diff --git a/packages/google-cloud-python-speech/google/cloud/speech_v1/proto/cloud_speech.proto b/packages/google-cloud-python-speech/google/cloud/speech_v1/proto/cloud_speech.proto
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--- a/packages/google-cloud-python-speech/google/cloud/speech_v1/proto/cloud_speech.proto
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@@ -1,759 +0,0 @@
-// 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
-//
-// http://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.
-//
-
-syntax = "proto3";
-
-package google.cloud.speech.v1;
-
-import "google/api/annotations.proto";
-import "google/api/client.proto";
-import "google/api/field_behavior.proto";
-import "google/longrunning/operations.proto";
-import "google/protobuf/any.proto";
-import "google/protobuf/duration.proto";
-import "google/protobuf/timestamp.proto";
-import "google/rpc/status.proto";
-
-option cc_enable_arenas = true;
-option go_package = "google.golang.org/genproto/googleapis/cloud/speech/v1;speech";
-option java_multiple_files = true;
-option java_outer_classname = "SpeechProto";
-option java_package = "com.google.cloud.speech.v1";
-option objc_class_prefix = "GCS";
-
-// Service that implements Google Cloud Speech API.
-service Speech {
- option (google.api.default_host) = "speech.googleapis.com";
- option (google.api.oauth_scopes) = "https://www.googleapis.com/auth/cloud-platform";
-
- // Performs synchronous speech recognition: receive results after all audio
- // has been sent and processed.
- rpc Recognize(RecognizeRequest) returns (RecognizeResponse) {
- option (google.api.http) = {
- post: "/v1/speech:recognize"
- body: "*"
- };
- option (google.api.method_signature) = "config,audio";
- }
-
- // Performs asynchronous speech recognition: receive results via the
- // google.longrunning.Operations interface. Returns either an
- // `Operation.error` or an `Operation.response` which contains
- // a `LongRunningRecognizeResponse` message.
- // For more information on asynchronous speech recognition, see the
- // [how-to](https://cloud.google.com/speech-to-text/docs/async-recognize).
- rpc LongRunningRecognize(LongRunningRecognizeRequest) returns (google.longrunning.Operation) {
- option (google.api.http) = {
- post: "/v1/speech:longrunningrecognize"
- body: "*"
- };
- option (google.api.method_signature) = "config,audio";
- option (google.longrunning.operation_info) = {
- response_type: "LongRunningRecognizeResponse"
- metadata_type: "LongRunningRecognizeMetadata"
- };
- }
-
- // Performs bidirectional streaming speech recognition: receive results while
- // sending audio. This method is only available via the gRPC API (not REST).
- rpc StreamingRecognize(stream StreamingRecognizeRequest) returns (stream StreamingRecognizeResponse) {
- }
-}
-
-// The top-level message sent by the client for the `Recognize` method.
-message RecognizeRequest {
- // Required. Provides information to the recognizer that specifies how to
- // process the request.
- RecognitionConfig config = 1 [(google.api.field_behavior) = REQUIRED];
-
- // Required. The audio data to be recognized.
- RecognitionAudio audio = 2 [(google.api.field_behavior) = REQUIRED];
-}
-
-// The top-level message sent by the client for the `LongRunningRecognize`
-// method.
-message LongRunningRecognizeRequest {
- // Required. Provides information to the recognizer that specifies how to
- // process the request.
- RecognitionConfig config = 1 [(google.api.field_behavior) = REQUIRED];
-
- // Required. The audio data to be recognized.
- RecognitionAudio audio = 2 [(google.api.field_behavior) = REQUIRED];
-}
-
-// 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_content`. All subsequent messages must contain `audio_content` and
-// must not contain a `streaming_config` message.
-message StreamingRecognizeRequest {
- // The streaming request, which is either a streaming config or audio content.
- oneof streaming_request {
- // Provides information to the recognizer that specifies how to process the
- // request. The first `StreamingRecognizeRequest` message must contain a
- // `streaming_config` message.
- StreamingRecognitionConfig streaming_config = 1;
-
- // 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, proto buffers use a
- // pure binary representation (not base64). See
- // [content limits](https://cloud.google.com/speech-to-text/quotas#content).
- bytes audio_content = 2;
- }
-}
-
-// Provides information to the recognizer that specifies how to process the
-// request.
-message StreamingRecognitionConfig {
- // Required. Provides information to the recognizer that specifies how to
- // process the request.
- RecognitionConfig config = 1 [(google.api.field_behavior) = REQUIRED];
-
- // 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`.
- bool single_utterance = 2;
-
- // 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.
- bool interim_results = 3;
-}
-
-// Provides information to the recognizer that specifies how to process the
-// request.
-message RecognitionConfig {
- // The encoding of the audio data sent in the request.
- //
- // All encodings support only 1 channel (mono) audio, unless the
- // `audio_channel_count` and `enable_separate_recognition_per_channel` fields
- // are set.
- //
- // 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`, `SPEEX_WITH_HEADER_BYTE`, and `MP3`.
- //
- // 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][google.rpc.Code.INVALID_ARGUMENT] error code.
- enum 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;
- }
-
- // 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 [AudioEncoding][google.cloud.speech.v1.RecognitionConfig.AudioEncoding].
- AudioEncoding encoding = 1;
-
- // 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, but is
- // required for all other audio formats. For details, see [AudioEncoding][google.cloud.speech.v1.RecognitionConfig.AudioEncoding].
- int32 sample_rate_hertz = 2;
-
- // 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'.
- int32 audio_channel_count = 7;
-
- // 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.
- bool enable_separate_recognition_per_channel = 12;
-
- // 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.
