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Excise speech v1_1beta1. (#494) 

They have the v1p1beta1 proto namespace; this can confuse protoc.
These files are already removed from google3, but deletes do not sync.
This commit is contained in:
Luke Sneeringer 2018-02-07 11:32:31 -08:00 committed by Jacob Geiger
parent 7c643ad10a
commit 16a1933a4c
4 changed files with 0 additions and 748 deletions
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20
google/cloud/speech/cloud_speech_v1_1beta1.yaml
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# Google Cloud Speech API service configuration
type: google.api.Service
config_version: 3
name: speech.googleapis.com
title: Google Cloud Speech API
documentation:
summary:
Google Cloud Speech API.
apis:
- name: google.cloud.speech.v1p1beta1.Speech
authentication:
rules:
- selector: '*'
oauth:
canonical_scopes: https://www.googleapis.com/auth/cloud-platform

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google/cloud/speech/speech_v1_1beta1.yaml
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# Google Cloud Speech API service configuration
type: google.api.Service
config_version: 3
name: speech.googleapis.com
title: Google Cloud Speech API
documentation:
summary:
Google Cloud Speech API.
apis:
- name: google.cloud.speech.v1p1beta1.Speech
authentication:
rules:
- selector: '*'
oauth:
canonical_scopes: https://www.googleapis.com/auth/cloud-platform
http:
rules:
- selector: google.longrunning.Operations.GetOperation
get: '/v1p1beta1/operations/{name=*}'

607
google/cloud/speech/v1_1beta1/cloud_speech.proto
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// Copyright 2017 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.
syntax = "proto3";
package google.cloud.speech.v1p1beta1;
import "google/api/annotations.proto";
import "google/longrunning/operations.proto";
import "google/protobuf/duration.proto";
import "google/protobuf/timestamp.proto";
import "google/rpc/status.proto";
option cc_enable_arenas = true;
option go_package = "google.golang.org/genproto/googleapis/cloud/speech/v1p1beta1;speech";
option java_multiple_files = true;
option java_outer_classname = "SpeechProto";
option java_package = "com.google.cloud.speech.v1p1beta1";
// Service that implements Google Cloud Speech API.
service Speech {
// Performs synchronous speech recognition: receive results after all audio
// has been sent and processed.
rpc Recognize(RecognizeRequest) returns (RecognizeResponse) {
option (google.api.http) = { post: "/v1p1beta1/speech:recognize" body: "*" };
}
// 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) returns (google.longrunning.Operation) {
option (google.api.http) = { post: "/v1p1beta1/speech:longrunningrecognize" body: "*" };
}
// Performs bidirectional streaming speech recognition: receive results while
// sending audio. This method is only available via the gRPC API (not REST).
rpc StreamingRecognize(stream StreamingRecognizeRequest) returns (stream StreamingRecognizeResponse);
}
// The top-level message sent by the client for the `Recognize` method.
message RecognizeRequest {
// *Required* Provides information to the recognizer that specifies how to
// process the request.
RecognitionConfig config = 1;
// *Required* The audio data to be recognized.
RecognitionAudio audio = 2;
}
// The top-level message sent by the client for the `LongRunningRecognize`
// method.
message LongRunningRecognizeRequest {
// *Required* Provides information to the recognizer that specifies how to
// process the request.
RecognitionConfig config = 1;
// *Required* The audio data to be recognized.
RecognitionAudio audio = 2;
}
// 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.
message StreamingRecognizeRequest {
// The streaming request, which is either a streaming config or audio content.
oneof streaming_request {
// Provides information to the recognizer that specifies how to process the
// request. The first `StreamingRecognizeRequest` message must contain a
// `streaming_config` message.
