google-cloud-speech 0.29.0 → 0.30.0
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
- checksums.yaml +4 -4
- data/.yardopts +2 -1
- data/LICENSE +1 -1
- data/README.md +69 -43
- data/lib/google/cloud/speech.rb +94 -252
- data/lib/google/cloud/speech/v1.rb +11 -1
- data/lib/google/cloud/speech/v1/cloud_speech_services_pb.rb +1 -1
- data/lib/google/cloud/speech/{version.rb → v1/credentials.rb} +12 -2
- data/lib/google/cloud/speech/v1/doc/google/cloud/speech/v1/cloud_speech.rb +2 -2
- data/lib/google/cloud/speech/v1/doc/google/longrunning/operations.rb +92 -0
- data/lib/google/cloud/speech/v1/doc/google/protobuf/any.rb +1 -1
- data/lib/google/cloud/speech/v1/doc/google/protobuf/duration.rb +1 -1
- data/lib/google/cloud/speech/v1/doc/google/rpc/status.rb +1 -1
- data/lib/google/cloud/speech/v1/doc/overview.rb +1 -1
- data/lib/google/cloud/speech/v1/helpers.rb +93 -0
- data/lib/google/cloud/speech/v1/speech_client.rb +26 -49
- data/lib/google/cloud/speech/v1/speech_client_config.json +5 -5
- data/lib/google/cloud/speech/v1/stream.rb +614 -0
- data/lib/google/cloud/speech/v1p1beta1.rb +126 -0
- data/lib/google/cloud/speech/v1p1beta1/cloud_speech_pb.rb +175 -0
- data/lib/google/cloud/speech/v1p1beta1/cloud_speech_services_pb.rb +54 -0
- data/lib/google/cloud/speech/v1p1beta1/credentials.rb +32 -0
- data/lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb +625 -0
- data/lib/google/cloud/speech/v1p1beta1/doc/google/longrunning/operations.rb +92 -0
- data/lib/google/cloud/speech/v1p1beta1/doc/google/protobuf/any.rb +124 -0
- data/lib/google/cloud/speech/v1p1beta1/doc/google/protobuf/duration.rb +90 -0
- data/lib/google/cloud/speech/v1p1beta1/doc/google/rpc/status.rb +83 -0
- data/lib/google/cloud/speech/v1p1beta1/doc/overview.rb +73 -0
- data/lib/google/cloud/speech/v1p1beta1/helpers.rb +93 -0
- data/lib/google/cloud/speech/v1p1beta1/speech_client.rb +322 -0
- data/lib/google/cloud/speech/v1p1beta1/speech_client_config.json +41 -0
- data/lib/google/cloud/speech/v1p1beta1/stream.rb +614 -0
- metadata +29 -120
- data/lib/google-cloud-speech.rb +0 -142
- data/lib/google/cloud/speech/audio.rb +0 -330
- data/lib/google/cloud/speech/convert.rb +0 -46
- data/lib/google/cloud/speech/credentials.rb +0 -57
- data/lib/google/cloud/speech/operation.rb +0 -262
- data/lib/google/cloud/speech/project.rb +0 -651
- data/lib/google/cloud/speech/result.rb +0 -240
- data/lib/google/cloud/speech/service.rb +0 -121
- data/lib/google/cloud/speech/stream.rb +0 -564
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# Copyright 2018 Google LLC
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# https://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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require "google/cloud/speech/v1p1beta1/speech_client"
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require "google/cloud/speech/v1p1beta1/helpers"
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module Google
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module Cloud
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# rubocop:disable LineLength
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##
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# # Ruby Client for Cloud Speech API ([Alpha](https://github.com/GoogleCloudPlatform/google-cloud-ruby#versioning))
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#
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# [Cloud Speech API][Product Documentation]:
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# Converts audio to text by applying powerful neural network models.
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# - [Product Documentation][]
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#
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# ## Quick Start
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# In order to use this library, you first need to go through the following
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# steps:
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#
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# 1. [Select or create a Cloud Platform project.](https://console.cloud.google.com/project)
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# 2. [Enable billing for your project.](https://cloud.google.com/billing/docs/how-to/modify-project#enable_billing_for_a_project)
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# 3. [Enable the Cloud Speech API.](https://console.cloud.google.com/apis/api/speech)
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# 4. [Setup Authentication.](https://googlecloudplatform.github.io/google-cloud-ruby/#/docs/google-cloud/master/guides/authentication)
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#
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# ### Preview
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# #### SpeechClient
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# ```rb
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# require "google/cloud/speech/v1p1beta1"
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#
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# speech_client = Google::Cloud::Speech::V1p1beta1.new
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# language_code = "en-US"
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# sample_rate_hertz = 44100
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# encoding = :FLAC
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# config = {
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# language_code: language_code,
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# sample_rate_hertz: sample_rate_hertz,
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# encoding: encoding
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# }
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# uri = "gs://gapic-toolkit/hello.flac"
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# audio = { uri: uri }
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# response = speech_client.recognize(config, audio)
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# ```
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#
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# ### Next Steps
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# - Read the [Cloud Speech API Product documentation][Product Documentation]
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# to learn more about the product and see How-to Guides.
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# - View this [repository's main README](https://github.com/GoogleCloudPlatform/google-cloud-ruby/blob/master/README.md)
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# to see the full list of Cloud APIs that we cover.
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#
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# [Product Documentation]: https://cloud.google.com/speech
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#
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#
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module Speech
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module V1p1beta1
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# rubocop:enable LineLength
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##
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# Service that implements Google Cloud Speech API.
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#
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# @param credentials [Google::Auth::Credentials, String, Hash, GRPC::Core::Channel, GRPC::Core::ChannelCredentials, Proc]
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# Provides the means for authenticating requests made by the client. This parameter can
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# be many types.
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# A `Google::Auth::Credentials` uses a the properties of its represented keyfile for
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# authenticating requests made by this client.
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# A `String` will be treated as the path to the keyfile to be used for the construction of
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# credentials for this client.
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# A `Hash` will be treated as the contents of a keyfile to be used for the construction of
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# credentials for this client.
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# A `GRPC::Core::Channel` will be used to make calls through.
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# A `GRPC::Core::ChannelCredentials` for the setting up the RPC client. The channel credentials
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# should already be composed with a `GRPC::Core::CallCredentials` object.
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# A `Proc` will be used as an updater_proc for the Grpc channel. The proc transforms the
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# metadata for requests, generally, to give OAuth credentials.
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# @param scopes [Array<String>]
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# The OAuth scopes for this service. This parameter is ignored if
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# an updater_proc is supplied.
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# @param client_config [Hash]
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# A Hash for call options for each method. See
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# Google::Gax#construct_settings for the structure of
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# this data. Falls back to the default config if not specified
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# or the specified config is missing data points.
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# @param timeout [Numeric]
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# The default timeout, in seconds, for calls made through this client.
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# @param metadata [Hash]
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# Default metadata to be sent with each request. This can be overridden on a per call basis.
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# @param exception_transformer [Proc]
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# An optional proc that intercepts any exceptions raised during an API call to inject
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# custom error handling.