- string language_code = 3 [(google.api.field_behavior) = REQUIRED];
-
- // 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.
- int32 max_alternatives = 4;
-
- // 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.
- bool profanity_filter = 5;
-
- // Array of [SpeechContext][google.cloud.speech.v1.SpeechContext].
- // A means to provide context to assist the speech recognition. For more
- // information, see
- // [speech
- // adaptation](https://cloud.google.com/speech-to-text/docs/context-strength).
- repeated SpeechContext speech_contexts = 6;
-
- // 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`.
- bool enable_word_time_offsets = 8;
-
- // 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.
- bool enable_automatic_punctuation = 11;
-
- // 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.
- SpeakerDiarizationConfig diarization_config = 19;
-
- // Metadata regarding this request.
- RecognitionMetadata metadata = 9;
-
- // 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.
- //
- //
- // Model |
- // Description |
- //
- //
- // command_and_search |
- // Best for short queries such as voice commands or voice search. |
- //
- //
- // phone_call |
- // Best for audio that originated from a phone call (typically
- // recorded at an 8khz sampling rate). |
- //
- //
- // video |
- // 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. |
- //
- //
- // default |
- // 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. |
- //
- //
- string model = 13;
-
- // 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.
- bool use_enhanced = 14;
-}
-
-// Config to enable speaker diarization.
-message SpeakerDiarizationConfig {
- // 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.
- bool enable_speaker_diarization = 1;
-
- // 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.
- int32 min_speaker_count = 2;
-
- // 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.
- int32 max_speaker_count = 3;
-
- // Unused.
- int32 speaker_tag = 5
- [(google.api.field_behavior) = OUTPUT_ONLY, deprecated = true];
-}
-
-// Description of audio data to be recognized.
-message RecognitionMetadata {
- // Use case categories that the audio recognition request can be described
- // by.
- enum 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;
- }
-
- // Enumerates the types of capture settings describing an audio file.
- enum 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;
- }
-
- // The original media the speech was recorded on.
- enum 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;
- }
-
- // The type of device the speech was recorded with.
- enum 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;
- }
-
- // The use case most closely describing the audio content to be recognized.
- InteractionType interaction_type = 1;
-
- // 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/.
- uint32 industry_naics_code_of_audio = 3;
-
- // The audio type that most closely describes the audio being recognized.
- MicrophoneDistance microphone_distance = 4;
-
- // The original media the speech was recorded on.
- OriginalMediaType original_media_type = 5;
-
- // The type of device the speech was recorded with.
- RecordingDeviceType recording_device_type = 6;
-
- // The device used to make the recording. Examples 'Nexus 5X' or
- // 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
- // 'Cardioid Microphone'.
- string recording_device_name = 7;
-
- // 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
- string original_mime_type = 8;
-
- // Description of the content. Eg. "Recordings of federal supreme court
- // hearings from 2012".
- string audio_topic = 10;
-}
-
-// Provides "hints" to the speech recognizer to favor specific words and phrases
-// in the results.
-message SpeechContext {
- // 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.
- repeated string phrases = 1;
-}
-
-// 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][google.rpc.Code.INVALID_ARGUMENT]. See
-// [content limits](https://cloud.google.com/speech-to-text/quotas#content).
-message RecognitionAudio {
- // The audio source, which is either inline content or a Google Cloud
- // Storage uri.
- oneof audio_source {
- // The audio data bytes encoded as specified in
- // `RecognitionConfig`. Note: as with all bytes fields, proto buffers use a
- // pure binary representation, whereas JSON representations use base64.
- bytes content = 1;
-
- // 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][google.rpc.Code.INVALID_ARGUMENT]). For more information, see
- // [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
- string uri = 2;
- }
-}
-
-// The only message returned to the client by the `Recognize` method. It
-// contains the result as zero or more sequential `SpeechRecognitionResult`
-// messages.
-message RecognizeResponse {
- // Sequential list of transcription results corresponding to
- // sequential portions of audio.
- repeated SpeechRecognitionResult results = 2;
-}
-
-// 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.
-message LongRunningRecognizeResponse {
- // Sequential list of transcription results corresponding to
- // sequential portions of audio.
- repeated SpeechRecognitionResult results = 2;
-}
-
-// 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.
-message LongRunningRecognizeMetadata {
- // Approximate percentage of audio processed thus far. Guaranteed to be 100
- // when the audio is fully processed and the results are available.
- int32 progress_percent = 1;
-
- // Time when the request was received.
- google.protobuf.Timestamp start_time = 2;
-
- // Time of the most recent processing update.
- google.protobuf.Timestamp last_update_time = 3;
-}
-
-// `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`.
-message StreamingRecognizeResponse {
- // Indicates the type of speech event.
- enum 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;
- }
-
- // If set, returns a [google.rpc.Status][google.rpc.Status] message that
- // specifies the error for the operation.
- google.rpc.Status error = 1;
-
- // 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).
- repeated StreamingRecognitionResult results = 2;
-
- // Indicates the type of speech event.
- SpeechEventType speech_event_type = 4;
-}
-
-// A streaming speech recognition result corresponding to a portion of the audio
-// that is currently being processed.
-message StreamingRecognitionResult {
- // 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.
- repeated SpeechRecognitionAlternative alternatives = 1;
-
- // 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.
- bool is_final = 2;
-
- // 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.
- float stability = 3;
-
- // Time offset of the end of this result relative to the
- // beginning of the audio.
- google.protobuf.Duration result_end_time = 4;
-
- // 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'.
- int32 channel_tag = 5;
-
- // 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.
