google-cloud-speech 0.29.0 → 0.30.0

data/lib/google/cloud/speech/v1p1beta1.rb

@@ -0,0 +1,126 @@
+# Copyright 2018 Google LLC
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     https://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+require "google/cloud/speech/v1p1beta1/speech_client"
+require "google/cloud/speech/v1p1beta1/helpers"
+
+module Google
+  module Cloud
+    # rubocop:disable LineLength
+
+    ##
+    # # Ruby Client for Cloud Speech API ([Alpha](https://github.com/GoogleCloudPlatform/google-cloud-ruby#versioning))
+    #
+    # [Cloud Speech API][Product Documentation]:
+    # Converts audio to text by applying powerful neural network models.
+    # - [Product Documentation][]
+    #
+    # ## Quick Start
+    # In order to use this library, you first need to go through the following
+    # steps:
+    #
+    # 1. [Select or create a Cloud Platform project.](https://console.cloud.google.com/project)
+    # 2. [Enable billing for your project.](https://cloud.google.com/billing/docs/how-to/modify-project#enable_billing_for_a_project)
+    # 3. [Enable the Cloud Speech API.](https://console.cloud.google.com/apis/api/speech)
+    # 4. [Setup Authentication.](https://googlecloudplatform.github.io/google-cloud-ruby/#/docs/google-cloud/master/guides/authentication)
+    #
+    # ### Preview
+    # #### SpeechClient
+    # ```rb
+    # require "google/cloud/speech/v1p1beta1"
+    #
+    # speech_client = Google::Cloud::Speech::V1p1beta1.new
+    # language_code = "en-US"
+    # sample_rate_hertz = 44100
+    # encoding = :FLAC
+    # config = {
+    #   language_code: language_code,
+    #   sample_rate_hertz: sample_rate_hertz,
+    #   encoding: encoding
+    # }
+    # uri = "gs://gapic-toolkit/hello.flac"
+    # audio = { uri: uri }
+    # response = speech_client.recognize(config, audio)
+    # ```
+    #
+    # ### Next Steps
+    # - Read the [Cloud Speech API Product documentation][Product Documentation]
+    #   to learn more about the product and see How-to Guides.
+    # - View this [repository's main README](https://github.com/GoogleCloudPlatform/google-cloud-ruby/blob/master/README.md)
+    #   to see the full list of Cloud APIs that we cover.
+    #
+    # [Product Documentation]: https://cloud.google.com/speech
+    #
+    #
+    module Speech
+      module V1p1beta1
+        # rubocop:enable LineLength
+
+        ##
+        # Service that implements Google Cloud Speech API.
+        #
+        # @param credentials [Google::Auth::Credentials, String, Hash, GRPC::Core::Channel, GRPC::Core::ChannelCredentials, Proc]
+        #   Provides the means for authenticating requests made by the client. This parameter can
+        #   be many types.
+        #   A `Google::Auth::Credentials` uses a the properties of its represented keyfile for
+        #   authenticating requests made by this client.
+        #   A `String` will be treated as the path to the keyfile to be used for the construction of
+        #   credentials for this client.
+        #   A `Hash` will be treated as the contents of a keyfile to be used for the construction of
+        #   credentials for this client.
+        #   A `GRPC::Core::Channel` will be used to make calls through.
+        #   A `GRPC::Core::ChannelCredentials` for the setting up the RPC client. The channel credentials
+        #   should already be composed with a `GRPC::Core::CallCredentials` object.
+        #   A `Proc` will be used as an updater_proc for the Grpc channel. The proc transforms the
+        #   metadata for requests, generally, to give OAuth credentials.
+        # @param scopes [Array<String>]
+        #   The OAuth scopes for this service. This parameter is ignored if
+        #   an updater_proc is supplied.
+        # @param client_config [Hash]
+        #   A Hash for call options for each method. See
+        #   Google::Gax#construct_settings for the structure of
+        #   this data. Falls back to the default config if not specified
+        #   or the specified config is missing data points.
+        # @param timeout [Numeric]
+        #   The default timeout, in seconds, for calls made through this client.
+        # @param metadata [Hash]
+        #   Default metadata to be sent with each request. This can be overridden on a per call basis.
+        # @param exception_transformer [Proc]
+        #   An optional proc that intercepts any exceptions raised during an API call to inject
+        #   custom error handling.
+        def self.new \
+            credentials: nil,
+            scopes: nil,
+            client_config: nil,
+            timeout: nil,
+            metadata: nil,
+            exception_transformer: nil,
+            lib_name: nil,
+            lib_version: nil
+          kwargs = {
+            credentials: credentials,
+            scopes: scopes,
+            client_config: client_config,
+            timeout: timeout,
+            metadata: metadata,
+            exception_transformer: exception_transformer,
+            lib_name: lib_name,
+            lib_version: lib_version
+          }.select { |_, v| v != nil }
+          Google::Cloud::Speech::V1p1beta1::SpeechClient.new(**kwargs)
+        end
+      end
+    end
+  end
+end

data/lib/google/cloud/speech/v1p1beta1/cloud_speech_pb.rb

@@ -0,0 +1,175 @@
+# Generated by the protocol buffer compiler.  DO NOT EDIT!