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def self.new \
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credentials: nil,
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scopes: nil,
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client_config: nil,
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timeout: nil,
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metadata: nil,
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exception_transformer: nil,
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lib_name: nil,
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lib_version: nil
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kwargs = {
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credentials: credentials,
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scopes: scopes,
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client_config: client_config,
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timeout: timeout,
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metadata: metadata,
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exception_transformer: exception_transformer,
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lib_name: lib_name,
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lib_version: lib_version
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}.select { |_, v| v != nil }
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Google::Cloud::Speech::V1p1beta1::SpeechClient.new(**kwargs)
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end
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end
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end
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end
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end
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# Generated by the protocol buffer compiler. DO NOT EDIT!
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# source: google/cloud/speech/v1p1beta1/cloud_speech.proto
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require 'google/protobuf'
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require 'google/api/annotations_pb'
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require 'google/longrunning/operations_pb'
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require 'google/protobuf/any_pb'
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require 'google/protobuf/duration_pb'
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require 'google/protobuf/timestamp_pb'
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require 'google/rpc/status_pb'
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Google::Protobuf::DescriptorPool.generated_pool.build do
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add_message "google.cloud.speech.v1p1beta1.RecognizeRequest" do
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optional :config, :message, 1, "google.cloud.speech.v1p1beta1.RecognitionConfig"
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optional :audio, :message, 2, "google.cloud.speech.v1p1beta1.RecognitionAudio"
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end
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add_message "google.cloud.speech.v1p1beta1.LongRunningRecognizeRequest" do
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optional :config, :message, 1, "google.cloud.speech.v1p1beta1.RecognitionConfig"
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optional :audio, :message, 2, "google.cloud.speech.v1p1beta1.RecognitionAudio"
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end
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add_message "google.cloud.speech.v1p1beta1.StreamingRecognizeRequest" do
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oneof :streaming_request do
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optional :streaming_config, :message, 1, "google.cloud.speech.v1p1beta1.StreamingRecognitionConfig"
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optional :audio_content, :bytes, 2
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end
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end
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add_message "google.cloud.speech.v1p1beta1.StreamingRecognitionConfig" do
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optional :config, :message, 1, "google.cloud.speech.v1p1beta1.RecognitionConfig"
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optional :single_utterance, :bool, 2
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optional :interim_results, :bool, 3
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end
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add_message "google.cloud.speech.v1p1beta1.RecognitionConfig" do
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optional :encoding, :enum, 1, "google.cloud.speech.v1p1beta1.RecognitionConfig.AudioEncoding"
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optional :sample_rate_hertz, :int32, 2
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optional :language_code, :string, 3
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optional :max_alternatives, :int32, 4
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optional :profanity_filter, :bool, 5
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repeated :speech_contexts, :message, 6, "google.cloud.speech.v1p1beta1.SpeechContext"
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optional :enable_word_time_offsets, :bool, 8
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optional :enable_automatic_punctuation, :bool, 11
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optional :metadata, :message, 9, "google.cloud.speech.v1p1beta1.RecognitionMetadata"
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optional :model, :string, 13
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optional :use_enhanced, :bool, 14
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end
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add_enum "google.cloud.speech.v1p1beta1.RecognitionConfig.AudioEncoding" do
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value :ENCODING_UNSPECIFIED, 0
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value :LINEAR16, 1
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value :FLAC, 2
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value :MULAW, 3
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value :AMR, 4
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value :AMR_WB, 5
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value :OGG_OPUS, 6
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value :SPEEX_WITH_HEADER_BYTE, 7
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end
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add_message "google.cloud.speech.v1p1beta1.RecognitionMetadata" do
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optional :interaction_type, :enum, 1, "google.cloud.speech.v1p1beta1.RecognitionMetadata.InteractionType"
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optional :industry_naics_code_of_audio, :uint32, 3
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optional :microphone_distance, :enum, 4, "google.cloud.speech.v1p1beta1.RecognitionMetadata.MicrophoneDistance"
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optional :original_media_type, :enum, 5, "google.cloud.speech.v1p1beta1.RecognitionMetadata.OriginalMediaType"
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optional :recording_device_type, :enum, 6, "google.cloud.speech.v1p1beta1.RecognitionMetadata.RecordingDeviceType"
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optional :recording_device_name, :string, 7
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optional :original_mime_type, :string, 8
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optional :obfuscated_id, :int64, 9
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optional :audio_topic, :string, 10
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end
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add_enum "google.cloud.speech.v1p1beta1.RecognitionMetadata.InteractionType" do
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value :INTERACTION_TYPE_UNSPECIFIED, 0
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value :DISCUSSION, 1
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value :PRESENTATION, 2
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value :PHONE_CALL, 3
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value :VOICEMAIL, 4
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value :PROFESSIONALLY_PRODUCED, 5
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value :VOICE_SEARCH, 6
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value :VOICE_COMMAND, 7
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value :DICTATION, 8
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end
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add_enum "google.cloud.speech.v1p1beta1.RecognitionMetadata.MicrophoneDistance" do
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value :MICROPHONE_DISTANCE_UNSPECIFIED, 0
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value :NEARFIELD, 1
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value :MIDFIELD, 2
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value :FARFIELD, 3
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end
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add_enum "google.cloud.speech.v1p1beta1.RecognitionMetadata.OriginalMediaType" do
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value :ORIGINAL_MEDIA_TYPE_UNSPECIFIED, 0
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value :AUDIO, 1
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value :VIDEO, 2
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end
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add_enum "google.cloud.speech.v1p1beta1.RecognitionMetadata.RecordingDeviceType" do
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value :RECORDING_DEVICE_TYPE_UNSPECIFIED, 0
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value :SMARTPHONE, 1
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value :PC, 2
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value :PHONE_LINE, 3
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value :VEHICLE, 4
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value :OTHER_OUTDOOR_DEVICE, 5
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value :OTHER_INDOOR_DEVICE, 6
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end
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add_message "google.cloud.speech.v1p1beta1.SpeechContext" do
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repeated :phrases, :string, 1
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end
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add_message "google.cloud.speech.v1p1beta1.RecognitionAudio" do
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oneof :audio_source do
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optional :content, :bytes, 1
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optional :uri, :string, 2
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end
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end
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add_message "google.cloud.speech.v1p1beta1.RecognizeResponse" do
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repeated :results, :message, 2, "google.cloud.speech.v1p1beta1.SpeechRecognitionResult"
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end
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add_message "google.cloud.speech.v1p1beta1.LongRunningRecognizeResponse" do
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repeated :results, :message, 2, "google.cloud.speech.v1p1beta1.SpeechRecognitionResult"
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end
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add_message "google.cloud.speech.v1p1beta1.LongRunningRecognizeMetadata" do
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optional :progress_percent, :int32, 1
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optional :start_time, :message, 2, "google.protobuf.Timestamp"
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optional :last_update_time, :message, 3, "google.protobuf.Timestamp"
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end
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add_message "google.cloud.speech.v1p1beta1.StreamingRecognizeResponse" do
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optional :error, :message, 1, "google.rpc.Status"
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repeated :results, :message, 2, "google.cloud.speech.v1p1beta1.StreamingRecognitionResult"
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optional :speech_event_type, :enum, 4, "google.cloud.speech.v1p1beta1.StreamingRecognizeResponse.SpeechEventType"
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end