- string language_code = 6
- [(google.api.field_behavior) = OUTPUT_ONLY];
-}
-
-// A speech recognition result corresponding to a portion of the audio.
-message SpeechRecognitionResult {
- // 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.
- repeated SpeechRecognitionAlternative alternatives = 1;
-
- // 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'.
- int32 channel_tag = 2;
-}
-
-// Alternative hypotheses (a.k.a. n-best list).
-message SpeechRecognitionAlternative {
- // Transcript text representing the words that the user spoke.
- string transcript = 1;
-
- // 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.
- float confidence = 2;
-
- // 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.
- repeated WordInfo words = 3;
-}
-
-// Word-specific information for recognized words.
-message WordInfo {
- // 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.
- google.protobuf.Duration start_time = 1;
-
- // 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.
- google.protobuf.Duration end_time = 2;
-
- // The word corresponding to this set of information.
- string word = 3;
-
- // 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.
- int32 speaker_tag = 5
- [(google.api.field_behavior) = OUTPUT_ONLY];
-}
diff --git a/packages/google-cloud-python-speech/google/cloud/speech_v1p1beta1/proto/cloud_speech.proto b/packages/google-cloud-python-speech/google/cloud/speech_v1p1beta1/proto/cloud_speech.proto
deleted file mode 100644
index cf183d0129d3..000000000000
--- a/packages/google-cloud-python-speech/google/cloud/speech_v1p1beta1/proto/cloud_speech.proto
+++ /dev/null
@@ -1,881 +0,0 @@
-
-// Copyright 2021 Google LLC
-//
-// Licensed under the Apache License, Version 2.0 (the "License");
-// you may not use this file except in compliance with the License.
-// You may obtain a copy of the License at
-//
-// http://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.
-
-syntax = "proto3";
-
-package google.cloud.speech.v1p1beta1;
-
-import "google/api/annotations.proto";
-import "google/api/client.proto";
-import "google/api/field_behavior.proto";
-import "google/cloud/speech/v1p1beta1/resource.proto";
-import "google/longrunning/operations.proto";
-import "google/protobuf/any.proto";
-import "google/protobuf/duration.proto";
-import "google/protobuf/timestamp.proto";
-import "google/protobuf/wrappers.proto";
-import "google/rpc/status.proto";
-
-option cc_enable_arenas = true;
-option go_package = "google.golang.org/genproto/googleapis/cloud/speech/v1p1beta1;speech";
-option java_multiple_files = true;
-option java_outer_classname = "SpeechProto";
-option java_package = "com.google.cloud.speech.v1p1beta1";
-option objc_class_prefix = "GCS";
-
-// Service that implements Google Cloud Speech API.
-service Speech {
- option (google.api.default_host) = "speech.googleapis.com";
- option (google.api.oauth_scopes) = "https://www.googleapis.com/auth/cloud-platform";
-
- // Performs synchronous speech recognition: receive results after all audio
- // has been sent and processed.
- rpc Recognize(RecognizeRequest) returns (RecognizeResponse) {
- option (google.api.http) = {
- post: "/v1p1beta1/speech:recognize"
- body: "*"
- };
- option (google.api.method_signature) = "config,audio";
- }
-
- // Performs asynchronous speech recognition: receive results via the
- // google.longrunning.Operations interface. Returns either an
- // `Operation.error` or an `Operation.response` which contains
- // a `LongRunningRecognizeResponse` message.
- // For more information on asynchronous speech recognition, see the
- // [how-to](https://cloud.google.com/speech-to-text/docs/async-recognize).
- rpc LongRunningRecognize(LongRunningRecognizeRequest) returns (google.longrunning.Operation) {
- option (google.api.http) = {
- post: "/v1p1beta1/speech:longrunningrecognize"
- body: "*"
- };
- option (google.api.method_signature) = "config,audio";
- option (google.longrunning.operation_info) = {
- response_type: "LongRunningRecognizeResponse"
- metadata_type: "LongRunningRecognizeMetadata"
- };
- }
-
- // Performs bidirectional streaming speech recognition: receive results while
- // sending audio. This method is only available via the gRPC API (not REST).
- rpc StreamingRecognize(stream StreamingRecognizeRequest) returns (stream StreamingRecognizeResponse) {
- }
-}
-
-// The top-level message sent by the client for the `Recognize` method.
-message RecognizeRequest {
- // Required. Provides information to the recognizer that specifies how to
- // process the request.
- RecognitionConfig config = 1 [(google.api.field_behavior) = REQUIRED];
-
- // Required. The audio data to be recognized.
- RecognitionAudio audio = 2 [(google.api.field_behavior) = REQUIRED];
-}
-
-// The top-level message sent by the client for the `LongRunningRecognize`
-// method.
-message LongRunningRecognizeRequest {
- // Required. Provides information to the recognizer that specifies how to
- // process the request.
- RecognitionConfig config = 1 [(google.api.field_behavior) = REQUIRED];
-
- // Required. The audio data to be recognized.
- RecognitionAudio audio = 2 [(google.api.field_behavior) = REQUIRED];
-
- // Optional. Specifies an optional destination for the recognition results.
- TranscriptOutputConfig output_config = 4 [(google.api.field_behavior) = OPTIONAL];
-}
-
-// Specifies an optional destination for the recognition results.
-message TranscriptOutputConfig {
- oneof output_type {
- // Specifies a Cloud Storage URI for the recognition results. Must be
- // specified in the format: `gs://bucket_name/object_name`, and the bucket
- // must already exist.
- string gcs_uri = 1;
- }
-}
-
-// 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_content`. All subsequent messages must contain `audio_content` and
-// must not contain a `streaming_config` message.