+# source: google/cloud/speech/v1p1beta1/cloud_speech.proto
+
+require 'google/protobuf'
+
+require 'google/api/annotations_pb'
+require 'google/longrunning/operations_pb'
+require 'google/protobuf/any_pb'
+require 'google/protobuf/duration_pb'
+require 'google/protobuf/timestamp_pb'
+require 'google/rpc/status_pb'
+Google::Protobuf::DescriptorPool.generated_pool.build do
+  add_message "google.cloud.speech.v1p1beta1.RecognizeRequest" do
+    optional :config, :message, 1, "google.cloud.speech.v1p1beta1.RecognitionConfig"
+    optional :audio, :message, 2, "google.cloud.speech.v1p1beta1.RecognitionAudio"
+  end
+  add_message "google.cloud.speech.v1p1beta1.LongRunningRecognizeRequest" do
+    optional :config, :message, 1, "google.cloud.speech.v1p1beta1.RecognitionConfig"
+    optional :audio, :message, 2, "google.cloud.speech.v1p1beta1.RecognitionAudio"
+  end
+  add_message "google.cloud.speech.v1p1beta1.StreamingRecognizeRequest" do
+    oneof :streaming_request do
+      optional :streaming_config, :message, 1, "google.cloud.speech.v1p1beta1.StreamingRecognitionConfig"
+      optional :audio_content, :bytes, 2
+    end
+  end
+  add_message "google.cloud.speech.v1p1beta1.StreamingRecognitionConfig" do
+    optional :config, :message, 1, "google.cloud.speech.v1p1beta1.RecognitionConfig"
+    optional :single_utterance, :bool, 2
+    optional :interim_results, :bool, 3
+  end
+  add_message "google.cloud.speech.v1p1beta1.RecognitionConfig" do
+    optional :encoding, :enum, 1, "google.cloud.speech.v1p1beta1.RecognitionConfig.AudioEncoding"
+    optional :sample_rate_hertz, :int32, 2
+    optional :language_code, :string, 3
+    optional :max_alternatives, :int32, 4
+    optional :profanity_filter, :bool, 5
+    repeated :speech_contexts, :message, 6, "google.cloud.speech.v1p1beta1.SpeechContext"
+    optional :enable_word_time_offsets, :bool, 8
+    optional :enable_automatic_punctuation, :bool, 11
+    optional :metadata, :message, 9, "google.cloud.speech.v1p1beta1.RecognitionMetadata"
+    optional :model, :string, 13
+    optional :use_enhanced, :bool, 14
+  end
+  add_enum "google.cloud.speech.v1p1beta1.RecognitionConfig.AudioEncoding" do
+    value :ENCODING_UNSPECIFIED, 0
+    value :LINEAR16, 1
+    value :FLAC, 2
+    value :MULAW, 3
+    value :AMR, 4
+    value :AMR_WB, 5
+    value :OGG_OPUS, 6
+    value :SPEEX_WITH_HEADER_BYTE, 7
+  end
+  add_message "google.cloud.speech.v1p1beta1.RecognitionMetadata" do
+    optional :interaction_type, :enum, 1, "google.cloud.speech.v1p1beta1.RecognitionMetadata.InteractionType"
+    optional :industry_naics_code_of_audio, :uint32, 3
+    optional :microphone_distance, :enum, 4, "google.cloud.speech.v1p1beta1.RecognitionMetadata.MicrophoneDistance"
+    optional :original_media_type, :enum, 5, "google.cloud.speech.v1p1beta1.RecognitionMetadata.OriginalMediaType"
+    optional :recording_device_type, :enum, 6, "google.cloud.speech.v1p1beta1.RecognitionMetadata.RecordingDeviceType"
+    optional :recording_device_name, :string, 7
+    optional :original_mime_type, :string, 8
+    optional :obfuscated_id, :int64, 9
+    optional :audio_topic, :string, 10
+  end
+  add_enum "google.cloud.speech.v1p1beta1.RecognitionMetadata.InteractionType" do
+    value :INTERACTION_TYPE_UNSPECIFIED, 0
+    value :DISCUSSION, 1
+    value :PRESENTATION, 2
+    value :PHONE_CALL, 3
+    value :VOICEMAIL, 4
+    value :PROFESSIONALLY_PRODUCED, 5
+    value :VOICE_SEARCH, 6
+    value :VOICE_COMMAND, 7
+    value :DICTATION, 8
+  end
+  add_enum "google.cloud.speech.v1p1beta1.RecognitionMetadata.MicrophoneDistance" do
+    value :MICROPHONE_DISTANCE_UNSPECIFIED, 0
+    value :NEARFIELD, 1
+    value :MIDFIELD, 2
+    value :FARFIELD, 3
+  end
+  add_enum "google.cloud.speech.v1p1beta1.RecognitionMetadata.OriginalMediaType" do
+    value :ORIGINAL_MEDIA_TYPE_UNSPECIFIED, 0
+    value :AUDIO, 1
+    value :VIDEO, 2
+  end
+  add_enum "google.cloud.speech.v1p1beta1.RecognitionMetadata.RecordingDeviceType" do
+    value :RECORDING_DEVICE_TYPE_UNSPECIFIED, 0
+    value :SMARTPHONE, 1
+    value :PC, 2
+    value :PHONE_LINE, 3
+    value :VEHICLE, 4
+    value :OTHER_OUTDOOR_DEVICE, 5
+    value :OTHER_INDOOR_DEVICE, 6
+  end
+  add_message "google.cloud.speech.v1p1beta1.SpeechContext" do
+    repeated :phrases, :string, 1
+  end
+  add_message "google.cloud.speech.v1p1beta1.RecognitionAudio" do
+    oneof :audio_source do
+      optional :content, :bytes, 1
+      optional :uri, :string, 2
+    end
+  end
+  add_message "google.cloud.speech.v1p1beta1.RecognizeResponse" do
+    repeated :results, :message, 2, "google.cloud.speech.v1p1beta1.SpeechRecognitionResult"
+  end
+  add_message "google.cloud.speech.v1p1beta1.LongRunningRecognizeResponse" do
+    repeated :results, :message, 2, "google.cloud.speech.v1p1beta1.SpeechRecognitionResult"
+  end
+  add_message "google.cloud.speech.v1p1beta1.LongRunningRecognizeMetadata" do
+    optional :progress_percent, :int32, 1
+    optional :start_time, :message, 2, "google.protobuf.Timestamp"
+    optional :last_update_time, :message, 3, "google.protobuf.Timestamp"
+  end
+  add_message "google.cloud.speech.v1p1beta1.StreamingRecognizeResponse" do
+    optional :error, :message, 1, "google.rpc.Status"
+    repeated :results, :message, 2, "google.cloud.speech.v1p1beta1.StreamingRecognitionResult"
+    optional :speech_event_type, :enum, 4, "google.cloud.speech.v1p1beta1.StreamingRecognizeResponse.SpeechEventType"
+  end
+  add_enum "google.cloud.speech.v1p1beta1.StreamingRecognizeResponse.SpeechEventType" do
+    value :SPEECH_EVENT_UNSPECIFIED, 0
+    value :END_OF_SINGLE_UTTERANCE, 1
+  end
+  add_message "google.cloud.speech.v1p1beta1.StreamingRecognitionResult" do
+    repeated :alternatives, :message, 1, "google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative"
+    optional :is_final, :bool, 2
+    optional :stability, :float, 3
+  end
+  add_message "google.cloud.speech.v1p1beta1.SpeechRecognitionResult" do
+    repeated :alternatives, :message, 1, "google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative"
+  end
+  add_message "google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative" do
+    optional :transcript, :string, 1
+    optional :confidence, :float, 2
+    repeated :words, :message, 3, "google.cloud.speech.v1p1beta1.WordInfo"
+  end