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add_enum "google.cloud.speech.v1p1beta1.StreamingRecognizeResponse.SpeechEventType" do
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value :SPEECH_EVENT_UNSPECIFIED, 0
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value :END_OF_SINGLE_UTTERANCE, 1
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end
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add_message "google.cloud.speech.v1p1beta1.StreamingRecognitionResult" do
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repeated :alternatives, :message, 1, "google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative"
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optional :is_final, :bool, 2
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optional :stability, :float, 3
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end
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add_message "google.cloud.speech.v1p1beta1.SpeechRecognitionResult" do
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repeated :alternatives, :message, 1, "google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative"
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end
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add_message "google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative" do
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optional :transcript, :string, 1
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optional :confidence, :float, 2
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repeated :words, :message, 3, "google.cloud.speech.v1p1beta1.WordInfo"
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end
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add_message "google.cloud.speech.v1p1beta1.WordInfo" do
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optional :start_time, :message, 1, "google.protobuf.Duration"
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optional :end_time, :message, 2, "google.protobuf.Duration"
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optional :word, :string, 3
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end
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end
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module Google
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module Cloud
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module Speech
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module V1p1beta1
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RecognizeRequest = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognizeRequest").msgclass
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LongRunningRecognizeRequest = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.LongRunningRecognizeRequest").msgclass
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StreamingRecognizeRequest = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.StreamingRecognizeRequest").msgclass
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StreamingRecognitionConfig = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.StreamingRecognitionConfig").msgclass
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RecognitionConfig = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionConfig").msgclass
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RecognitionConfig::AudioEncoding = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionConfig.AudioEncoding").enummodule
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RecognitionMetadata = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionMetadata").msgclass
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RecognitionMetadata::InteractionType = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionMetadata.InteractionType").enummodule
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RecognitionMetadata::MicrophoneDistance = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionMetadata.MicrophoneDistance").enummodule
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RecognitionMetadata::OriginalMediaType = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionMetadata.OriginalMediaType").enummodule
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RecognitionMetadata::RecordingDeviceType = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionMetadata.RecordingDeviceType").enummodule
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SpeechContext = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.SpeechContext").msgclass
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RecognitionAudio = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionAudio").msgclass
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RecognizeResponse = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognizeResponse").msgclass
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LongRunningRecognizeResponse = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.LongRunningRecognizeResponse").msgclass
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LongRunningRecognizeMetadata = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.LongRunningRecognizeMetadata").msgclass
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StreamingRecognizeResponse = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.StreamingRecognizeResponse").msgclass
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StreamingRecognizeResponse::SpeechEventType = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.StreamingRecognizeResponse.SpeechEventType").enummodule
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StreamingRecognitionResult = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.StreamingRecognitionResult").msgclass
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SpeechRecognitionResult = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.SpeechRecognitionResult").msgclass
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SpeechRecognitionAlternative = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative").msgclass
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WordInfo = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.WordInfo").msgclass
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end
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end
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end
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end
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# Generated by the protocol buffer compiler. DO NOT EDIT!
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# Source: google/cloud/speech/v1p1beta1/cloud_speech.proto for package 'google.cloud.speech.v1p1beta1'
|
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# Original file comments:
|
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# Copyright 2018 Google Inc.
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
|
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+
#
|
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+
# http://www.apache.org/licenses/LICENSE-2.0
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+
#
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# Unless required by applicable law or agreed to in writing, software
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+
# distributed under the License is distributed on an "AS IS" BASIS,
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+
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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+
# See the License for the specific language governing permissions and
|
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+
# limitations under the License.
|
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+
#
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+
|
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require 'grpc'
|
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+
require 'google/cloud/speech/v1p1beta1/cloud_speech_pb'
|
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|
+
|
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module Google
|
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module Cloud
|
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module Speech
|
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module V1p1beta1
|
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|
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module Speech
|
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# Service that implements Google Cloud Speech API.
|
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class Service
|
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+
|
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include GRPC::GenericService
|
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+
|
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self.marshal_class_method = :encode
|
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+
self.unmarshal_class_method = :decode
|
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|
+
self.service_name = 'google.cloud.speech.v1p1beta1.Speech'
|
35
|
+
|
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+
# Performs synchronous speech recognition: receive results after all audio
|
37
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+
# has been sent and processed.
|
38
|
+
rpc :Recognize, RecognizeRequest, RecognizeResponse
|
39
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+
# Performs asynchronous speech recognition: receive results via the
|
40
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+
# google.longrunning.Operations interface. Returns either an
|
41
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+
# `Operation.error` or an `Operation.response` which contains
|
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# a `LongRunningRecognizeResponse` message.
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rpc :LongRunningRecognize, LongRunningRecognizeRequest, Google::Longrunning::Operation
|
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+
# Performs bidirectional streaming speech recognition: receive results while
|
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+
# sending audio. This method is only available via the gRPC API (not REST).
|
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+
rpc :StreamingRecognize, stream(StreamingRecognizeRequest), stream(StreamingRecognizeResponse)
|
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|
+
end
|
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|
+
|
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+
Stub = Service.rpc_stub_class
|
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|
+
end
|
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|
+
end
|
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+
end
|
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end
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+
end
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|
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# Copyright 2018 Google LLC
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+
#
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+
# Licensed under the Apache License, Version 2.0 (the "License");
|
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+
# you may not use this file except in compliance with the License.
|
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+
# You may obtain a copy of the License at
|
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|
+
#
|
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|
+
# https://www.apache.org/licenses/LICENSE-2.0
|
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|
+
#
|
9
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+
# Unless required by applicable law or agreed to in writing, software
|
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+
# distributed under the License is distributed on an "AS IS" BASIS,
|
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+
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
|
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|
+
# See the License for the specific language governing permissions and
|
13
|
+
# limitations under the License.
|
14
|
+
|
15
|
+
require "googleauth"
|
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|
+
|
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|
+
module Google
|
18
|
+
module Cloud
|
19
|
+
module Speech
|
20
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+
module V1p1beta1
|
21
|
+
class Credentials < Google::Auth::Credentials
|
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+
SCOPE = [
|
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+
"https://www.googleapis.com/auth/cloud-platform"
|
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+
].freeze
|
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+
PATH_ENV_VARS = %w(SPEECH_KEYFILE GOOGLE_CLOUD_KEYFILE GCLOUD_KEYFILE)
|
26
|
+
JSON_ENV_VARS = %w(SPEECH_KEYFILE_JSON GOOGLE_CLOUD_KEYFILE_JSON GCLOUD_KEYFILE_JSON)
|
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+
DEFAULT_PATHS = ["~/.config/gcloud/application_default_credentials.json"]