-message StreamingRecognizeRequest {
- // The streaming request, which is either a streaming config or audio content.
- oneof streaming_request {
- // Provides information to the recognizer that specifies how to process the
- // request. The first `StreamingRecognizeRequest` message must contain a
- // `streaming_config` message.
- StreamingRecognitionConfig streaming_config = 1;
-
- // 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, proto buffers use a
- // pure binary representation (not base64). See
- // [content limits](https://cloud.google.com/speech-to-text/quotas#content).
- bytes audio_content = 2;
- }
-}
-
-// Provides information to the recognizer that specifies how to process the
-// request.
-message StreamingRecognitionConfig {
- // Required. Provides information to the recognizer that specifies how to
- // process the request.
- RecognitionConfig config = 1 [(google.api.field_behavior) = REQUIRED];
-
- // 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`.
- //
- // The `single_utterance` field can only be used with specified models,
- // otherwise an error is thrown. The `model` field in [`RecognitionConfig`][]
- // must be set to:
- //
- // * `command_and_search`
- // * `phone_call` AND additional field `useEnhanced`=`true`
- // * The `model` field is left undefined. In this case the API auto-selects
- // a model based on any other parameters that you set in
- // `RecognitionConfig`.
- bool single_utterance = 2;
-
- // 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.
- bool interim_results = 3;
-}
-
-// Provides information to the recognizer that specifies how to process the
-// request.
-message RecognitionConfig {
- // The encoding of the audio data sent in the request.
- //
- // All encodings support only 1 channel (mono) audio, unless the
- // `audio_channel_count` and `enable_separate_recognition_per_channel` fields
- // are set.
- //
- // 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`, `SPEEX_WITH_HEADER_BYTE`, `MP3`.
- //
- // 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][google.rpc.Code.INVALID_ARGUMENT] error code.
- enum 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. MP3 encoding is a Beta feature and only available in
- // v1p1beta1. Support all standard MP3 bitrates (which range from 32-320
- // kbps). When using this encoding, `sample_rate_hertz` has to match the
- // sample rate of the file being used.
- MP3 = 8;
- }
-
- // 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 [AudioEncoding][google.cloud.speech.v1p1beta1.RecognitionConfig.AudioEncoding].
- AudioEncoding encoding = 1;
-
- // 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, but is
- // required for all other audio formats. For details, see [AudioEncoding][google.cloud.speech.v1p1beta1.RecognitionConfig.AudioEncoding].
- int32 sample_rate_hertz = 2;
-
- // 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'.
- int32 audio_channel_count = 7;
-
- // 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.
- bool enable_separate_recognition_per_channel = 12;
-
- // 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.
- string language_code = 3 [(google.api.field_behavior) = REQUIRED];
-
- // 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).
- repeated string alternative_language_codes = 18;
-
- // 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.
- int32 max_alternatives = 4;
-
- // 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.
- bool profanity_filter = 5;
-
- // Speech adaptation configuration improves the accuracy of speech
- // recognition. When speech adaptation is set it supersedes the
- // `speech_contexts` field. For more information, see the [speech
- // adaptation](https://cloud.google.com/speech-to-text/docs/adaptation)
- // documentation.
- SpeechAdaptation adaptation = 20;
-
- // Array of [SpeechContext][google.cloud.speech.v1p1beta1.SpeechContext].
- // A means to provide context to assist the speech recognition. For more
- // information, see
- // [speech
- // adaptation](https://cloud.google.com/speech-to-text/docs/adaptation).
- repeated SpeechContext speech_contexts = 6;
-
- // 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`.
- bool enable_word_time_offsets = 8;
-
- // 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`.
- bool enable_word_confidence = 15;
-
- // 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.
- bool enable_automatic_punctuation = 11;
-
- // The spoken punctuation behavior for the call
- // If not set, uses default behavior based on model of choice
- // e.g. command_and_search will enable spoken punctuation by default
- // If 'true', replaces spoken punctuation with the corresponding symbols in
- // the request. For example, "how are you question mark" becomes "how are
- // you?". See https://cloud.google.com/speech-to-text/docs/spoken-punctuation
- // for support. If 'false', spoken punctuation is not replaced.
- google.protobuf.BoolValue enable_spoken_punctuation = 22;
-
- // The spoken emoji behavior for the call
- // If not set, uses default behavior based on model of choice
- // If 'true', adds spoken emoji formatting for the request. This will replace
- // spoken emojis with the corresponding Unicode symbols in the final
- // transcript. If 'false', spoken emojis are not replaced.
- google.protobuf.BoolValue enable_spoken_emojis = 23;
-
- // 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.
- bool enable_speaker_diarization = 16 [deprecated = true];
-
- // 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.
- int32 diarization_speaker_count = 17 [deprecated = true];
-
- // 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.
- SpeakerDiarizationConfig diarization_config = 19;
-
- // Metadata regarding this request.
- RecognitionMetadata metadata = 9;
-
- // 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.
- //
- //
- // Model |
- // Description |
- //
- //
- // command_and_search |
- // Best for short queries such as voice commands or voice search. |
- //
- //
- // phone_call |
- // Best for audio that originated from a phone call (typically
- // recorded at an 8khz sampling rate). |
- //
- //
- // video |
- // 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. |
- //
- //
- // default |
- // 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. |
- //
- //
- string model = 13;
-
- // 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.
- bool use_enhanced = 14;
-}
-
-// Config to enable speaker diarization.
-message SpeakerDiarizationConfig {
- // 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.
- bool enable_speaker_diarization = 1;
-
- // 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.
- int32 min_speaker_count = 2;
-
- // 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.