+  add_message "google.cloud.speech.v1p1beta1.WordInfo" do
+    optional :start_time, :message, 1, "google.protobuf.Duration"
+    optional :end_time, :message, 2, "google.protobuf.Duration"
+    optional :word, :string, 3
+  end
+end
+
+module Google
+  module Cloud
+    module Speech
+      module V1p1beta1
+        RecognizeRequest = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognizeRequest").msgclass
+        LongRunningRecognizeRequest = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.LongRunningRecognizeRequest").msgclass
+        StreamingRecognizeRequest = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.StreamingRecognizeRequest").msgclass
+        StreamingRecognitionConfig = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.StreamingRecognitionConfig").msgclass
+        RecognitionConfig = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionConfig").msgclass
+        RecognitionConfig::AudioEncoding = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionConfig.AudioEncoding").enummodule
+        RecognitionMetadata = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionMetadata").msgclass
+        RecognitionMetadata::InteractionType = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionMetadata.InteractionType").enummodule
+        RecognitionMetadata::MicrophoneDistance = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionMetadata.MicrophoneDistance").enummodule
+        RecognitionMetadata::OriginalMediaType = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionMetadata.OriginalMediaType").enummodule
+        RecognitionMetadata::RecordingDeviceType = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionMetadata.RecordingDeviceType").enummodule
+        SpeechContext = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.SpeechContext").msgclass
+        RecognitionAudio = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognitionAudio").msgclass
+        RecognizeResponse = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.RecognizeResponse").msgclass
+        LongRunningRecognizeResponse = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.LongRunningRecognizeResponse").msgclass
+        LongRunningRecognizeMetadata = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.LongRunningRecognizeMetadata").msgclass
+        StreamingRecognizeResponse = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.StreamingRecognizeResponse").msgclass
+        StreamingRecognizeResponse::SpeechEventType = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.StreamingRecognizeResponse.SpeechEventType").enummodule
+        StreamingRecognitionResult = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.StreamingRecognitionResult").msgclass
+        SpeechRecognitionResult = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.SpeechRecognitionResult").msgclass
+        SpeechRecognitionAlternative = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.SpeechRecognitionAlternative").msgclass
+        WordInfo = Google::Protobuf::DescriptorPool.generated_pool.lookup("google.cloud.speech.v1p1beta1.WordInfo").msgclass
+      end
+    end
+  end
+end

data/lib/google/cloud/speech/v1p1beta1/cloud_speech_services_pb.rb

@@ -0,0 +1,54 @@
+# Generated by the protocol buffer compiler.  DO NOT EDIT!
+# Source: google/cloud/speech/v1p1beta1/cloud_speech.proto for package 'google.cloud.speech.v1p1beta1'
+# Original file comments:
+# Copyright 2018 Google Inc.
+#
+# 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.
+#
+
+require 'grpc'
+require 'google/cloud/speech/v1p1beta1/cloud_speech_pb'
+
+module Google
+  module Cloud
+    module Speech
+      module V1p1beta1
+        module Speech
+          # Service that implements Google Cloud Speech API.
+          class Service
+
+            include GRPC::GenericService
+
+            self.marshal_class_method = :encode
+            self.unmarshal_class_method = :decode
+            self.service_name = 'google.cloud.speech.v1p1beta1.Speech'
+
+            # Performs synchronous speech recognition: receive results after all audio
+            # has been sent and processed.
+            rpc :Recognize, RecognizeRequest, RecognizeResponse
+            # 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.
+            rpc :LongRunningRecognize, LongRunningRecognizeRequest, Google::Longrunning::Operation
+            # 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), stream(StreamingRecognizeResponse)
+          end
+
+          Stub = Service.rpc_stub_class
+        end
+      end
+    end
+  end
+end

data/lib/google/cloud/speech/v1p1beta1/credentials.rb

@@ -0,0 +1,32 @@
+# Copyright 2018 Google LLC
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     https://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+require "googleauth"
+
+module Google
+  module Cloud
+    module Speech
+      module V1p1beta1
+        class Credentials < Google::Auth::Credentials
+          SCOPE = [
+            "https://www.googleapis.com/auth/cloud-platform"
+          ].freeze
+          PATH_ENV_VARS = %w(SPEECH_KEYFILE GOOGLE_CLOUD_KEYFILE GCLOUD_KEYFILE)
+          JSON_ENV_VARS = %w(SPEECH_KEYFILE_JSON GOOGLE_CLOUD_KEYFILE_JSON GCLOUD_KEYFILE_JSON)
+          DEFAULT_PATHS = ["~/.config/gcloud/application_default_credentials.json"]
+        end
+      end
+    end
+  end
+end

data/lib/google/cloud/speech/v1p1beta1/doc/google/cloud/speech/v1p1beta1/cloud_speech.rb

@@ -0,0 +1,625 @@
+# Copyright 2018 Google LLC
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     https://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+module Google
+  module Cloud
+    module Speech
+      ##
+      # # Cloud Speech API Contents
+      #
+      # | Class | Description |
+      # | ----- | ----------- |
+      # | [SpeechClient][] | Service that implements Google Cloud Speech API. |
+      # | [Data Types][] | Data types for Google::Cloud::Speech::V1p1beta1 |
+      #
+      # [SpeechClient]: https://googlecloudplatform.github.io/google-cloud-ruby/#/docs/google-cloud-speech/latest/google/cloud/speech/v1p1beta1/speechclient
+      # [Data Types]: https://googlecloudplatform.github.io/google-cloud-ruby/#/docs/google-cloud-speech/latest/google/cloud/speech/v1p1beta1/datatypes
+      #
+      module V1p1beta1
+        # The top-level message sent by the client for the +Recognize+ method.