StreamingRecognitionConfig streaming_config = 1;
// The audio data to be recognized. Sequential chunks of audio data are sent
// in sequential `StreamingRecognizeRequest` messages. The first
// `StreamingRecognizeRequest` message must not contain `audio_content` data
// and all subsequent `StreamingRecognizeRequest` messages must contain
// `audio_content` data. The audio bytes must be encoded as specified in
// `RecognitionConfig`. Note: as with all bytes fields, protobuffers use a
// pure binary representation (not base64). See
// [audio limits](https://cloud.google.com/speech/limits#content).
bytes audio_content = 2;
}
}
// Provides information to the recognizer that specifies how to process the
// request.
message StreamingRecognitionConfig {
// *Required* Provides information to the recognizer that specifies how to
// process the request.
RecognitionConfig config = 1;
// *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`.
bool single_utterance = 2;
// *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.
bool interim_results = 3;
}
// Provides information to the recognizer that specifies how to process the
// request.
message RecognitionConfig {
// Audio encoding of the data sent in the audio message. All encodings support
// only 1 channel (mono) audio. Only `FLAC` includes a header that describes
// the bytes of audio that follow the header. The other encodings are raw
// audio bytes with no header.
//
// For best results, the audio source should be captured and transmitted using
// a lossless encoding (`FLAC` or `LINEAR16`). Recognition accuracy may be
// reduced if lossy codecs, which include the other codecs listed in
// this section, are used to capture or transmit the audio, particularly if
// background noise is present.
enum AudioEncoding {
// Not specified. Will return result [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT].
ENCODING_UNSPECIFIED = 0;
// Uncompressed 16-bit signed little-endian samples (Linear PCM).
LINEAR16 = 1;
// [`FLAC`](https://xiph.org/flac/documentation.html) (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 16000.
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;
}
// *Required* Encoding of audio data sent in all `RecognitionAudio` messages.
AudioEncoding encoding = 1;
// *Required* 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).
int32 sample_rate_hertz = 2;
// *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.
string language_code = 3;
// *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.
int32 max_alternatives = 4;
// *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.
bool profanity_filter = 5;
// *Optional* A means to provide context to assist the speech recognition.
repeated SpeechContext speech_contexts = 6;
// *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`.
bool enable_word_time_offsets = 8;
// *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."
bool enable_automatic_punctuation = 11;
// *Optional* Metadata regarding this request.
RecognitionMetadata metadata = 9;
}
// Description of audio data to be recognized.
message RecognitionMetadata {
// Use case categories that the audio recognition request can be described
// by.
enum InteractionType {
// Use case is either unknown or is something other than one of the other
// values below.
INTERACTION_TYPE_UNSPECIFIED = 0;
// Multiple people in a conversation or discussion. For example in a
// meeting with two or more people actively participating. Typically
// all the primary people speaking would be in the same room (if not,
// see PHONE_CALL)
DISCUSSION = 1;
// One or more persons lecturing or presenting to others, mostly
// uninterrupted.
PRESENTATION = 2;
// A phone-call or video-conference in which two or more people, who are
// not in the same room, are actively participating.
PHONE_CALL = 3;
// A recorded message intended for another person to listen to.
VOICEMAIL = 4;
// Professionally produced audio (eg. TV Show, Podcast).
PROFESSIONALLY_PRODUCED = 5;
// Transcribe spoken questions and queries into text.
VOICE_SEARCH = 6;
// Transcribe voice commands, such as for controlling a device.
VOICE_COMMAND = 7;
// Transcribe speech to text to create a written document, such as a
// text-message, email or report.
DICTATION = 8;
}
// Enumerates the types of capture settings describing an audio file.
enum MicrophoneDistance {
// Audio type is not known.
MICROPHONE_DISTANCE_UNSPECIFIED = 0;
// The audio was captured from a closely placed microphone. Eg. phone,
// dictaphone, or handheld microphone. Generally if there speaker is within
// 1 meter of the microphone.
NEARFIELD = 1;
// The speaker if within 3 meters of the microphone.
MIDFIELD = 2;
// The speaker is more than 3 meters away from the microphone.
FARFIELD = 3;
}
// The original media the speech was recorded on.
enum OriginalMediaType {
// Unknown original media type.
ORIGINAL_MEDIA_TYPE_UNSPECIFIED = 0;
// The speech data is an audio recording.