- int32 max_speaker_count = 3;
-
- // Output only. Unused.
- int32 speaker_tag = 5 [
- deprecated = true,
- (google.api.field_behavior) = OUTPUT_ONLY
- ];
-}
-
-// Description of audio data to be recognized.
-message RecognitionMetadata {
- // Use case categories that the audio recognition request can be described
- // by.
- enum 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;
- }
-
- // Enumerates the types of capture settings describing an audio file.
- enum 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;
- }
-
- // The original media the speech was recorded on.
- enum 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;
- }
-
- // The type of device the speech was recorded with.
- enum 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;
- }
-
- // The use case most closely describing the audio content to be recognized.
- InteractionType interaction_type = 1;
-
- // 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/.
- uint32 industry_naics_code_of_audio = 3;
-
- // The audio type that most closely describes the audio being recognized.
- MicrophoneDistance microphone_distance = 4;
-
- // The original media the speech was recorded on.
- OriginalMediaType original_media_type = 5;
-
- // The type of device the speech was recorded with.
- RecordingDeviceType recording_device_type = 6;
-
- // The device used to make the recording. Examples 'Nexus 5X' or
- // 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
- // 'Cardioid Microphone'.
- string recording_device_name = 7;
-
- // 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
- string original_mime_type = 8;
-
- // Obfuscated (privacy-protected) ID of the user, to identify number of
- // unique users using the service.
- int64 obfuscated_id = 9 [deprecated = true];
-
- // Description of the content. Eg. "Recordings of federal supreme court
- // hearings from 2012".
- string audio_topic = 10;
-}
-
-// Provides "hints" to the speech recognizer to favor specific words and phrases
-// in the results.
-message SpeechContext {
- // 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.
- repeated string phrases = 1;
-
- // 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.
- float boost = 4;
-}
-
-// 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][google.rpc.Code.INVALID_ARGUMENT]. See
-// [content limits](https://cloud.google.com/speech-to-text/quotas#content).
-message RecognitionAudio {
- // The audio source, which is either inline content or a Google Cloud
- // Storage uri.
- oneof audio_source {
- // The audio data bytes encoded as specified in
- // `RecognitionConfig`. Note: as with all bytes fields, proto buffers use a
- // pure binary representation, whereas JSON representations use base64.
- bytes content = 1;
-
- // 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][google.rpc.Code.INVALID_ARGUMENT]). For more information, see
- // [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
- string uri = 2;
- }
-}
-
-// The only message returned to the client by the `Recognize` method. It
-// contains the result as zero or more sequential `SpeechRecognitionResult`
-// messages.
-message RecognizeResponse {
- // Sequential list of transcription results corresponding to
- // sequential portions of audio.
- repeated SpeechRecognitionResult results = 2;
-}
-
-// 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.
-message LongRunningRecognizeResponse {
- // Sequential list of transcription results corresponding to
- // sequential portions of audio.
- repeated SpeechRecognitionResult results = 2;
-
- // Original output config if present in the request.
- TranscriptOutputConfig output_config = 6;
-
- // If the transcript output fails this field contains the relevant error.
- google.rpc.Status output_error = 7;
-}
-
-// 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.
-message LongRunningRecognizeMetadata {
- // Approximate percentage of audio processed thus far. Guaranteed to be 100
- // when the audio is fully processed and the results are available.
- int32 progress_percent = 1;
-
- // Time when the request was received.
- google.protobuf.Timestamp start_time = 2;
-
- // Time of the most recent processing update.
- google.protobuf.Timestamp last_update_time = 3;
-
- // Output only. The URI of the audio file being transcribed. Empty if the audio was sent
- // as byte content.
- string uri = 4 [(google.api.field_behavior) = OUTPUT_ONLY];
-
- // Output only. A copy of the TranscriptOutputConfig if it was set in the request.
- TranscriptOutputConfig output_config = 5 [(google.api.field_behavior) = OUTPUT_ONLY];
-}
-
-// `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 `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`.
-message StreamingRecognizeResponse {
- // Indicates the type of speech event.
- enum 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;
- }
-
- // If set, returns a [google.rpc.Status][google.rpc.Status] message that
- // specifies the error for the operation.
- google.rpc.Status error = 1;
-
- // 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).
- repeated StreamingRecognitionResult results = 2;
-
- // Indicates the type of speech event.
- SpeechEventType speech_event_type = 4;
-}
-
-// A streaming speech recognition result corresponding to a portion of the audio
-// that is currently being processed.
-message StreamingRecognitionResult {
- // 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.
- repeated SpeechRecognitionAlternative alternatives = 1;
-
- // 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.
- bool is_final = 2;
-
- // 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.
- float stability = 3;
-
- // Time offset of the end of this result relative to the
- // beginning of the audio.
- google.protobuf.Duration result_end_time = 4;
-
- // 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'.
- int32 channel_tag = 5;
-
- // 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.
- string language_code = 6 [(google.api.field_behavior) = OUTPUT_ONLY];
-}
-
-// A speech recognition result corresponding to a portion of the audio.
-message SpeechRecognitionResult {
- // 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.
- repeated SpeechRecognitionAlternative alternatives = 1;
-
- // 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'.
- int32 channel_tag = 2;
-
- // 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.
- string language_code = 5 [(google.api.field_behavior) = OUTPUT_ONLY];
-}
-
-// Alternative hypotheses (a.k.a. n-best list).
-message SpeechRecognitionAlternative {
- // Transcript text representing the words that the user spoke.
- string transcript = 1;
-
- // 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.
- float confidence = 2;
-
- // 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.
- repeated WordInfo words = 3;
-}
-
-// Word-specific information for recognized words.