|
28
|
+
end
|
29
|
+
end
|
30
|
+
end
|
31
|
+
end
|
32
|
+
end
|
@@ -0,0 +1,625 @@
|
|
1
|
+
# Copyright 2018 Google LLC
|
2
|
+
#
|
3
|
+
# Licensed under the Apache License, Version 2.0 (the "License");
|
4
|
+
# you may not use this file except in compliance with the License.
|
5
|
+
# You may obtain a copy of the License at
|
6
|
+
#
|
7
|
+
# https://www.apache.org/licenses/LICENSE-2.0
|
8
|
+
#
|
9
|
+
# Unless required by applicable law or agreed to in writing, software
|
10
|
+
# distributed under the License is distributed on an "AS IS" BASIS,
|
11
|
+
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
|
12
|
+
# See the License for the specific language governing permissions and
|
13
|
+
# limitations under the License.
|
14
|
+
|
15
|
+
module Google
|
16
|
+
module Cloud
|
17
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+
module Speech
|
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+
##
|
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+
# # Cloud Speech API Contents
|
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+
#
|
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+
# | Class | Description |
|
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+
# | ----- | ----------- |
|
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+
# | [SpeechClient][] | Service that implements Google Cloud Speech API. |
|
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+
# | [Data Types][] | Data types for Google::Cloud::Speech::V1p1beta1 |
|
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+
#
|
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+
# [SpeechClient]: https://googlecloudplatform.github.io/google-cloud-ruby/#/docs/google-cloud-speech/latest/google/cloud/speech/v1p1beta1/speechclient
|
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|
+
# [Data Types]: https://googlecloudplatform.github.io/google-cloud-ruby/#/docs/google-cloud-speech/latest/google/cloud/speech/v1p1beta1/datatypes
|
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|
+
#
|
29
|
+
module V1p1beta1
|
30
|
+
# The top-level message sent by the client for the +Recognize+ method.
|
31
|
+
# @!attribute [rw] config
|
32
|
+
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionConfig]
|
33
|
+
# *Required* Provides information to the recognizer that specifies how to
|
34
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+
# process the request.
|
35
|
+
# @!attribute [rw] audio
|
36
|
+
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionAudio]
|
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|
+
# *Required* The audio data to be recognized.
|
38
|
+
class RecognizeRequest; end
|
39
|
+
|
40
|
+
# The top-level message sent by the client for the +LongRunningRecognize+
|
41
|
+
# method.
|
42
|
+
# @!attribute [rw] config
|
43
|
+
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionConfig]
|
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|
+
# *Required* Provides information to the recognizer that specifies how to
|
45
|
+
# process the request.
|
46
|
+
# @!attribute [rw] audio
|
47
|
+
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionAudio]
|
48
|
+
# *Required* The audio data to be recognized.
|
49
|
+
class LongRunningRecognizeRequest; end
|
50
|
+
|
51
|
+
# The top-level message sent by the client for the +StreamingRecognize+ method.
|
52
|
+
# Multiple +StreamingRecognizeRequest+ messages are sent. The first message
|
53
|
+
# must contain a +streaming_config+ message and must not contain +audio+ data.
|
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|
+
# All subsequent messages must contain +audio+ data and must not contain a
|
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|
+
# +streaming_config+ message.
|
56
|
+
# @!attribute [rw] streaming_config
|
57
|
+
# @return [Google::Cloud::Speech::V1p1beta1::StreamingRecognitionConfig]
|
58
|
+
# Provides information to the recognizer that specifies how to process the
|
59
|
+
# request. The first +StreamingRecognizeRequest+ message must contain a
|
60
|
+
# +streaming_config+ message.
|
61
|
+
# @!attribute [rw] audio_content
|
62
|
+
# @return [String]
|
63
|
+
# The audio data to be recognized. Sequential chunks of audio data are sent
|
64
|
+
# in sequential +StreamingRecognizeRequest+ messages. The first
|
65
|
+
# +StreamingRecognizeRequest+ message must not contain +audio_content+ data
|
66
|
+
# and all subsequent +StreamingRecognizeRequest+ messages must contain
|
67
|
+
# +audio_content+ data. The audio bytes must be encoded as specified in
|
68
|
+
# +RecognitionConfig+. Note: as with all bytes fields, protobuffers use a
|
69
|
+
# pure binary representation (not base64). See
|
70
|
+
# [audio limits](https://cloud.google.com/speech/limits#content).
|
71
|
+
class StreamingRecognizeRequest; end
|
72
|
+
|
73
|
+
# Provides information to the recognizer that specifies how to process the
|
74
|
+
# request.
|
75
|
+
# @!attribute [rw] config
|
76
|
+
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionConfig]
|
77
|
+
# *Required* Provides information to the recognizer that specifies how to
|
78
|
+
# process the request.
|
79
|
+
# @!attribute [rw] single_utterance
|
80
|
+
# @return [true, false]
|
81
|
+
# *Optional* If +false+ or omitted, the recognizer will perform continuous
|
82
|
+
# recognition (continuing to wait for and process audio even if the user
|
83
|
+
# pauses speaking) until the client closes the input stream (gRPC API) or
|
84
|
+
# until the maximum time limit has been reached. May return multiple
|
85
|
+
# +StreamingRecognitionResult+s with the +is_final+ flag set to +true+.
|
86
|
+
#
|
87
|
+
# If +true+, the recognizer will detect a single spoken utterance. When it
|
88
|
+
# detects that the user has paused or stopped speaking, it will return an
|
89
|
+
# +END_OF_SINGLE_UTTERANCE+ event and cease recognition. It will return no
|
90
|
+
# more than one +StreamingRecognitionResult+ with the +is_final+ flag set to
|
91
|
+
# +true+.
|
92
|
+
# @!attribute [rw] interim_results
|
93
|
+
# @return [true, false]
|
94
|
+
# *Optional* If +true+, interim results (tentative hypotheses) may be
|
95
|
+
# returned as they become available (these interim results are indicated with
|
96
|
+
# the +is_final=false+ flag).
|
97
|
+
# If +false+ or omitted, only +is_final=true+ result(s) are returned.
|
98
|
+
class StreamingRecognitionConfig; end
|
99
|
+
|
100
|
+
# Provides information to the recognizer that specifies how to process the
|
101
|
+
# request.
|
102
|
+
# @!attribute [rw] encoding
|
103
|
+
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionConfig::AudioEncoding]
|
104
|
+
# Encoding of audio data sent in all +RecognitionAudio+ messages.
|
105
|
+
# This field is optional for +FLAC+ and +WAV+ audio files and required
|
106
|
+
# for all other audio formats. For details, see {Google::Cloud::Speech::V1p1beta1::RecognitionConfig::AudioEncoding AudioEncoding}.
|
107
|
+
# @!attribute [rw] sample_rate_hertz
|
108
|
+
# @return [Integer]
|
109
|
+
# Sample rate in Hertz of the audio data sent in all
|
110
|
+
# +RecognitionAudio+ messages. Valid values are: 8000-48000.
|
111
|
+
# 16000 is optimal. For best results, set the sampling rate of the audio
|
112
|
+
# source to 16000 Hz. If that's not possible, use the native sample rate of
|
113
|
+
# the audio source (instead of re-sampling).
|
114
|
+
# This field is optional for +FLAC+ and +WAV+ audio files and required
|
115
|
+
# for all other audio formats. For details, see {Google::Cloud::Speech::V1p1beta1::RecognitionConfig::AudioEncoding AudioEncoding}.