+        # @!attribute [rw] config
+        #   @return [Google::Cloud::Speech::V1p1beta1::RecognitionConfig]
+        #     *Required* Provides information to the recognizer that specifies how to
+        #     process the request.
+        # @!attribute [rw] audio
+        #   @return [Google::Cloud::Speech::V1p1beta1::RecognitionAudio]
+        #     *Required* The audio data to be recognized.
+        class RecognizeRequest; end
+
+        # The top-level message sent by the client for the +LongRunningRecognize+
+        # method.
+        # @!attribute [rw] config
+        #   @return [Google::Cloud::Speech::V1p1beta1::RecognitionConfig]
+        #     *Required* Provides information to the recognizer that specifies how to
+        #     process the request.
+        # @!attribute [rw] audio
+        #   @return [Google::Cloud::Speech::V1p1beta1::RecognitionAudio]
+        #     *Required* The audio data to be recognized.
+        class LongRunningRecognizeRequest; end
+
+        # The top-level message sent by the client for the +StreamingRecognize+ method.
+        # Multiple +StreamingRecognizeRequest+ messages are sent. The first message
+        # must contain a +streaming_config+ message and must not contain +audio+ data.
+        # All subsequent messages must contain +audio+ data and must not contain a
+        # +streaming_config+ message.
+        # @!attribute [rw] streaming_config
+        #   @return [Google::Cloud::Speech::V1p1beta1::StreamingRecognitionConfig]
+        #     Provides information to the recognizer that specifies how to process the
+        #     request. The first +StreamingRecognizeRequest+ message must contain a
+        #     +streaming_config+  message.
+        # @!attribute [rw] audio_content
+        #   @return [String]
+        #     The audio data to be recognized. Sequential chunks of audio data are sent
+        #     in sequential +StreamingRecognizeRequest+ messages. The first
+        #     +StreamingRecognizeRequest+ message must not contain +audio_content+ data
+        #     and all subsequent +StreamingRecognizeRequest+ messages must contain
+        #     +audio_content+ data. The audio bytes must be encoded as specified in
+        #     +RecognitionConfig+. Note: as with all bytes fields, protobuffers use a
+        #     pure binary representation (not base64). See
+        #     [audio limits](https://cloud.google.com/speech/limits#content).
+        class StreamingRecognizeRequest; end
+
+        # Provides information to the recognizer that specifies how to process the
+        # request.
+        # @!attribute [rw] config
+        #   @return [Google::Cloud::Speech::V1p1beta1::RecognitionConfig]
+        #     *Required* Provides information to the recognizer that specifies how to
+        #     process the request.
+        # @!attribute [rw] single_utterance
+        #   @return [true, false]
+        #     *Optional* If +false+ or omitted, the recognizer will perform continuous
+        #     recognition (continuing to wait for and process audio even if the user
+        #     pauses speaking) until the client closes the input stream (gRPC API) or
+        #     until the maximum time limit has been reached. May return multiple
+        #     +StreamingRecognitionResult+s with the +is_final+ flag set to +true+.
+        #
+        #     If +true+, the recognizer will detect a single spoken utterance. When it
+        #     detects that the user has paused or stopped speaking, it will return an
+        #     +END_OF_SINGLE_UTTERANCE+ event and cease recognition. It will return no
+        #     more than one +StreamingRecognitionResult+ with the +is_final+ flag set to
+        #     +true+.
+        # @!attribute [rw] interim_results
+        #   @return [true, false]
+        #     *Optional* If +true+, interim results (tentative hypotheses) may be
+        #     returned as they become available (these interim results are indicated with
+        #     the +is_final=false+ flag).
+        #     If +false+ or omitted, only +is_final=true+ result(s) are returned.
+        class StreamingRecognitionConfig; end
+
+        # Provides information to the recognizer that specifies how to process the
+        # request.
+        # @!attribute [rw] encoding
+        #   @return [Google::Cloud::Speech::V1p1beta1::RecognitionConfig::AudioEncoding]
+        #     Encoding of audio data sent in all +RecognitionAudio+ messages.
+        #     This field is optional for +FLAC+ and +WAV+ audio files and required
+        #     for all other audio formats. For details, see {Google::Cloud::Speech::V1p1beta1::RecognitionConfig::AudioEncoding AudioEncoding}.
+        # @!attribute [rw] sample_rate_hertz
+        #   @return [Integer]
+        #     Sample rate in Hertz of the audio data sent in all
+        #     +RecognitionAudio+ messages. Valid values are: 8000-48000.
+        #     16000 is optimal. For best results, set the sampling rate of the audio
+        #     source to 16000 Hz. If that's not possible, use the native sample rate of
+        #     the audio source (instead of re-sampling).
+        #     This field is optional for +FLAC+ and +WAV+ audio files and required
+        #     for all other audio formats. For details, see {Google::Cloud::Speech::V1p1beta1::RecognitionConfig::AudioEncoding AudioEncoding}.