-message WordInfo {
- // 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.
- google.protobuf.Duration start_time = 1;
-
- // 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.
- google.protobuf.Duration end_time = 2;
-
- // The word corresponding to this set of information.
- string word = 3;
-
- // 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.
- float confidence = 4;
-
- // 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.
- int32 speaker_tag = 5 [(google.api.field_behavior) = OUTPUT_ONLY];
-}
diff --git a/packages/google-cloud-python-speech/google/cloud/speech_v1p1beta1/proto/cloud_speech_adaptation.proto b/packages/google-cloud-python-speech/google/cloud/speech_v1p1beta1/proto/cloud_speech_adaptation.proto
deleted file mode 100644
index 16789739d6e2..000000000000
--- a/packages/google-cloud-python-speech/google/cloud/speech_v1p1beta1/proto/cloud_speech_adaptation.proto
+++ /dev/null
@@ -1,320 +0,0 @@
-// Copyright 2021 Google LLC
-//
-// Licensed under the Apache License, Version 2.0 (the "License");
-// you may not use this file except in compliance with the License.
-// You may obtain a copy of the License at
-//
-// http://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.
-
-syntax = "proto3";
-
-package google.cloud.speech.v1p1beta1;
-
-import "google/api/annotations.proto";
-import "google/api/client.proto";
-import "google/api/field_behavior.proto";
-import "google/api/resource.proto";
-import "google/cloud/speech/v1p1beta1/resource.proto";
-import "google/protobuf/empty.proto";
-import "google/protobuf/field_mask.proto";
-
-option cc_enable_arenas = true;
-option go_package = "google.golang.org/genproto/googleapis/cloud/speech/v1p1beta1;speech";
-option java_multiple_files = true;
-option java_outer_classname = "SpeechAdaptationProto";
-option java_package = "com.google.cloud.speech.v1p1beta1";
-option objc_class_prefix = "GCS";
-
-// Service that implements Google Cloud Speech Adaptation API.
-service Adaptation {
- option (google.api.default_host) = "speech.googleapis.com";
- option (google.api.oauth_scopes) = "https://www.googleapis.com/auth/cloud-platform";
-
- // Create a set of phrase hints. Each item in the set can be a single word or
- // a multi-word phrase. The items in the PhraseSet are favored by the
- // recognition model when you send a call that includes the PhraseSet.
- rpc CreatePhraseSet(CreatePhraseSetRequest) returns (PhraseSet) {
- option (google.api.http) = {
- post: "/v1p1beta1/{parent=projects/*/locations/*}/phraseSets"
- body: "*"
- };
- option (google.api.method_signature) = "parent,phrase_set,phrase_set_id";
- }
-
- // Get a phrase set.
- rpc GetPhraseSet(GetPhraseSetRequest) returns (PhraseSet) {
- option (google.api.http) = {
- get: "/v1p1beta1/{name=projects/*/locations/*/phraseSets/*}"
- };
- option (google.api.method_signature) = "name";
- }
-
- // List phrase sets.
- rpc ListPhraseSet(ListPhraseSetRequest) returns (ListPhraseSetResponse) {
- option (google.api.http) = {
- get: "/v1p1beta1/{parent=projects/*/locations/*}/phraseSets"
- };
- option (google.api.method_signature) = "parent";
- }
-
- // Update a phrase set.
- rpc UpdatePhraseSet(UpdatePhraseSetRequest) returns (PhraseSet) {
- option (google.api.http) = {
- patch: "/v1p1beta1/{phrase_set.name=projects/*/locations/*/phraseSets/*}"
- body: "phrase_set"
- };
- option (google.api.method_signature) = "phrase_set,update_mask";
- }
-
- // Delete a phrase set.
- rpc DeletePhraseSet(DeletePhraseSetRequest) returns (google.protobuf.Empty) {
- option (google.api.http) = {
- delete: "/v1p1beta1/{name=projects/*/locations/*/phraseSets/*}"
- };
- option (google.api.method_signature) = "name";
- }
-
- // Create a custom class.
- rpc CreateCustomClass(CreateCustomClassRequest) returns (CustomClass) {
- option (google.api.http) = {
- post: "/v1p1beta1/{parent=projects/*/locations/*}/customClasses"
- body: "*"
- };
- option (google.api.method_signature) = "parent,custom_class,custom_class_id";
- }
-
- // Get a custom class.
- rpc GetCustomClass(GetCustomClassRequest) returns (CustomClass) {
- option (google.api.http) = {
- get: "/v1p1beta1/{name=projects/*/locations/*/customClasses/*}"
- };
- option (google.api.method_signature) = "name";
- }
-
- // List custom classes.
- rpc ListCustomClasses(ListCustomClassesRequest) returns (ListCustomClassesResponse) {
- option (google.api.http) = {
- get: "/v1p1beta1/{parent=projects/*/locations/*}/customClasses"
- };
- option (google.api.method_signature) = "parent";
- }
-
- // Update a custom class.
- rpc UpdateCustomClass(UpdateCustomClassRequest) returns (CustomClass) {
- option (google.api.http) = {
- patch: "/v1p1beta1/{custom_class.name=projects/*/locations/*/customClasses/*}"
- body: "custom_class"
- };
- option (google.api.method_signature) = "custom_class,update_mask";
- }
-
- // Delete a custom class.
- rpc DeleteCustomClass(DeleteCustomClassRequest) returns (google.protobuf.Empty) {
- option (google.api.http) = {
- delete: "/v1p1beta1/{name=projects/*/locations/*/customClasses/*}"
- };
- option (google.api.method_signature) = "name";
- }
-}
-
-// Message sent by the client for the `CreatePhraseSet` method.