AUDIO = 1;
// The speech data originally recorded on a video.
VIDEO = 2;
}
// How many speakers expected in the speech to be recognized.
enum NumberOfSpeakers {
// Unknown number of persons speaking.
NUMBER_OF_SPEAKERS_UNSPECIFIED = 0;
// Only one person is the prominent speaker (ignore background voices).
ONE_SPEAKER = 1;
// Two people are the prominent speakers (transcript should focus
// on the two most prominent speakers).
TWO_SPEAKERS = 2;
// Transcribe all voices.
MULTIPLE_SPEAKERS = 3;
}
// The type of device the speech was recorded with.
enum RecordingDeviceType {
// The recording device is unknown.
RECORDING_DEVICE_TYPE_UNSPECIFIED = 0;
// Speech was recorded on a smartphone.
SMARTPHONE = 1;
// Speech was recorded using a personal computer or tablet.
PC = 2;
// Speech was recorded over a phone line.
PHONE_LINE = 3;
// Speech was recorded in a vehicle.
VEHICLE = 4;
// Speech was recorded outdoors.
OTHER_OUTDOOR_DEVICE = 5;
// Speech was recorded indoors.
OTHER_INDOOR_DEVICE = 6;
}
// The use case most closely describing the audio content to be recognized.
InteractionType interaction_type = 1;
// The industry vertical to which this speech recognition request most
// closely applies. This is most indicative of the topics contained
// in the audio. Use the 6-digit NAICS code to identify the industry
// vertical - see https://www.naics.com/search/.
uint32 industry_naics_code_of_audio = 3;
// The audio type that most closely describes the audio being recognized.
MicrophoneDistance microphone_distance = 4;
// The original media the speech was recorded on.
OriginalMediaType original_media_type = 5;
// How many people are speaking prominently in the audio and expected to be
// recognized.
NumberOfSpeakers number_of_speakers = 6;
// The type of device the speech was recorded with.
RecordingDeviceType recording_device_type = 7;
// The device used to make the recording. Examples 'Nexus 5X' or
// 'Polycom SoundStation IP 6000' or 'POTS' or 'VoIP' or
// 'Cardioid Microphone'.
string recording_device_name = 8;
// Mime type of the original audio file. For example `audio/m4a`,
// `audio/x-alaw-basic`, `audio/mp3`, `audio/3gpp`.
// A list of possible audio mime types is maintained at
// http://www.iana.org/assignments/media-types/media-types.xhtml#audio
string original_mime_type = 9;
// Obfuscated (privacy-protected) ID of the user, to identify number of
// unique users using the service.
int64 obfuscated_id = 10;
// Description of the content. Eg. "Recordings of federal supreme court
// hearings from 2012".
string audio_topic = 11;
}
// Provides "hints" to the speech recognizer to favor specific words and phrases
// in the results.
message SpeechContext {
// *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).
repeated string phrases = 1;
}
// Contains audio data in the encoding specified in the `RecognitionConfig`.
// Either `content` or `uri` must be supplied. Supplying both or neither
// returns [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT]. See
// [audio limits](https://cloud.google.com/speech/limits#content).
message RecognitionAudio {
// The audio source, which is either inline content or a GCS uri.
oneof audio_source {
// 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.
bytes content = 1;
// 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][google.rpc.Code.INVALID_ARGUMENT]). For more information, see
// [Request URIs](https://cloud.google.com/storage/docs/reference-uris).
string uri = 2;
}
}
// The only message returned to the client by the `Recognize` method. It
// contains the result as zero or more sequential `SpeechRecognitionResult`
// messages.
message RecognizeResponse {
// *Output-only* Sequential list of transcription results corresponding to
// sequential portions of audio.
repeated SpeechRecognitionResult results = 2;
}
// The only message returned to the client by the `LongRunningRecognize` method.