+        # @!attribute [rw] language_code
+        #   @return [String]
+        #     *Required* The language of the supplied audio as a
+        #     [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag.
+        #     Example: "en-US".
+        #     See [Language Support](https://cloud.google.com/speech/docs/languages)
+        #     for a list of the currently supported language codes.
+        # @!attribute [rw] max_alternatives
+        #   @return [Integer]
+        #     *Optional* Maximum number of recognition hypotheses to be returned.
+        #     Specifically, the maximum number of +SpeechRecognitionAlternative+ messages
+        #     within each +SpeechRecognitionResult+.
+        #     The server may return fewer than +max_alternatives+.
+        #     Valid values are +0+-+30+. A value of +0+ or +1+ will return a maximum of
+        #     one. If omitted, will return a maximum of one.
+        # @!attribute [rw] profanity_filter
+        #   @return [true, false]
+        #     *Optional* If set to +true+, the server will attempt to filter out
+        #     profanities, replacing all but the initial character in each filtered word
+        #     with asterisks, e.g. "f***". If set to +false+ or omitted, profanities
+        #     won't be filtered out.
+        # @!attribute [rw] speech_contexts
+        #   @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechContext>]
+        #     *Optional* A means to provide context to assist the speech recognition.
+        # @!attribute [rw] enable_word_time_offsets
+        #   @return [true, false]
+        #     *Optional* If +true+, the top result includes a list of words and
+        #     the start and end time offsets (timestamps) for those words. If
+        #     +false+, no word-level time offset information is returned. The default is
+        #     +false+.
+        # @!attribute [rw] enable_automatic_punctuation
+        #   @return [true, false]
+        #     *Optional* If 'true', adds punctuation to recognition result hypotheses.
+        #     This feature is only available in select languages. Setting this for
+        #     requests in other languages has no effect at all.
+        #     The default 'false' value does not add punctuation to result hypotheses.
+        #     NOTE: "This is currently offered as an experimental service, complimentary
+        #     to all users. In the future this may be exclusively available as a
+        #     premium feature."
+        # @!attribute [rw] metadata
+        #   @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata]
+        #     *Optional* Metadata regarding this request.
+        # @!attribute [rw] model
+        #   @return [String]
+        #     *Optional* Which model to select for the given request. Select the model
+        #     best suited to your domain to get best results. If a model is not
+        #     explicitly specified, then we auto-select a model based on the parameters
+        #     in the RecognitionConfig.
+        #     <table>
+        #       <tr>
+        #         <td><b>Model</b></td>
+        #         <td><b>Description</b></td>
+        #       </tr>
+        #       <tr>
+        #         <td><code>command_and_search</code></td>
+        #         <td>Best for short queries such as voice commands or voice search.</td>
+        #       </tr>
+        #       <tr>
+        #         <td><code>phone_call</code></td>
+        #         <td>Best for audio that originated from a phone call (typically
+        #         recorded at an 8khz sampling rate).</td>
+        #       </tr>
+        #       <tr>
+        #         <td><code>video</code></td>
+        #         <td>Best for audio that originated from from video or includes multiple
+        #             speakers. Ideally the audio is recorded at a 16khz or greater
+        #             sampling rate. This is a premium model that costs more than the
+        #             standard rate.</td>
+        #       </tr>
+        #       <tr>
+        #         <td><code>default</code></td>
+        #         <td>Best for audio that is not one of the specific audio models.
+        #             For example, long-form audio. Ideally the audio is high-fidelity,
+        #             recorded at a 16khz or greater sampling rate.</td>
+        #       </tr>
+        #     </table>
+        # @!attribute [rw] use_enhanced
+        #   @return [true, false]
+        #     *Optional* Set to true to use an enhanced model for speech recognition.
+        #     You must also set the +model+ field to a valid, enhanced model. If
+        #     +use_enhanced+ is set to true and the +model+ field is not set, then
+        #     +use_enhanced+ is ignored. If +use_enhanced+ is true and an enhanced
+        #     version of the specified model does not exist, then the speech is
+        #     recognized using the standard version of the specified model.
+        #
+        #     Enhanced speech models require that you opt-in to the audio logging using
+        #     instructions in the [alpha documentation](https://cloud.google.com/speech/data-sharing). If you set
+        #     +use_enhanced+ to true and you have not enabled audio logging, then you
+        #     will receive an error.
+        class RecognitionConfig
+          # The encoding of the audio data sent in the request.
+          #
+          # All encodings support only 1 channel (mono) audio.
+          #
+          # For best results, the audio source should be captured and transmitted using
+          # a lossless encoding (+FLAC+ or +LINEAR16+). The accuracy of the speech
+          # recognition can be reduced if lossy codecs are used to capture or transmit
+          # audio, particularly if background noise is present. Lossy codecs include
+          # +MULAW+, +AMR+, +AMR_WB+, +OGG_OPUS+, and +SPEEX_WITH_HEADER_BYTE+.
+          #
+          # The +FLAC+ and +WAV+ audio file formats include a header that describes the
+          # included audio content. You can request recognition for +WAV+ files that
+          # contain either +LINEAR16+ or +MULAW+ encoded audio.
+          # If you send +FLAC+ or +WAV+ audio file format in
+          # your request, you do not need to specify an +AudioEncoding+; the audio
+          # encoding format is determined from the file header. If you specify
+          # an +AudioEncoding+ when you send  send +FLAC+ or +WAV+ audio, the
+          # encoding configuration must match the encoding described in the audio
+          # header; otherwise the request returns an
+          # {Google::Rpc::Code::INVALID_ARGUMENT} error code.
+          module AudioEncoding
+            # Not specified.
+            ENCODING_UNSPECIFIED = 0
+
+            # Uncompressed 16-bit signed little-endian samples (Linear PCM).