-message CreatePhraseSetRequest {
- // Required. The parent resource where this phrase set will be created.
- // Format:
- // {api_version}/projects/{project}/locations/{location}/phraseSets
- string parent = 1 [
- (google.api.field_behavior) = REQUIRED,
- (google.api.resource_reference) = {
- child_type: "speech.googleapis.com/PhraseSet"
- }
- ];
-
- // The ID to use for the phrase set, which will become the final
- // component of the phrase set's resource name.
- //
- // This value should be 4-63 characters, and valid characters
- // are /[a-z][0-9]-/.
- string phrase_set_id = 2;
-
- // Required. The phrase set to create.
- PhraseSet phrase_set = 3 [(google.api.field_behavior) = REQUIRED];
-}
-
-// Message sent by the client for the `UpdatePhraseSet` method.
-message UpdatePhraseSetRequest {
- // Required. The phrase set to update.
- //
- // The phrase set's `name` field is used to identify the set to be
- // updated. Format:
- // {api_version}/projects/{project}/locations/{location}/phraseSets/{phrase_set}
- PhraseSet phrase_set = 1 [(google.api.field_behavior) = REQUIRED];
-
- // The list of fields to be updated.
- google.protobuf.FieldMask update_mask = 2;
-}
-
-// Message sent by the client for the `GetPhraseSet` method.
-message GetPhraseSetRequest {
- // Required. The name of the phrase set to retrieve.
- // Format:
- // {api_version}/projects/{project}/locations/{location}/phraseSets/{phrase_set}
- string name = 1 [
- (google.api.field_behavior) = REQUIRED,
- (google.api.resource_reference) = {
- type: "speech.googleapis.com/PhraseSet"
- }
- ];
-}
-
-// Message sent by the client for the `ListPhraseSet` method.
-message ListPhraseSetRequest {
- // Required. The parent, which owns this collection of phrase set.
- // Format:
- // projects/{project}/locations/{location}
- string parent = 1 [
- (google.api.field_behavior) = REQUIRED,
- (google.api.resource_reference) = {
- child_type: "speech.googleapis.com/PhraseSet"
- }
- ];
-
- // The maximum number of phrase sets to return. The service may return
- // fewer than this value. If unspecified, at most 50 phrase sets will be
- // returned. The maximum value is 1000; values above 1000 will be coerced to
- // 1000.
- int32 page_size = 2;
-
- // A page token, received from a previous `ListPhraseSet` call.
- // Provide this to retrieve the subsequent page.
- //
- // When paginating, all other parameters provided to `ListPhraseSet` must
- // match the call that provided the page token.
- string page_token = 3;
-}
-
-// Message returned to the client by the `ListPhraseSet` method.
-message ListPhraseSetResponse {
- // The phrase set.
- repeated PhraseSet phrase_sets = 1;
-
- // A token, which can be sent as `page_token` to retrieve the next page.
- // If this field is omitted, there are no subsequent pages.
- string next_page_token = 2;
-}
-
-// Message sent by the client for the `DeletePhraseSet` method.
-message DeletePhraseSetRequest {
- // Required. The name of the phrase set to delete.
- // Format:
- // {api_version}/projects/{project}/locations/{location}/phraseSets/{phrase_set}
- string name = 1 [
- (google.api.field_behavior) = REQUIRED,
- (google.api.resource_reference) = {
- type: "speech.googleapis.com/PhraseSet"
- }
- ];
-}
-
-// Message sent by the client for the `CreateCustomClass` method.
-message CreateCustomClassRequest {
- // Required. The parent resource where this custom class will be created.
- // Format:
- // {api_version}/projects/{project}/locations/{location}/customClasses
- string parent = 1 [
- (google.api.field_behavior) = REQUIRED,
- (google.api.resource_reference) = {
- child_type: "speech.googleapis.com/CustomClass"
- }
- ];
-
- // The ID to use for the custom class, which will become the final
- // component of the custom class' resource name.
- //
- // This value should be 4-63 characters, and valid characters
- // are /[a-z][0-9]-/.
- string custom_class_id = 2;
-
- // Required. The custom class to create.
- CustomClass custom_class = 3 [(google.api.field_behavior) = REQUIRED];
-}
-
-// Message sent by the client for the `UpdateCustomClass` method.
-message UpdateCustomClassRequest {
- // Required. The custom class to update.
- //
- // The custom class's `name` field is used to identify the custom class to be
- // updated. Format:
- // {api_version}/projects/{project}/locations/{location}/customClasses/{custom_class}
- CustomClass custom_class = 1 [(google.api.field_behavior) = REQUIRED];
-
- // The list of fields to be updated.
- google.protobuf.FieldMask update_mask = 2;
-}
-
-// Message sent by the client for the `GetCustomClass` method.
-message GetCustomClassRequest {
- // Required. The name of the custom class to retrieve.
- // Format:
- // {api_version}/projects/{project}/locations/{location}/customClasses/{custom_class}
- string name = 1 [
- (google.api.field_behavior) = REQUIRED,
- (google.api.resource_reference) = {
- type: "speech.googleapis.com/CustomClass"
- }
- ];
-}
-
-// Message sent by the client for the `ListCustomClasses` method.
-message ListCustomClassesRequest {
- // Required. The parent, which owns this collection of custom classes.
- // Format:
- // {api_version}/projects/{project}/locations/{location}/customClasses
- string parent = 1 [
- (google.api.field_behavior) = REQUIRED,
- (google.api.resource_reference) = {
- child_type: "speech.googleapis.com/CustomClass"
- }
- ];
-
- // The maximum number of custom classes to return. The service may return
- // fewer than this value. If unspecified, at most 50 custom classes will be
- // returned. The maximum value is 1000; values above 1000 will be coerced to
- // 1000.