|
116
|
+
# @!attribute [rw] language_code
|
117
|
+
# @return [String]
|
118
|
+
# *Required* The language of the supplied audio as a
|
119
|
+
# [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag.
|
120
|
+
# Example: "en-US".
|
121
|
+
# See [Language Support](https://cloud.google.com/speech/docs/languages)
|
122
|
+
# for a list of the currently supported language codes.
|
123
|
+
# @!attribute [rw] max_alternatives
|
124
|
+
# @return [Integer]
|
125
|
+
# *Optional* Maximum number of recognition hypotheses to be returned.
|
126
|
+
# Specifically, the maximum number of +SpeechRecognitionAlternative+ messages
|
127
|
+
# within each +SpeechRecognitionResult+.
|
128
|
+
# The server may return fewer than +max_alternatives+.
|
129
|
+
# Valid values are +0+-+30+. A value of +0+ or +1+ will return a maximum of
|
130
|
+
# one. If omitted, will return a maximum of one.
|
131
|
+
# @!attribute [rw] profanity_filter
|
132
|
+
# @return [true, false]
|
133
|
+
# *Optional* If set to +true+, the server will attempt to filter out
|
134
|
+
# profanities, replacing all but the initial character in each filtered word
|
135
|
+
# with asterisks, e.g. "f***". If set to +false+ or omitted, profanities
|
136
|
+
# won't be filtered out.
|
137
|
+
# @!attribute [rw] speech_contexts
|
138
|
+
# @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechContext>]
|
139
|
+
# *Optional* A means to provide context to assist the speech recognition.
|
140
|
+
# @!attribute [rw] enable_word_time_offsets
|
141
|
+
# @return [true, false]
|
142
|
+
# *Optional* If +true+, the top result includes a list of words and
|
143
|
+
# the start and end time offsets (timestamps) for those words. If
|
144
|
+
# +false+, no word-level time offset information is returned. The default is
|
145
|
+
# +false+.
|
146
|
+
# @!attribute [rw] enable_automatic_punctuation
|
147
|
+
# @return [true, false]
|
148
|
+
# *Optional* If 'true', adds punctuation to recognition result hypotheses.
|
149
|
+
# This feature is only available in select languages. Setting this for
|
150
|
+
# requests in other languages has no effect at all.
|
151
|
+
# The default 'false' value does not add punctuation to result hypotheses.
|
152
|
+
# NOTE: "This is currently offered as an experimental service, complimentary
|
153
|
+
# to all users. In the future this may be exclusively available as a
|
154
|
+
# premium feature."
|
155
|
+
# @!attribute [rw] metadata
|
156
|
+
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata]
|
157
|
+
# *Optional* Metadata regarding this request.
|
158
|
+
# @!attribute [rw] model
|
159
|
+
# @return [String]
|
160
|
+
# *Optional* Which model to select for the given request. Select the model
|
161
|
+
# best suited to your domain to get best results. If a model is not
|
162
|
+
# explicitly specified, then we auto-select a model based on the parameters
|
163
|
+
# in the RecognitionConfig.
|
164
|
+
# <table>
|
165
|
+
# <tr>
|
166
|
+
# <td><b>Model</b></td>
|
167
|
+
# <td><b>Description</b></td>
|
168
|
+
# </tr>
|
169
|
+
# <tr>
|
170
|
+
# <td><code>command_and_search</code></td>
|
171
|
+
# <td>Best for short queries such as voice commands or voice search.</td>
|
172
|
+
# </tr>
|
173
|
+
# <tr>
|
174
|
+
# <td><code>phone_call</code></td>
|
175
|
+
# <td>Best for audio that originated from a phone call (typically
|
176
|
+
# recorded at an 8khz sampling rate).</td>
|
177
|
+
# </tr>
|
178
|
+
# <tr>
|
179
|
+
# <td><code>video</code></td>
|
180
|
+
# <td>Best for audio that originated from from video or includes multiple
|
181
|
+
# speakers. Ideally the audio is recorded at a 16khz or greater
|
182
|
+
# sampling rate. This is a premium model that costs more than the
|
183
|
+
# standard rate.</td>
|
184
|
+
# </tr>
|
185
|
+
# <tr>
|
186
|
+
# <td><code>default</code></td>
|
187
|
+
# <td>Best for audio that is not one of the specific audio models.
|
188
|
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# For example, long-form audio. Ideally the audio is high-fidelity,
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# recorded at a 16khz or greater sampling rate.</td>
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# </tr>
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# </table>
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# @!attribute [rw] use_enhanced
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# @return [true, false]
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# *Optional* Set to true to use an enhanced model for speech recognition.
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# You must also set the +model+ field to a valid, enhanced model. If
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# +use_enhanced+ is set to true and the +model+ field is not set, then
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# +use_enhanced+ is ignored. If +use_enhanced+ is true and an enhanced
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# version of the specified model does not exist, then the speech is
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# recognized using the standard version of the specified model.
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#
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# Enhanced speech models require that you opt-in to the audio logging using
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# instructions in the [alpha documentation](https://cloud.google.com/speech/data-sharing). If you set
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# +use_enhanced+ to true and you have not enabled audio logging, then you
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# will receive an error.
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class RecognitionConfig
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# The encoding of the audio data sent in the request.
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#
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# All encodings support only 1 channel (mono) audio.
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#
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# For best results, the audio source should be captured and transmitted using
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# a lossless encoding (+FLAC+ or +LINEAR16+). The accuracy of the speech
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# recognition can be reduced if lossy codecs are used to capture or transmit
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# audio, particularly if background noise is present. Lossy codecs include
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# +MULAW+, +AMR+, +AMR_WB+, +OGG_OPUS+, and +SPEEX_WITH_HEADER_BYTE+.
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+
#
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+
# The +FLAC+ and +WAV+ audio file formats include a header that describes the
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# included audio content. You can request recognition for +WAV+ files that
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+
# contain either +LINEAR16+ or +MULAW+ encoded audio.
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+
# If you send +FLAC+ or +WAV+ audio file format in
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# your request, you do not need to specify an +AudioEncoding+; the audio
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+
# encoding format is determined from the file header. If you specify
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# an +AudioEncoding+ when you send send +FLAC+ or +WAV+ audio, the
|
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# encoding configuration must match the encoding described in the audio
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# header; otherwise the request returns an
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# {Google::Rpc::Code::INVALID_ARGUMENT} error code.
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module AudioEncoding
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# Not specified.
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+
ENCODING_UNSPECIFIED = 0
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+
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# Uncompressed 16-bit signed little-endian samples (Linear PCM).
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+
LINEAR16 = 1
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+
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# +FLAC+ (Free Lossless Audio
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+
# Codec) is the recommended encoding because it is
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# lossless--therefore recognition is not compromised--and
|
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# requires only about half the bandwidth of +LINEAR16+. +FLAC+ stream
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# encoding supports 16-bit and 24-bit samples, however, not all fields in
|
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# +STREAMINFO+ are supported.