// It contains the result as zero or more sequential `SpeechRecognitionResult`
// messages. It is included in the `result.response` field of the `Operation`
// returned by the `GetOperation` call of the `google::longrunning::Operations`
// service.
message LongRunningRecognizeResponse {
// *Output-only* Sequential list of transcription results corresponding to
// sequential portions of audio.
repeated SpeechRecognitionResult results = 2;
}
// Describes the progress of a long-running `LongRunningRecognize` call. It is
// included in the `metadata` field of the `Operation` returned by the
// `GetOperation` call of the `google::longrunning::Operations` service.
message LongRunningRecognizeMetadata {
// Approximate percentage of audio processed thus far. Guaranteed to be 100
// when the audio is fully processed and the results are available.
int32 progress_percent = 1;
// Time when the request was received.
google.protobuf.Timestamp start_time = 2;
// Time of the most recent processing update.
google.protobuf.Timestamp last_update_time = 3;
}
// `StreamingRecognizeResponse` is the only message returned to the client by
// `StreamingRecognize`. A series of zero or more `StreamingRecognizeResponse`
// messages are streamed back to the client. If there is no recognizable
// audio, and `single_utterance` is set to false, then no messages are streamed
// back to the client.
//
// Here's an example of a series of ten `StreamingRecognizeResponse`s that might
// be returned while processing audio:
//
// 1. results { alternatives { transcript: "tube" } stability: 0.01 }
//
// 2. results { alternatives { transcript: "to be a" } stability: 0.01 }
//
// 3. results { alternatives { transcript: "to be" } stability: 0.9 }
// results { alternatives { transcript: " or not to be" } stability: 0.01 }
//
// 4. results { alternatives { transcript: "to be or not to be"
// confidence: 0.92 }
// alternatives { transcript: "to bee or not to bee" }
// is_final: true }
//
// 5. results { alternatives { transcript: " that's" } stability: 0.01 }
//
// 6. results { alternatives { transcript: " that is" } stability: 0.9 }
// results { alternatives { transcript: " the question" } stability: 0.01 }
//
// 7. results { alternatives { transcript: " that is the question"
// confidence: 0.98 }
// alternatives { transcript: " that was the question" }
// is_final: true }
//
// Notes:
//
// - Only two of the above responses #4 and #7 contain final results; they are
// indicated by `is_final: true`. Concatenating these together generates the
// full transcript: "to be or not to be that is the question".
//
// - The others contain interim `results`. #3 and #6 contain two interim
// `results`: the first portion has a high stability and is less likely to
// change; the second portion has a low stability and is very likely to
// change. A UI designer might choose to show only high stability `results`.
//
// - The specific `stability` and `confidence` values shown above are only for
// illustrative purposes. Actual values may vary.
//
// - In each response, only one of these fields will be set:
// `error`,
// `speech_event_type`, or
// one or more (repeated) `results`.
message StreamingRecognizeResponse {
// Indicates the type of speech event.
enum SpeechEventType {
// No speech event specified.
SPEECH_EVENT_UNSPECIFIED = 0;
// This event indicates that the server has detected the end of the user's
// speech utterance and expects no additional speech. Therefore, the server
// will not process additional audio (although it may subsequently return
// additional results). The client should stop sending additional audio
// data, half-close the gRPC connection, and wait for any additional results
// until the server closes the gRPC connection. This event is only sent if
// `single_utterance` was set to `true`, and is not used otherwise.
END_OF_SINGLE_UTTERANCE = 1;
}
// *Output-only* If set, returns a [google.rpc.Status][google.rpc.Status] message that
// specifies the error for the operation.
google.rpc.Status error = 1;
// *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 more `is_final=false` results followed by zero or one
// `is_final=true` result (the newly settled portion).
repeated StreamingRecognitionResult results = 2;
// *Output-only* Indicates the type of speech event.
