# 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] # *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] # *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: When this is true, 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] diarization_speaker_count # @return [Integer] # *Optional* # If set, specifies the estimated number of speakers in the conversation. # If not set, defaults to '2'. # Ignored unless enable_speaker_diarization is set to true." # @!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. # # # # # # # # # # # # # # # # # # # # # #
ModelDescription
command_and_searchBest for short queries such as voice commands or voice search.
phone_callBest for audio that originated from a phone call (typically # recorded at an 8khz sampling rate).
videoBest 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.
defaultBest 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.
# @!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: # 1. project is eligible for requesting enhanced models # 2. 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. # # Enhanced speech models require that you opt-in to data logging using # instructions in the # [documentation](https://cloud.google.com/speech-to-text/docs/enable-data-logging). If you set # `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 # recognition can be reduced if lossy codecs are used to capture or transmit # audio, particularly if background noise is present. Lossy codecs include # `MULAW`, `AMR`, `AMR_WB`, `OGG_OPUS`, and `SPEEX_WITH_HEADER_BYTE`. # # The `FLAC` and `WAV` audio file formats include a header that describes the # included audio content. You can request recognition for `WAV` files that # contain either `LINEAR16` or `MULAW` encoded audio. # If you send `FLAC` or `WAV` audio file format in # your request, you do not need to specify an `AudioEncoding`; the audio # encoding format is determined from the file header. If you specify # an `AudioEncoding` when you send send `FLAC` or `WAV` audio, the # encoding configuration must match the encoding described in the audio # header; otherwise the request returns an # {Google::Rpc::Code::INVALID_ARGUMENT} error code. module AudioEncoding # Not specified. ENCODING_UNSPECIFIED = 0 # Uncompressed 16-bit signed little-endian samples (Linear PCM). LINEAR16 = 1 # `FLAC` (Free Lossless Audio # Codec) is the recommended encoding because it is # lossless--therefore recognition is not compromised--and # requires only about half the bandwidth of `LINEAR16`. `FLAC` stream # encoding supports 16-bit and 24-bit samples, however, not all fields in # `STREAMINFO` are supported. FLAC = 2 # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law. MULAW = 3 # Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000. AMR = 4 # Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000. AMR_WB = 5 # Opus encoded audio frames in Ogg container # ([OggOpus](https://wiki.xiph.org/OggOpus)). # `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000, or 48000. OGG_OPUS = 6 # Although the use of lossy encodings is not recommended, if a very low # bitrate encoding is required, `OGG_OPUS` is highly preferred over # Speex encoding. The [Speex](https://speex.org/) encoding supported by # Cloud Speech API has a header byte in each block, as in MIME type # `audio/x-speex-with-header-byte`. # It is a variant of the RTP Speex encoding defined in # [RFC 5574](https://tools.ietf.org/html/rfc5574). # The stream is a sequence of blocks, one block per RTP packet. Each block # starts with a byte containing the length of the block, in bytes, followed # by one or more frames of Speex data, padded to an integral number of # bytes (octets) as specified in RFC 5574. In other words, each RTP header # is replaced with a single byte containing the block length. Only Speex # wideband is supported. `sample_rate_hertz` must be 16000. SPEEX_WITH_HEADER_BYTE = 7 end end # 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] # *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). 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] # 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] # 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] # 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] # 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] # 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] # 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