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+
FLAC = 2
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+
|
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# 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
|
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+
MULAW = 3
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+
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# Adaptive Multi-Rate Narrowband codec. +sample_rate_hertz+ must be 8000.
|
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|
+
AMR = 4
|
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|
+
|
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# Adaptive Multi-Rate Wideband codec. +sample_rate_hertz+ must be 16000.
|
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+
AMR_WB = 5
|
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|
+
|
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|
+
# Opus encoded audio frames in Ogg container
|
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|
+
# ([OggOpus](https://wiki.xiph.org/OggOpus)).
|
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|
+
# +sample_rate_hertz+ must be one of 8000, 12000, 16000, 24000, or 48000.
|
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|
+
OGG_OPUS = 6
|
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|
+
|
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|
+
# Although the use of lossy encodings is not recommended, if a very low
|
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|
+
# bitrate encoding is required, +OGG_OPUS+ is highly preferred over
|
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|
+
# Speex encoding. The [Speex](https://speex.org/) encoding supported by
|
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|
+
# Cloud Speech API has a header byte in each block, as in MIME type
|
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|
+
# +audio/x-speex-with-header-byte+.
|
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|
+
# It is a variant of the RTP Speex encoding defined in
|
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|
+
# [RFC 5574](https://tools.ietf.org/html/rfc5574).
|
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|
+
# The stream is a sequence of blocks, one block per RTP packet. Each block
|
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|
+
# starts with a byte containing the length of the block, in bytes, followed
|
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|
+
# by one or more frames of Speex data, padded to an integral number of
|
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|
+
# bytes (octets) as specified in RFC 5574. In other words, each RTP header
|
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|
+
# is replaced with a single byte containing the block length. Only Speex
|
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|
+
# wideband is supported. +sample_rate_hertz+ must be 16000.
|
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|
+
SPEEX_WITH_HEADER_BYTE = 7
|
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|
+
end
|
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|
+
end
|
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|
+
|
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|
+
# Description of audio data to be recognized.
|
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|
+
# @!attribute [rw] interaction_type
|
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|
+
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata::InteractionType]
|
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|
+
# The use case most closely describing the audio content to be recognized.
|
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|
+
# @!attribute [rw] industry_naics_code_of_audio
|
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|
+
# @return [Integer]
|
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|
+
# The industry vertical to which this speech recognition request most
|
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|
+
# closely applies. This is most indicative of the topics contained
|
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|
+
# in the audio. Use the 6-digit NAICS code to identify the industry
|
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|
+
# vertical - see https://www.naics.com/search/.
|
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|
+
# @!attribute [rw] microphone_distance
|
283
|
+
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata::MicrophoneDistance]
|
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|
+
# The audio type that most closely describes the audio being recognized.
|
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|
+
# @!attribute [rw] original_media_type
|
286
|
+
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata::OriginalMediaType]