SpeechEventType speech_event_type = 4;
}
// A streaming speech recognition result corresponding to a portion of the audio
// that is currently being processed.
message StreamingRecognitionResult {
// *Output-only* May contain one or more recognition hypotheses (up to the
// maximum specified in `max_alternatives`).
repeated SpeechRecognitionAlternative alternatives = 1;
// *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.
bool is_final = 2;
// *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.
float stability = 3;
}
// A speech recognition result corresponding to a portion of the audio.
message SpeechRecognitionResult {
// *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.
repeated SpeechRecognitionAlternative alternatives = 1;
}
// Alternative hypotheses (a.k.a. n-best list).
message SpeechRecognitionAlternative {
// *Output-only* Transcript text representing the words that the user spoke.
string transcript = 1;
// *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 typically provided only for the top hypothesis, and
// only for `is_final=true` results. Clients should not rely on the
// `confidence` field as it is not guaranteed to be accurate, or even set, in
// any of the results.
// The default of 0.0 is a sentinel value indicating `confidence` was not set.
float confidence = 2;
// *Output-only* A list of word-specific information for each recognized word.
repeated WordInfo words = 3;
}
// Word-specific information for recognized words. Word information is only
// included in the response when certain request parameters are set, such
// as `enable_word_time_offsets`.
message WordInfo {
// *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.
google.protobuf.Duration start_time = 1;
// *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.
google.protobuf.Duration end_time = 2;
// *Output-only* The word corresponding to this set of information.
string word = 3;
}

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google/cloud/speech/v1_1beta1/cloud_speech_gapic.yaml
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type: com.google.api.codegen.ConfigProto
language_settings:
java:
package_name: com.google.cloud.speech.v1_1beta1
python:
package_name: google.cloud.gapic.speech.v1_1beta1
go:
package_name: cloud.google.com/go/speech/apiv1_1beta1
csharp:
package_name: Google.Cloud.Speech.V1_1Beta1
ruby:
package_name: Google::Cloud::Speech::V1_1beta1
php:
package_name: Google\Cloud\Speech\V1_1Beta1
nodejs:
package_name: speech.v1_1beta1
domain_layer_location: google-cloud
license_header:
copyright_file: copyright-google.txt
license_file: license-header-apache-2.0.txt
interfaces:
- name: google.cloud.speech.v1p1beta1.Speech
smoke_test:
method: Recognize
init_fields:
- config.language_code="en-US"
- config.sample_rate_hertz=44100
- config.encoding=FLAC
- audio.uri="gs://gapic-toolkit/hello.flac"
collections: []
retry_codes_def:
- name: idempotent
retry_codes:
- UNAVAILABLE
- DEADLINE_EXCEEDED
- name: non_idempotent
retry_codes: []
retry_params_def:
- name: default
initial_retry_delay_millis: 100
retry_delay_multiplier: 1.3
max_retry_delay_millis: 60000
initial_rpc_timeout_millis: 190000
rpc_timeout_multiplier: 1
max_rpc_timeout_millis: 190000
total_timeout_millis: 600000
methods:
- name: Recognize
flattening:
groups:
- parameters:
- config
- audio
required_fields:
- config
- audio
sample_code_init_fields:
- config.encoding=FLAC
- config.sample_rate_hertz=44100
- config.language_code="en-US"
- audio.uri=gs://bucket_name/file_name.flac
request_object_method: true
retry_codes_name: idempotent
retry_params_name: default
timeout_millis: 190000
- name: LongRunningRecognize
flattening:
groups:
- parameters:
- config
- audio
required_fields:
- config
- audio
sample_code_init_fields:
- config.encoding=FLAC
- config.sample_rate_hertz=44100
- config.language_code="en-US"
- audio.uri=gs://bucket_name/file_name.flac
request_object_method: true
retry_codes_name: non_idempotent
retry_params_name: default
timeout_millis: 60000
long_running:
return_type: google.cloud.speech.v1p1beta1.LongRunningRecognizeResponse
metadata_type: google.cloud.speech.v1p1beta1.LongRunningRecognizeMetadata
initial_poll_delay_millis: 20000
poll_delay_multiplier: 1.5
max_poll_delay_millis: 45000
total_poll_timeout_millis: 86400000
- name: StreamingRecognize
request_object_method: false
retry_codes_name: idempotent
retry_params_name: default
timeout_millis: 190000