lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb in google-cloud-speech-0.31.0 vs lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb in google-cloud-speech-0.31.1
- old
+ new
@@ -15,49 +15,49 @@
module Google
module Cloud
module Speech
module V1p1beta1
- # The top-level message sent by the client for the +Recognize+ method.
+ # 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+
+ # 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.
+ # 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.
+ # 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
+ # 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
# [audio limits](https://cloud.google.com/speech/limits#content).
class StreamingRecognizeRequest; end
# Provides information to the recognizer that specifies how to process the
@@ -66,53 +66,53 @@
# @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
+ # *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+.
+ # `StreamingRecognitionResult`s with the `is_final` flag set to `true`.
#
- # If +true+, the recognizer will detect a single spoken utterance. When it
+ # 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+.
+ # `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
+ # *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.
+ # 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
+ # 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.
+ # `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
+ # 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 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).
+ # 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]
@@ -144,35 +144,35 @@
# 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
+ # 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
+ # *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
+ # 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* A means to provide context to assist the speech recognition.
# @!attribute [rw] enable_word_time_offsets
# @return [true, false]
- # *Optional* If +true+, the top result includes a list of words and
+ # *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+.
+ # `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+.
+ # *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.
@@ -233,83 +233,83 @@
# </tr>
# </table>
# @!attribute [rw] use_enhanced
# @return [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
+ # 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](https://cloud.google.com/speech/data-sharing). If you set
- # +use_enhanced+ to true and you have not enabled audio logging, then you
+ # `use_enhanced` to true and you have not enabled audio logging, then you
# will receive an error.
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
+ # 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+.
+ # `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
+ # 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
+ # 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
+ # `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
+ # 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.
+ # `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.
+ # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
AMR = 4
- # Adaptive Multi-Rate Wideband codec. +sample_rate_hertz+ must be 16000.
+ # 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.
+ # `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
+ # 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+.
+ # `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.
+ # wideband is supported. `sample_rate_hertz` must be 16000.
SPEEX_WITH_HEADER_BYTE = 7
end
end
# Description of audio data to be recognized.
@@ -336,12 +336,12 @@
# 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+.
+ # 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
@@ -453,52 +453,52 @@
# 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/limits#content).
class SpeechContext; end
- # Contains audio data in the encoding specified in the +RecognitionConfig+.
- # Either +content+ or +uri+ must be supplied. Supplying both or neither
+ # 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
# [audio limits](https://cloud.google.com/speech/limits#content).
# @!attribute [rw] content
# @return [String]
# The audio data bytes encoded as specified in
- # +RecognitionConfig+. Note: as with all bytes fields, protobuffers use a
+ # `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+. Currently, only Google Cloud Storage URIs are
+ # `RecognitionConfig`. 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
+ # `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+
+ # 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+
+ # 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.
+ # 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
@@ -507,17 +507,17 @@
# @!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+
+ # `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
+ # 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
+ # 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 }
@@ -541,35 +541,35 @@
# 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
+ # 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
+ # * 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+.
+ # change. A UI designer might choose to show only high stability `results`.
#
- # * The specific +stability+ and +confidence+ values shown above are only for
+ # * 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+.
+ # `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).
+ # 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.
@@ -581,37 +581,37 @@
# 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.
+ # `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+).
+ # 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+,
+ # 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.
+ # 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] 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'.
@@ -625,11 +625,11 @@
# 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+).
+ # 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
@@ -650,14 +650,14 @@
# @!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+.
+ # 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.
+ # 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.
@@ -666,19 +666,19 @@
# 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
+ # 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
+ # 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]
@@ -686,13 +686,13 @@
# @!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+.
+ # 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.
+ # 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.
\ No newline at end of file