|
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|
+
# The original media the speech was recorded on.
|
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|
+
# @!attribute [rw] recording_device_type
|
289
|
+
# @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata::RecordingDeviceType]
|
290
|
+
# The type of device the speech was recorded with.
|
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|
+
# @!attribute [rw] recording_device_name
|
292
|
+
# @return [String]
|
293
|
+
# The device used to make the recording. Examples 'Nexus 5X' or
|
294
|
+
# 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
|
295
|
+
# 'Cardioid Microphone'.
|
296
|
+
# @!attribute [rw] original_mime_type
|
297
|
+
# @return [String]
|
298
|
+
# Mime type of the original audio file. For example +audio/m4a+,
|
299
|
+
# +audio/x-alaw-basic+, +audio/mp3+, +audio/3gpp+.
|
300
|
+
# A list of possible audio mime types is maintained at
|
301
|
+
# http://www.iana.org/assignments/media-types/media-types.xhtml#audio
|
302
|
+
# @!attribute [rw] obfuscated_id
|
303
|
+
# @return [Integer]
|
304
|
+
# Obfuscated (privacy-protected) ID of the user, to identify number of
|
305
|
+
# unique users using the service.
|
306
|
+
# @!attribute [rw] audio_topic
|
307
|
+
# @return [String]
|
308
|
+
# Description of the content. Eg. "Recordings of federal supreme court
|
309
|
+
# hearings from 2012".
|
310
|
+
class RecognitionMetadata
|
311
|
+
# Use case categories that the audio recognition request can be described
|
312
|
+
# by.
|
313
|
+
module InteractionType
|
314
|
+
# Use case is either unknown or is something other than one of the other
|
315
|
+
# values below.
|
316
|
+
INTERACTION_TYPE_UNSPECIFIED = 0
|
317
|
+
|
318
|
+
# Multiple people in a conversation or discussion. For example in a
|
319
|
+
# meeting with two or more people actively participating. Typically
|
320
|
+
# all the primary people speaking would be in the same room (if not,
|
321
|
+
# see PHONE_CALL)
|
322
|
+
DISCUSSION = 1
|
323
|
+
|
324
|
+
# One or more persons lecturing or presenting to others, mostly
|
325
|
+
# uninterrupted.
|
326
|
+
PRESENTATION = 2
|
327
|
+
|
328
|
+
# A phone-call or video-conference in which two or more people, who are
|
329
|
+
# not in the same room, are actively participating.
|
330
|
+
PHONE_CALL = 3
|
331
|
+
|
332
|
+
# A recorded message intended for another person to listen to.
|
333
|
+
VOICEMAIL = 4
|
334
|
+
|
335
|
+
# Professionally produced audio (eg. TV Show, Podcast).
|
336
|
+
PROFESSIONALLY_PRODUCED = 5
|
337
|
+
|
338
|
+
# Transcribe spoken questions and queries into text.
|
339
|
+
VOICE_SEARCH = 6
|
340
|
+
|
341
|
+
# Transcribe voice commands, such as for controlling a device.
|
342
|
+
VOICE_COMMAND = 7
|
343
|
+
|
344
|
+
# Transcribe speech to text to create a written document, such as a
|
345
|
+
# text-message, email or report.
|
346
|
+
DICTATION = 8
|
347
|
+
end
|
348
|
+
|
349
|
+
# Enumerates the types of capture settings describing an audio file.
|
350
|
+
module MicrophoneDistance
|
351
|
+
# Audio type is not known.
|
352
|
+
MICROPHONE_DISTANCE_UNSPECIFIED = 0
|
353
|
+
|
354
|
+
# The audio was captured from a closely placed microphone. Eg. phone,
|
355
|
+
# dictaphone, or handheld microphone. Generally if there speaker is within
|
356
|
+
# 1 meter of the microphone.
|
357
|
+
NEARFIELD = 1
|
358
|
+
|
359
|
+
# The speaker if within 3 meters of the microphone.
|
360
|
+
MIDFIELD = 2
|
361
|
+
|
362
|
+
# The speaker is more than 3 meters away from the microphone.
|
363
|
+
FARFIELD = 3
|
364
|
+
end
|
365
|
+
|
366
|
+
# The original media the speech was recorded on.
|
367
|
+
module OriginalMediaType
|
368
|
+
# Unknown original media type.
|
369
|
+
ORIGINAL_MEDIA_TYPE_UNSPECIFIED = 0
|
370
|
+
|
371
|
+
# The speech data is an audio recording.
|
372
|
+
AUDIO = 1
|
373
|
+
|
374
|
+
# The speech data originally recorded on a video.
|
375
|
+
VIDEO = 2
|
376
|
+
end
|
377
|
+
|
378
|
+
# The type of device the speech was recorded with.
|
379
|
+
module RecordingDeviceType
|
380
|
+
# The recording device is unknown.
|
381
|
+
RECORDING_DEVICE_TYPE_UNSPECIFIED = 0
|
382
|
+
|
383
|
+
# Speech was recorded on a smartphone.
|
384
|
+
SMARTPHONE = 1
|
385
|
+
|
386
|
+
# Speech was recorded using a personal computer or tablet.
|
387
|
+
PC = 2
|
388
|
+
|
389
|
+
# Speech was recorded over a phone line.
|
390
|
+
PHONE_LINE = 3
|
391
|
+
|
392
|
+
# Speech was recorded in a vehicle.
|
393
|
+
VEHICLE = 4
|
394
|
+
|
395
|
+
# Speech was recorded outdoors.
|
396
|
+
OTHER_OUTDOOR_DEVICE = 5
|
397
|
+
|
398
|
+
# Speech was recorded indoors.
|
399
|
+
OTHER_INDOOR_DEVICE = 6
|
400
|
+
end
|
401
|
+
end
|
402
|
+
|
403
|
+
# Provides "hints" to the speech recognizer to favor specific words and phrases
|
404
|
+
# in the results.
|
405
|
+
# @!attribute [rw] phrases
|
406
|
+
# @return [Array<String>]
|
407
|
+
# *Optional* A list of strings containing words and phrases "hints" so that
|
408
|
+
# the speech recognition is more likely to recognize them. This can be used
|
409
|
+
# to improve the accuracy for specific words and phrases, for example, if
|
410
|
+
# specific commands are typically spoken by the user. This can also be used
|
411
|
+
# to add additional words to the vocabulary of the recognizer. See
|
412
|
+
# [usage limits](https://cloud.google.com/speech/limits#content).
|
413
|
+
class SpeechContext; end
|
414
|
+
|
415
|
+
# Contains audio data in the encoding specified in the +RecognitionConfig+.
|
416
|
+
# Either +content+ or +uri+ must be supplied. Supplying both or neither
|
417
|
+
# returns {Google::Rpc::Code::INVALID_ARGUMENT}. See
|
418
|
+
# [audio limits](https://cloud.google.com/speech/limits#content).
|
419
|
+
# @!attribute [rw] content
|
420
|
+
# @return [String]
|
421
|
+
# The audio data bytes encoded as specified in
|
422
|
+
# +RecognitionConfig+. Note: as with all bytes fields, protobuffers use a
|
423
|
+
# pure binary representation, whereas JSON representations use base64.
|
424
|
+
# @!attribute [rw] uri
|
425
|
+
# @return [String]
|
426
|
+
# URI that points to a file that contains audio data bytes as specified in
|
427
|
+
# +RecognitionConfig+. Currently, only Google Cloud Storage URIs are
|
428
|
+
# supported, which must be specified in the following format:
|
429
|
+
# +gs://bucket_name/object_name+ (other URI formats return
|
430
|
+
# {Google::Rpc::Code::INVALID_ARGUMENT}). For more information, see
|
431
|
+
# [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
|
432
|
+
class RecognitionAudio; end
|
433
|
+
|
434
|
+
# The only message returned to the client by the +Recognize+ method. It
|
435
|
+
# contains the result as zero or more sequential +SpeechRecognitionResult+
|
436
|
+
# messages.
|
437
|
+
# @!attribute [rw] results
|
438
|
+
# @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechRecognitionResult>]
|
439
|
+
# Output only. Sequential list of transcription results corresponding to
|
440
|
+
# sequential portions of audio.
|
441
|
+
class RecognizeResponse; end
|
442
|
+
|
443
|
+
# The only message returned to the client by the +LongRunningRecognize+ method.
|
444
|
+
# It contains the result as zero or more sequential +SpeechRecognitionResult+
|
445
|
+
# messages. It is included in the +result.response+ field of the +Operation+
|
446
|
+
# returned by the +GetOperation+ call of the +google::longrunning::Operations+
|
447
|
+
# service.
|
448
|
+
# @!attribute [rw] results
|
449
|
+
# @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechRecognitionResult>]
|
450
|
+
# Output only. Sequential list of transcription results corresponding to
|
451
|
+
# sequential portions of audio.
|
452
|
+
class LongRunningRecognizeResponse; end
|
453
|
+
|
454
|
+
# Describes the progress of a long-running +LongRunningRecognize+ call. It is
|
455
|
+
# included in the +metadata+ field of the +Operation+ returned by the
|
456
|
+
# +GetOperation+ call of the +google::longrunning::Operations+ service.
|
457
|
+
# @!attribute [rw] progress_percent
|
458
|
+
# @return [Integer]
|
459
|
+
# Approximate percentage of audio processed thus far. Guaranteed to be 100
|
460
|
+
# when the audio is fully processed and the results are available.
|
461
|
+
# @!attribute [rw] start_time
|
462
|
+
# @return [Google::Protobuf::Timestamp]
|
463
|
+
# Time when the request was received.
|
464
|
+
# @!attribute [rw] last_update_time
|
465
|
+
# @return [Google::Protobuf::Timestamp]
|
466
|
+
# Time of the most recent processing update.
|
467
|
+
class LongRunningRecognizeMetadata; end
|
468
|
+
|
469
|
+
# +StreamingRecognizeResponse+ is the only message returned to the client by
|