+            LINEAR16 = 1
+
+            # +FLAC+ (Free Lossless Audio
+            # Codec) is the recommended encoding because it is
+            # lossless--therefore recognition is not compromised--and
+            # requires only about half the bandwidth of +LINEAR16+. +FLAC+ stream
+            # encoding supports 16-bit and 24-bit samples, however, not all fields in
+            # +STREAMINFO+ are supported.
+            FLAC = 2
+
+            # 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
+            MULAW = 3
+
+            # Adaptive Multi-Rate Narrowband codec. +sample_rate_hertz+ must be 8000.
+            AMR = 4
+
+            # Adaptive Multi-Rate Wideband codec. +sample_rate_hertz+ must be 16000.
+            AMR_WB = 5
+
+            # Opus encoded audio frames in Ogg container
+            # ([OggOpus](https://wiki.xiph.org/OggOpus)).
+            # +sample_rate_hertz+ must be one of 8000, 12000, 16000, 24000, or 48000.
+            OGG_OPUS = 6
+
+            # Although the use of lossy encodings is not recommended, if a very low
+            # bitrate encoding is required, +OGG_OPUS+ is highly preferred over
+            # Speex encoding. The [Speex](https://speex.org/)  encoding supported by
+            # Cloud Speech API has a header byte in each block, as in MIME type
+            # +audio/x-speex-with-header-byte+.
+            # It is a variant of the RTP Speex encoding defined in
+            # [RFC 5574](https://tools.ietf.org/html/rfc5574).
+            # The stream is a sequence of blocks, one block per RTP packet. Each block
+            # starts with a byte containing the length of the block, in bytes, followed
+            # by one or more frames of Speex data, padded to an integral number of
+            # bytes (octets) as specified in RFC 5574. In other words, each RTP header
+            # is replaced with a single byte containing the block length. Only Speex
+            # wideband is supported. +sample_rate_hertz+ must be 16000.
+            SPEEX_WITH_HEADER_BYTE = 7
+          end
+        end
+
+        # Description of audio data to be recognized.
+        # @!attribute [rw] interaction_type
+        #   @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata::InteractionType]
+        #     The use case most closely describing the audio content to be recognized.
+        # @!attribute [rw] industry_naics_code_of_audio
+        #   @return [Integer]
+        #     The industry vertical to which this speech recognition request most
+        #     closely applies. This is most indicative of the topics contained
+        #     in the audio.  Use the 6-digit NAICS code to identify the industry
+        #     vertical - see https://www.naics.com/search/.
+        # @!attribute [rw] microphone_distance
+        #   @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata::MicrophoneDistance]
+        #     The audio type that most closely describes the audio being recognized.
+        # @!attribute [rw] original_media_type
+        #   @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata::OriginalMediaType]
+        #     The original media the speech was recorded on.
+        # @!attribute [rw] recording_device_type
+        #   @return [Google::Cloud::Speech::V1p1beta1::RecognitionMetadata::RecordingDeviceType]
+        #     The type of device the speech was recorded with.
+        # @!attribute [rw] recording_device_name
+        #   @return [String]
+        #     The device used to make the recording.  Examples 'Nexus 5X' or
+        #     'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
+        #     'Cardioid Microphone'.
+        # @!attribute [rw] original_mime_type
+        #   @return [String]
+        #     Mime type of the original audio file.  For example +audio/m4a+,
+        #     +audio/x-alaw-basic+, +audio/mp3+, +audio/3gpp+.
+        #     A list of possible audio mime types is maintained at
+        #     http://www.iana.org/assignments/media-types/media-types.xhtml#audio
+        # @!attribute [rw] obfuscated_id
+        #   @return [Integer]
+        #     Obfuscated (privacy-protected) ID of the user, to identify number of
+        #     unique users using the service.
+        # @!attribute [rw] audio_topic
+        #   @return [String]
+        #     Description of the content. Eg. "Recordings of federal supreme court
+        #     hearings from 2012".
+        class RecognitionMetadata
+          # Use case categories that the audio recognition request can be described
+          # by.
+          module InteractionType
+            # Use case is either unknown or is something other than one of the other
+            # values below.
+            INTERACTION_TYPE_UNSPECIFIED = 0
+
+            # Multiple people in a conversation or discussion. For example in a
+            # meeting with two or more people actively participating. Typically
+            # all the primary people speaking would be in the same room (if not,
+            # see PHONE_CALL)
+            DISCUSSION = 1
+
+            # One or more persons lecturing or presenting to others, mostly
+            # uninterrupted.
+            PRESENTATION = 2
+
+            # A phone-call or video-conference in which two or more people, who are
+            # not in the same room, are actively participating.
+            PHONE_CALL = 3
+
+            # A recorded message intended for another person to listen to.
+            VOICEMAIL = 4
+
+            # Professionally produced audio (eg. TV Show, Podcast).
+            PROFESSIONALLY_PRODUCED = 5
+
+            # Transcribe spoken questions and queries into text.
+            VOICE_SEARCH = 6
+
+            # Transcribe voice commands, such as for controlling a device.
+            VOICE_COMMAND = 7
+
+            # Transcribe speech to text to create a written document, such as a
+            # text-message, email or report.
+            DICTATION = 8
+          end
+
+          # Enumerates the types of capture settings describing an audio file.
+          module MicrophoneDistance
+            # Audio type is not known.
+            MICROPHONE_DISTANCE_UNSPECIFIED = 0
+
+            # The audio was captured from a closely placed microphone. Eg. phone,
+            # dictaphone, or handheld microphone. Generally if there speaker is within
+            # 1 meter of the microphone.
+            NEARFIELD = 1
+
+            # The speaker if within 3 meters of the microphone.
+            MIDFIELD = 2
+
+            # The speaker is more than 3 meters away from the microphone.
+            FARFIELD = 3
+          end
+
+          # The original media the speech was recorded on.
+          module OriginalMediaType
+            # Unknown original media type.
