitionResult>]
# Output only. Sequential list of transcription results corresponding to
# sequential portions of audio.
class RecognizeResponse; end
# The only message returned to the client by the `LongRunningRecognize` method.
# It contains the result as zero or more sequential `SpeechRecognitionResult`
# messages. It is included in the `result.response` field of the `Operation`
# returned by the `GetOperation` call of the `google::longrunning::Operations`
# service.
# @!attribute [rw] results
# @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechRecognitionResult>]
# Output only. Sequential list of transcription results corresponding to
# sequential portions of audio.
class LongRunningRecognizeResponse; end
# Describes the progress of a long-running `LongRunningRecognize` call. It is
# included in the `metadata` field of the `Operation` returned by the
# `GetOperation` call of the `google::longrunning::Operations` service.
# @!attribute [rw] progress_percent
# @return [Integer]
# Approximate percentage of audio processed thus far. Guaranteed to be 100
# when the audio is fully processed and the results are available.
# @!attribute [rw] start_time
# @return [Google::Protobuf::Timestamp]
# Time when the request was received.
# @!attribute [rw] last_update_time
# @return [Google::Protobuf::Timestamp]
# Time of the most recent processing update.
class LongRunningRecognizeMetadata; end
# `StreamingRecognizeResponse` is the only message returned to the client by
# `StreamingRecognize`. A series of zero or more `StreamingRecognizeResponse`
# messages are streamed back to the client. If there is no recognizable
# audio, and `single_utterance` is set to false, then no messages are streamed
# back to the client.
#
# Here's an example of a series of ten `StreamingRecognizeResponse`s that might
# be returned while processing audio:
#
# 1. results { alternatives { transcript: "tube" } stability: 0.01 }
#
# 2. results { alternatives { transcript: "to be a" } stability: 0.01 }
#
# 3. results { alternatives { transcript: "to be" } stability: 0.9 }
# results { alternatives { transcript: " or not to be" } stability: 0.01 }
#
# 4. results { alternatives { transcript: "to be or not to be"
# confidence: 0.92 }
# alternatives { transcript: "to bee or not to bee" }
# is_final: true }
#
# 5. results { alternatives { transcript: " that's" } stability: 0.01 }
#
# 6. results { alternatives { transcript: " that is" } stability: 0.9 }
# results { alternatives { transcript: " the question" } stability: 0.01 }
#
# 7. results { alternatives { transcript: " that is the question"
# confidence: 0.98 }
# alternatives { transcript: " that was the question" }
# is_final: true }
#
# Notes:
#
# * Only two of the above responses #4 and #7 contain final results; they are
# indicated by `is_final: true`. Concatenating these together generates the
# full transcript: "to be or not to be that is the question".
#
# * The others contain interim `results`. #3 and #6 contain two interim
# `results`: the first portion has a high stability and is less likely to
# change; the second portion has a low stability and is very likely to
# change. A UI designer might choose to show only high stability `results`.
#
# * The specific `stability` and `confidence` values shown above are only for
# illustrative purposes. Actual values may vary.
#
# * In each response, only one of these fields will be set:
# `error`,
# `speech_event_type`, or
# one or more (repeated) `results`.
# @!attribute [rw] error
# @return [Google::Rpc::Status]
# Output only. If set, returns a {Google::Rpc::Status}
# message that specifies the error for the operation.
# @!attribute [rw] results
# @return [Array<Google::Cloud::Speech::V1p1beta1::StreamingRecognitionResult>]
# Output only. This repeated list contains zero or more results that
# correspond to consecutive portions of the audio currently being processed.
# It contains zero or one `is_final=true` result (the newly settled portion),
# followed by zero or more `is_final=false` results (the interim results).
# @!attribute [rw] speech_event_type
# @return [Google::Cloud::Speech::V1p1beta1::StreamingRecognizeResponse::SpeechEventType]
# Output only. Indicates the type of speech event.
class StreamingRecognizeResponse
# Indicates the type of speech event.
module SpeechEventType
# No speech event specified.
SPEECH_EVENT_UNSPECIFIED = 0
# This event indicates that the server has detected the end of the user's
# speech utterance and expects no additional speech. Therefore, the server
# will not process additional audio (although it may subsequently return
# additional results). The client should stop sending additional audio
# data, half-close the gRPC connection, and wait for any additional results
# until the server closes the gRPC connection. This event is only sent if
# `single_utterance` was set to `true`, and is not used otherwise.
END_OF_SINGLE_UTTERANCE = 1
end
end