470
|
+
# +StreamingRecognize+. A series of zero or more +StreamingRecognizeResponse+
|
471
|
+
# messages are streamed back to the client. If there is no recognizable
|
472
|
+
# audio, and +single_utterance+ is set to false, then no messages are streamed
|
473
|
+
# back to the client.
|
474
|
+
#
|
475
|
+
# Here's an example of a series of ten +StreamingRecognizeResponse+s that might
|
476
|
+
# be returned while processing audio:
|
477
|
+
#
|
478
|
+
# 1. results { alternatives { transcript: "tube" } stability: 0.01 }
|
479
|
+
#
|
480
|
+
# 2. results { alternatives { transcript: "to be a" } stability: 0.01 }
|
481
|
+
#
|
482
|
+
# 3. results { alternatives { transcript: "to be" } stability: 0.9 }
|
483
|
+
# results { alternatives { transcript: " or not to be" } stability: 0.01 }
|
484
|
+
#
|
485
|
+
# 4. results { alternatives { transcript: "to be or not to be"
|
486
|
+
# confidence: 0.92 }
|
487
|
+
# alternatives { transcript: "to bee or not to bee" }
|
488
|
+
# is_final: true }
|
489
|
+
#
|
490
|
+
# 5. results { alternatives { transcript: " that's" } stability: 0.01 }
|
491
|
+
#
|
492
|
+
# 6. results { alternatives { transcript: " that is" } stability: 0.9 }
|
493
|
+
# results { alternatives { transcript: " the question" } stability: 0.01 }
|
494
|
+
#
|
495
|
+
# 7. results { alternatives { transcript: " that is the question"
|
496
|
+
# confidence: 0.98 }
|
497
|
+
# alternatives { transcript: " that was the question" }
|
498
|
+
# is_final: true }
|
499
|
+
#
|
500
|
+
# Notes:
|
501
|
+
#
|
502
|
+
# * Only two of the above responses #4 and #7 contain final results; they are
|
503
|
+
# indicated by +is_final: true+. Concatenating these together generates the
|
504
|
+
# full transcript: "to be or not to be that is the question".
|
505
|
+
#
|
506
|
+
# * The others contain interim +results+. #3 and #6 contain two interim
|
507
|
+
# +results+: the first portion has a high stability and is less likely to
|
508
|
+
# change; the second portion has a low stability and is very likely to
|
509
|
+
# change. A UI designer might choose to show only high stability +results+.
|
510
|
+
#
|
511
|
+
# * The specific +stability+ and +confidence+ values shown above are only for
|
512
|
+
# illustrative purposes. Actual values may vary.
|
513
|
+
#
|
514
|
+
# * In each response, only one of these fields will be set:
|
515
|
+
# +error+,
|
516
|
+
# +speech_event_type+, or
|
517
|
+
# one or more (repeated) +results+.
|
518
|
+
# @!attribute [rw] error
|
519
|
+
# @return [Google::Rpc::Status]
|
520
|
+
# Output only. If set, returns a {Google::Rpc::Status} message that
|
521
|
+
# specifies the error for the operation.
|
522
|
+
# @!attribute [rw] results
|
523
|
+
# @return [Array<Google::Cloud::Speech::V1p1beta1::StreamingRecognitionResult>]
|
524
|
+
# Output only. This repeated list contains zero or more results that
|
525
|
+
# correspond to consecutive portions of the audio currently being processed.
|
526
|
+
# It contains zero or one +is_final=true+ result (the newly settled portion),
|
527
|
+
# followed by zero or more +is_final=false+ results (the interim results).
|
528
|
+
# @!attribute [rw] speech_event_type
|
529
|
+
# @return [Google::Cloud::Speech::V1p1beta1::StreamingRecognizeResponse::SpeechEventType]
|
530
|
+
# Output only. Indicates the type of speech event.
|
531
|
+
class StreamingRecognizeResponse
|
532
|
+
# Indicates the type of speech event.
|
533
|
+
module SpeechEventType
|
534
|
+
# No speech event specified.
|
535
|
+
SPEECH_EVENT_UNSPECIFIED = 0
|
536
|
+
|
537
|
+
# This event indicates that the server has detected the end of the user's
|
538
|
+
# speech utterance and expects no additional speech. Therefore, the server
|
539
|
+
# will not process additional audio (although it may subsequently return
|
540
|
+
# additional results). The client should stop sending additional audio
|
541
|
+
# data, half-close the gRPC connection, and wait for any additional results
|
542
|
+
# until the server closes the gRPC connection. This event is only sent if
|
543
|
+
# +single_utterance+ was set to +true+, and is not used otherwise.
|
544
|
+
END_OF_SINGLE_UTTERANCE = 1
|
545
|
+
end
|
546
|
+
end
|
547
|
+
|
548
|
+
# A streaming speech recognition result corresponding to a portion of the audio
|
549
|
+
# that is currently being processed.
|
550
|
+
# @!attribute [rw] alternatives
|
551
|
+
# @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechRecognitionAlternative>]
|
552
|
+
# Output only. May contain one or more recognition hypotheses (up to the
|
553
|
+
# maximum specified in +max_alternatives+).
|
554
|
+
# These alternatives are ordered in terms of accuracy, with the top (first)
|
555
|
+
# alternative being the most probable, as ranked by the recognizer.
|
556
|
+
# @!attribute [rw] is_final
|
557
|
+
# @return [true, false]
|
558
|
+
# Output only. If +false+, this +StreamingRecognitionResult+ represents an
|
559
|
+
# interim result that may change. If +true+, this is the final time the
|
560
|
+
# speech service will return this particular +StreamingRecognitionResult+,
|
561
|
+
# the recognizer will not return any further hypotheses for this portion of
|
562
|
+
# the transcript and corresponding audio.
|
563
|
+
# @!attribute [rw] stability
|
564
|
+
# @return [Float]
|
565
|
+
# Output only. An estimate of the likelihood that the recognizer will not
|
566
|
+
# change its guess about this interim result. Values range from 0.0
|
567
|
+
# (completely unstable) to 1.0 (completely stable).
|
568
|
+
# This field is only provided for interim results (+is_final=false+).
|
569
|
+
# The default of 0.0 is a sentinel value indicating +stability+ was not set.
|
570
|
+
class StreamingRecognitionResult; end
|
571
|
+
|
572
|
+
# A speech recognition result corresponding to a portion of the audio.
|
573
|
+
# @!attribute [rw] alternatives
|
574
|
+
# @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechRecognitionAlternative>]
|
575
|
+
# Output only. May contain one or more recognition hypotheses (up to the
|
576
|
+
# maximum specified in +max_alternatives+).
|
577
|
+
# These alternatives are ordered in terms of accuracy, with the top (first)
|
578
|
+
# alternative being the most probable, as ranked by the recognizer.
|
579
|
+
class SpeechRecognitionResult; end
|
580
|
+
|
581
|
+
# Alternative hypotheses (a.k.a. n-best list).
|
582
|
+
# @!attribute [rw] transcript
|
583
|
+
# @return [String]
|
584
|
+
# Output only. Transcript text representing the words that the user spoke.
|
585
|
+
# @!attribute [rw] confidence
|
586
|
+
# @return [Float]
|
587
|
+
# Output only. The confidence estimate between 0.0 and 1.0. A higher number
|
588
|
+
# indicates an estimated greater likelihood that the recognized words are
|
589
|
+
# correct. This field is set only for the top alternative of a non-streaming
|
590
|
+
# result or, of a streaming result where +is_final=true+.
|
591
|
+
# This field is not guaranteed to be accurate and users should not rely on it
|
592
|
+
# to be always provided.
|
593
|
+
# The default of 0.0 is a sentinel value indicating +confidence+ was not set.
|
594
|
+
# @!attribute [rw] words
|
595
|
+
# @return [Array<Google::Cloud::Speech::V1p1beta1::WordInfo>]
|
596
|
+
# Output only. A list of word-specific information for each recognized word.
|
597
|
+
# Note: When enable_speaker_diarization is true, you will see all the words
|
598
|
+
# from the beginning of the audio.
|
599
|
+
class SpeechRecognitionAlternative; end
|
600
|
+
|
601
|
+
# Word-specific information for recognized words.
|
602
|
+
# @!attribute [rw] start_time
|
603
|
+
# @return [Google::Protobuf::Duration]
|
604
|
+
# Output only. Time offset relative to the beginning of the audio,
|
605
|
+
# and corresponding to the start of the spoken word.
|
606
|
+
# This field is only set if +enable_word_time_offsets=true+ and only
|
607
|
+
# in the top hypothesis.
|
608
|
+
# This is an experimental feature and the accuracy of the time offset can
|
609
|
+
# vary.
|
610
|
+
# @!attribute [rw] end_time
|
611
|
+
# @return [Google::Protobuf::Duration]
|
612
|
+
# Output only. Time offset relative to the beginning of the audio,
|
613
|
+
# and corresponding to the end of the spoken word.
|
614
|
+
# This field is only set if +enable_word_time_offsets=true+ and only
|
615
|
+
# in the top hypothesis.
|
616
|
+
# This is an experimental feature and the accuracy of the time offset can
|
617
|
+
# vary.
|
618
|
+
# @!attribute [rw] word
|
619
|
+
# @return [String]
|
620
|
+
# Output only. The word corresponding to this set of information.
|
621
|
+
class WordInfo; end
|
622
|
+
end
|
623
|
+
end
|
624
|
+
end
|
625
|
+
end
|