+            ORIGINAL_MEDIA_TYPE_UNSPECIFIED = 0
+
+            # The speech data is an audio recording.
+            AUDIO = 1
+
+            # The speech data originally recorded on a video.
+            VIDEO = 2
+          end
+
+          # The type of device the speech was recorded with.
+          module RecordingDeviceType
+            # The recording device is unknown.
+            RECORDING_DEVICE_TYPE_UNSPECIFIED = 0
+
+            # Speech was recorded on a smartphone.
+            SMARTPHONE = 1
+
+            # Speech was recorded using a personal computer or tablet.
+            PC = 2
+
+            # Speech was recorded over a phone line.
+            PHONE_LINE = 3
+
+            # Speech was recorded in a vehicle.
+            VEHICLE = 4
+
+            # Speech was recorded outdoors.
+            OTHER_OUTDOOR_DEVICE = 5
+
+            # Speech was recorded indoors.
+            OTHER_INDOOR_DEVICE = 6
+          end
+        end
+
+        # Provides "hints" to the speech recognizer to favor specific words and phrases
+        # in the results.
+        # @!attribute [rw] phrases
+        #   @return [Array<String>]
+        #     *Optional* A list of strings containing words and phrases "hints" so that
+        #     the speech recognition is more likely to recognize them. This can be used
+        #     to improve the accuracy for specific words and phrases, for example, if
+        #     specific commands are typically spoken by the user. This can also be used
+        #     to add additional words to the vocabulary of the recognizer. See
+        #     [usage limits](https://cloud.google.com/speech/limits#content).
+        class SpeechContext; end
+
+        # Contains audio data in the encoding specified in the +RecognitionConfig+.
+        # Either +content+ or +uri+ must be supplied. Supplying both or neither
+        # returns {Google::Rpc::Code::INVALID_ARGUMENT}. See
+        # [audio limits](https://cloud.google.com/speech/limits#content).
+        # @!attribute [rw] content
+        #   @return [String]
+        #     The audio data bytes encoded as specified in
+        #     +RecognitionConfig+. Note: as with all bytes fields, protobuffers use a
+        #     pure binary representation, whereas JSON representations use base64.
+        # @!attribute [rw] uri
+        #   @return [String]
+        #     URI that points to a file that contains audio data bytes as specified in
+        #     +RecognitionConfig+. Currently, only Google Cloud Storage URIs are
+        #     supported, which must be specified in the following format:
+        #     +gs://bucket_name/object_name+ (other URI formats return
+        #     {Google::Rpc::Code::INVALID_ARGUMENT}). For more information, see
+        #     [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
+        class RecognitionAudio; end
+
+        # The only message returned to the client by the +Recognize+ method. It
+        # contains the result as zero or more sequential +SpeechRecognitionResult+
+        # messages.
+        # @!attribute [rw] results
+        #   @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechRecognitionResult>]
+        #     Output only. Sequential list of transcription results corresponding to
+        #     sequential portions of audio.
+        class RecognizeResponse; end
+
+        # The only message returned to the client by the +LongRunningRecognize+ method.
+        # It contains the result as zero or more sequential +SpeechRecognitionResult+
+        # messages. It is included in the +result.response+ field of the +Operation+
+        # returned by the +GetOperation+ call of the +google::longrunning::Operations+
+        # service.
+        # @!attribute [rw] results
+        #   @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechRecognitionResult>]
+        #     Output only. Sequential list of transcription results corresponding to
+        #     sequential portions of audio.
+        class LongRunningRecognizeResponse; end
+
+        # Describes the progress of a long-running +LongRunningRecognize+ call. It is
+        # included in the +metadata+ field of the +Operation+ returned by the
+        # +GetOperation+ call of the +google::longrunning::Operations+ service.
+        # @!attribute [rw] progress_percent
+        #   @return [Integer]
+        #     Approximate percentage of audio processed thus far. Guaranteed to be 100
+        #     when the audio is fully processed and the results are available.
+        # @!attribute [rw] start_time
+        #   @return [Google::Protobuf::Timestamp]
+        #     Time when the request was received.
+        # @!attribute [rw] last_update_time
+        #   @return [Google::Protobuf::Timestamp]
+        #     Time of the most recent processing update.
+        class LongRunningRecognizeMetadata; end
+
+        # +StreamingRecognizeResponse+ is the only message returned to the client by
+        # +StreamingRecognize+. A series of zero or more +StreamingRecognizeResponse+
+        # messages are streamed back to the client. If there is no recognizable
+        # audio, and +single_utterance+ is set to false, then no messages are streamed
+        # back to the client.
+        #
+        # Here's an example of a series of ten +StreamingRecognizeResponse+s that might
+        # be returned while processing audio:
+        #
+        # 1. results { alternatives { transcript: "tube" } stability: 0.01 }
+        #
+        # 2. results { alternatives { transcript: "to be a" } stability: 0.01 }
+        #
+        # 3. results { alternatives { transcript: "to be" } stability: 0.9 }
+        #    results { alternatives { transcript: " or not to be" } stability: 0.01 }
+        #
+        # 4. results { alternatives { transcript: "to be or not to be"
+        #                             confidence: 0.92 }
+        #              alternatives { transcript: "to bee or not to bee" }
+        #              is_final: true }
+        #
+        # 5. results { alternatives { transcript: " that's" } stability: 0.01 }
+        #
+        # 6. results { alternatives { transcript: " that is" } stability: 0.9 }
+        #    results { alternatives { transcript: " the question" } stability: 0.01 }
+        #
+        # 7. results { alternatives { transcript: " that is the question"
+        #                             confidence: 0.98 }
+        #              alternatives { transcript: " that was the question" }
+        #              is_final: true }
+        #
+        # Notes:
+        #
+        # * Only two of the above responses #4 and #7 contain final results; they are
+        #   indicated by +is_final: true+. Concatenating these together generates the
+        #   full transcript: "to be or not to be that is the question".