- int32 page_size = 2;
-
- // A page token, received from a previous `ListCustomClass` call.
- // Provide this to retrieve the subsequent page.
- //
- // When paginating, all other parameters provided to `ListCustomClass` must
- // match the call that provided the page token.
- string page_token = 3;
-}
-
-// Message returned to the client by the `ListCustomClasses` method.
-message ListCustomClassesResponse {
- // The custom classes.
- repeated CustomClass custom_classes = 1;
-
- // A token, which can be sent as `page_token` to retrieve the next page.
- // If this field is omitted, there are no subsequent pages.
- string next_page_token = 2;
-}
-
-// Message sent by the client for the `DeleteCustomClass` method.
-message DeleteCustomClassRequest {
- // Required. The name of the custom class to delete.
- // Format:
- // {api_version}/projects/{project}/locations/{location}/customClasses/{custom_class}
- string name = 1 [
- (google.api.field_behavior) = REQUIRED,
- (google.api.resource_reference) = {
- type: "speech.googleapis.com/CustomClass"
- }
- ];
-}
diff --git a/packages/google-cloud-python-speech/google/cloud/speech_v1p1beta1/proto/resource.proto b/packages/google-cloud-python-speech/google/cloud/speech_v1p1beta1/proto/resource.proto
deleted file mode 100644
index 74cee0647d3f..000000000000
--- a/packages/google-cloud-python-speech/google/cloud/speech_v1p1beta1/proto/resource.proto
+++ /dev/null
@@ -1,132 +0,0 @@
-// Copyright 2021 Google LLC
-//
-// Licensed under the Apache License, Version 2.0 (the "License");
-// you may not use this file except in compliance with the License.
-// You may obtain a copy of the License at
-//
-// http://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.
-
-syntax = "proto3";
-
-package google.cloud.speech.v1p1beta1;
-
-import "google/api/resource.proto";
-import "google/api/annotations.proto";
-
-option cc_enable_arenas = true;
-option go_package = "google.golang.org/genproto/googleapis/cloud/speech/v1p1beta1;speech";
-option java_multiple_files = true;
-option java_outer_classname = "SpeechResourceProto";
-option java_package = "com.google.cloud.speech.v1p1beta1";
-option objc_class_prefix = "GCS";
-
-// A set of words or phrases that represents a common concept likely to appear
-// in your audio, for example a list of passenger ship names. CustomClass items
-// can be substituted into placeholders that you set in PhraseSet phrases.
-message CustomClass {
- option (google.api.resource) = {
- type: "speech.googleapis.com/CustomClass"
- pattern: "projects/{project}/locations/{location}/customClasses/{custom_class}"
- };
-
- // An item of the class.
- message ClassItem {
- // The class item's value.
- string value = 1;
- }
-
- // The resource name of the custom class.
- string name = 1;
-
- // If this custom class is a resource, the custom_class_id is the resource id
- // of the CustomClass. Case sensitive.
- string custom_class_id = 2;
-
- // A collection of class items.
- repeated ClassItem items = 3;
-}
-
-// Provides "hints" to the speech recognizer to favor specific words and phrases
-// in the results.
-message PhraseSet {
- option (google.api.resource) = {
- type: "speech.googleapis.com/PhraseSet"
- pattern: "projects/{project}/locations/{location}/phraseSets/{phrase_set}"
- };
-
- // A phrases containing words and phrase "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 include pre-built or custom classes containing groups
- // of words that represent common concepts that occur in natural language. For
- // example, rather than providing a phrase hint for every month of the
- // year (e.g. "i was born in january", "i was born in febuary", ...), use the
- // pre-built `$MONTH` class improves the likelihood of correctly transcribing
- // audio that includes months (e.g. "i was born in $month").
- // To refer to pre-built classes, use the class' symbol prepended with `$`
- // e.g. `$MONTH`. To refer to custom classes that were defined inline in the
- // request, set the class's `custom_class_id` to a string unique to all class
- // resources and inline classes. Then use the class' id wrapped in $`{...}`
- // e.g. "${my-months}". To refer to custom classes resources, use the class'
- // id wrapped in `${}` (e.g. `${my-months}`).
- message Phrase {
- // The phrase itself.
- string value = 1;
-
- // Hint Boost. Overrides the boost set at the phrase set level.
- // 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. Speech recognition
- // will skip PhraseSets with a boost value of 0.
- float boost = 2;
- }
-
- // The resource name of the phrase set.
- string name = 1;
-
- // A list of word and phrases.
- repeated Phrase phrases = 2;
-
- // 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 (exclusive) and 20. We recommend using a binary search
- // approach to finding the optimal value for your use case. Speech recognition
- // will skip PhraseSets with a boost value of 0.
- float boost = 4;
-}
-
-// Speech adaptation configuration.
-message SpeechAdaptation {
- // A collection of phrase sets. To specify the hints inline, leave the
- // phrase set's `name` blank and fill in the rest of its fields. Any
- // phrase set can use any custom class.
- repeated PhraseSet phrase_sets = 1;
-
- // A collection of phrase set resource names to use.
- repeated string phrase_set_references = 2;
-
- // A collection of custom classes. To specify the classes inline, leave the
- // class' `name` blank and fill in the rest of its fields, giving it a unique
- // `custom_class_id`. Refer to the inline defined class in phrase hints by its
- // `custom_class_id`.
- repeated CustomClass custom_classes = 3;
-}