# A streaming speech recognition result corresponding to a portion of the audio
# that is currently being processed.
# @!attribute [rw] alternatives
# @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechRecognitionAlternative>]
# Output only. May contain one or more recognition hypotheses (up to the
# maximum specified in `max_alternatives`).
# These alternatives are ordered in terms of accuracy, with the top (first)
# alternative being the most probable, as ranked by the recognizer.
# @!attribute [rw] is_final
# @return [true, false]
# Output only. If `false`, this `StreamingRecognitionResult` represents an
# interim result that may change. If `true`, this is the final time the
# speech service will return this particular `StreamingRecognitionResult`,
# the recognizer will not return any further hypotheses for this portion of
# the transcript and corresponding audio.
# @!attribute [rw] stability
# @return [Float]
# Output only. An estimate of the likelihood that the recognizer will not
# change its guess about this interim result. Values range from 0.0
# (completely unstable) to 1.0 (completely stable).
# This field is only provided for interim results (`is_final=false`).
# The default of 0.0 is a sentinel value indicating `stability` was not set.
# @!attribute [rw] result_end_time
# @return [Google::Protobuf::Duration]
# Output only. Time offset of the end of this result relative to the
# beginning of the audio.
# @!attribute [rw] channel_tag
# @return [Integer]
# For multi-channel audio, this is the channel number corresponding to the
# recognized result for the audio from that channel.
# For audio_channel_count = N, its output values can range from '1' to 'N'.
# @!attribute [rw] language_code
# @return [String]
# Output only. The
# [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag of the
# language in this result. This language code was detected to have the most
# likelihood of being spoken in the audio.
class StreamingRecognitionResult; end
# A speech recognition result corresponding to a portion of the audio.
# @!attribute [rw] alternatives
# @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechRecognitionAlternative>]
# Output only. May contain one or more recognition hypotheses (up to the
# maximum specified in `max_alternatives`).
# These alternatives are ordered in terms of accuracy, with the top (first)
# alternative being the most probable, as ranked by the recognizer.
# @!attribute [rw] channel_tag
# @return [Integer]
# For multi-channel audio, this is the channel number corresponding to the
# recognized result for the audio from that channel.
# For audio_channel_count = N, its output values can range from '1' to 'N'.
# @!attribute [rw] language_code
# @return [String]
# Output only. The
# [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag of the
# language in this result. This language code was detected to have the most
# likelihood of being spoken in the audio.
class SpeechRecognitionResult; end
# Alternative hypotheses (a.k.a. n-best list).
# @!attribute [rw] transcript
# @return [String]
# Output only. Transcript text representing the words that the user spoke.
# @!attribute [rw] confidence
# @return [Float]
# Output only. The confidence estimate between 0.0 and 1.0. A higher number
# indicates an estimated greater likelihood that the recognized words are
# correct. This field is set only for the top alternative of a non-streaming
# result or, of a streaming result where `is_final=true`.
# This field is not guaranteed to be accurate and users should not rely on it
# to be always provided.
# The default of 0.0 is a sentinel value indicating `confidence` was not set.
# @!attribute [rw] words
# @return [Array<Google::Cloud::Speech::V1p1beta1::WordInfo>]
# Output only. A list of word-specific information for each recognized word.
# Note: When `enable_speaker_diarization` is true, you will see all the words
# from the beginning of the audio.
class SpeechRecognitionAlternative; end
# Word-specific information for recognized words.
# @!attribute [rw] start_time
# @return [Google::Protobuf::Duration]
# Output only. Time offset relative to the beginning of the audio,
# and corresponding to the start of the spoken word.
# This field is only set if `enable_word_time_offsets=true` and only
# in the top hypothesis.
# This is an experimental feature and the accuracy of the time offset can
# vary.
# @!attribute [rw] end_time
# @return [Google::Protobuf::Duration]
# Output only. Time offset relative to the beginning of the audio,
# and corresponding to the end of the spoken word.
# This field is only set if `enable_word_time_offsets=true` and only
# in the top hypothesis.
# This is an experimental feature and the accuracy of the time offset can
# vary.
# @!attribute [rw] word
# @return [String]
# Output only. The word corresponding to this set of information.
# @!attribute [rw] confidence
# @return [Float]
# Output only. The confidence estimate between 0.0 and 1.0. A higher number
# indicates an estimated greater likelihood that the recognized words are
# correct. This field is set only for the top alternative of a non-streaming
# result or, of a streaming result where `is_final=true`.
# This field is not guaranteed to be accurate and users should not rely on it
# to be always provided.
# The default of 0.0 is a sentinel value indicating `confidence` was not set.
# @!attribute [rw] speaker_tag
# @return [Integer]
# Output only. A distinct integer value is assigned for every speaker within
# the audio. This field specifies which one of those speakers was detected to
# have spoken this word. Value ranges from '1' to diarization_speaker_count.
# speaker_tag is set if enable_speaker_diarization = 'true' and only in the
# top alternative.
class WordInfo; end
end
end
end
end