+        #
+        # * The others contain interim +results+. #3 and #6 contain two interim
+        #   +results+: the first portion has a high stability and is less likely to
+        #   change; the second portion has a low stability and is very likely to
+        #   change. A UI designer might choose to show only high stability +results+.
+        #
+        # * The specific +stability+ and +confidence+ values shown above are only for
+        #   illustrative purposes. Actual values may vary.
+        #
+        # * In each response, only one of these fields will be set:
+        #   +error+,
+        #   +speech_event_type+, or
+        #   one or more (repeated) +results+.
+        # @!attribute [rw] error
+        #   @return [Google::Rpc::Status]
+        #     Output only. If set, returns a {Google::Rpc::Status} message that
+        #     specifies the error for the operation.
+        # @!attribute [rw] results
+        #   @return [Array<Google::Cloud::Speech::V1p1beta1::StreamingRecognitionResult>]
+        #     Output only. This repeated list contains zero or more results that
+        #     correspond to consecutive portions of the audio currently being processed.
+        #     It contains zero or one +is_final=true+ result (the newly settled portion),
+        #     followed by zero or more +is_final=false+ results (the interim results).
+        # @!attribute [rw] speech_event_type
+        #   @return [Google::Cloud::Speech::V1p1beta1::StreamingRecognizeResponse::SpeechEventType]
+        #     Output only. Indicates the type of speech event.
+        class StreamingRecognizeResponse
+          # Indicates the type of speech event.
+          module SpeechEventType
+            # No speech event specified.
+            SPEECH_EVENT_UNSPECIFIED = 0
+
+            # This event indicates that the server has detected the end of the user's
+            # speech utterance and expects no additional speech. Therefore, the server
+            # will not process additional audio (although it may subsequently return
+            # additional results). The client should stop sending additional audio
+            # data, half-close the gRPC connection, and wait for any additional results
+            # until the server closes the gRPC connection. This event is only sent if
+            # +single_utterance+ was set to +true+, and is not used otherwise.
+            END_OF_SINGLE_UTTERANCE = 1
+          end
+        end
+
+        # A streaming speech recognition result corresponding to a portion of the audio
+        # that is currently being processed.
+        # @!attribute [rw] alternatives
+        #   @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechRecognitionAlternative>]
+        #     Output only. May contain one or more recognition hypotheses (up to the
+        #     maximum specified in +max_alternatives+).
+        #     These alternatives are ordered in terms of accuracy, with the top (first)
+        #     alternative being the most probable, as ranked by the recognizer.
+        # @!attribute [rw] is_final
+        #   @return [true, false]
+        #     Output only. If +false+, this +StreamingRecognitionResult+ represents an
+        #     interim result that may change. If +true+, this is the final time the
+        #     speech service will return this particular +StreamingRecognitionResult+,
+        #     the recognizer will not return any further hypotheses for this portion of
+        #     the transcript and corresponding audio.
+        # @!attribute [rw] stability
+        #   @return [Float]
+        #     Output only. An estimate of the likelihood that the recognizer will not
+        #     change its guess about this interim result. Values range from 0.0
+        #     (completely unstable) to 1.0 (completely stable).
+        #     This field is only provided for interim results (+is_final=false+).
+        #     The default of 0.0 is a sentinel value indicating +stability+ was not set.
+        class StreamingRecognitionResult; end
+
+        # A speech recognition result corresponding to a portion of the audio.
+        # @!attribute [rw] alternatives
+        #   @return [Array<Google::Cloud::Speech::V1p1beta1::SpeechRecognitionAlternative>]
+        #     Output only. May contain one or more recognition hypotheses (up to the
+        #     maximum specified in +max_alternatives+).
+        #     These alternatives are ordered in terms of accuracy, with the top (first)
+        #     alternative being the most probable, as ranked by the recognizer.
+        class SpeechRecognitionResult; end
+
+        # Alternative hypotheses (a.k.a. n-best list).
+        # @!attribute [rw] transcript
+        #   @return [String]
+        #     Output only. Transcript text representing the words that the user spoke.
+        # @!attribute [rw] confidence
+        #   @return [Float]
+        #     Output only. The confidence estimate between 0.0 and 1.0. A higher number
+        #     indicates an estimated greater likelihood that the recognized words are
+        #     correct. This field is set only for the top alternative of a non-streaming
+        #     result or, of a streaming result where +is_final=true+.
+        #     This field is not guaranteed to be accurate and users should not rely on it
+        #     to be always provided.
+        #     The default of 0.0 is a sentinel value indicating +confidence+ was not set.
+        # @!attribute [rw] words
+        #   @return [Array<Google::Cloud::Speech::V1p1beta1::WordInfo>]
+        #     Output only. A list of word-specific information for each recognized word.
+        #     Note: When enable_speaker_diarization is true, you will see all the words
+        #     from the beginning of the audio.
+        class SpeechRecognitionAlternative; end
+
+        # Word-specific information for recognized words.
+        # @!attribute [rw] start_time
+        #   @return [Google::Protobuf::Duration]
+        #     Output only. Time offset relative to the beginning of the audio,
+        #     and corresponding to the start of the spoken word.
+        #     This field is only set if +enable_word_time_offsets=true+ and only
+        #     in the top hypothesis.
+        #     This is an experimental feature and the accuracy of the time offset can
+        #     vary.
+        # @!attribute [rw] end_time
+        #   @return [Google::Protobuf::Duration]
+        #     Output only. Time offset relative to the beginning of the audio,
+        #     and corresponding to the end of the spoken word.
+        #     This field is only set if +enable_word_time_offsets=true+ and only
+        #     in the top hypothesis.
+        #     This is an experimental feature and the accuracy of the time offset can
+        #     vary.
+        # @!attribute [rw] word
+        #   @return [String]
+        #     Output only. The word corresponding to this set of information.
+        class WordInfo; end
+      end
+    end
+  end
+end