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// Copyright 2021 Google LLC
//
// Licensed under the Apache License, Version 2.0 (the "License");
// you may not use this file except in compliance with the License.
// You may obtain a copy of the License at
//
// http://www.apache.org/licenses/LICENSE-2.0
//
// Unless required by applicable law or agreed to in writing, software
// distributed under the License is distributed on an "AS IS" BASIS,
// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
// See the License for the specific language governing permissions and
// limitations under the License.
syntax = "proto3";
package google.cloud.mediatranslation.v1beta1;
import "google/api/field_behavior.proto";
import "google/rpc/status.proto";
import "google/api/client.proto";
option cc_enable_arenas = true;
option go_package = "google.golang.org/genproto/googleapis/cloud/mediatranslation/v1beta1;mediatranslation";
option java_multiple_files = true;
option java_outer_classname = "MediaTranslationProto";
option java_package = "com.google.cloud.mediatranslation.v1beta1";
option csharp_namespace = "Google.Cloud.MediaTranslation.V1Beta1";
option ruby_package = "Google::Cloud::MediaTranslation::V1beta1";
option php_namespace = "Google\\Cloud\\MediaTranslation\\V1beta1";
// Provides translation from/to media types.
service SpeechTranslationService {
option (google.api.default_host) = "mediatranslation.googleapis.com";
option (google.api.oauth_scopes) = "https://www.googleapis.com/auth/cloud-platform";
// Performs bidirectional streaming speech translation: receive results while
// sending audio. This method is only available via the gRPC API (not REST).
rpc StreamingTranslateSpeech(stream StreamingTranslateSpeechRequest) returns (stream StreamingTranslateSpeechResponse) {
}
}
// Provides information to the speech translation that specifies how to process
// the request.
message TranslateSpeechConfig {
// Required. Encoding of audio data.
// Supported formats:
//
// - `linear16`
//
// Uncompressed 16-bit signed little-endian samples (Linear PCM).
//
// - `flac`
//
// `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`.
//
// - `mulaw`
//
// 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
//
// - `amr`
//
// Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
//
// - `amr-wb`
//
// Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
//
// - `ogg-opus`
//
// Opus encoded audio frames in [Ogg](https://wikipedia.org/wiki/Ogg)
// container. `sample_rate_hertz` must be one of 8000, 12000, 16000, 24000,
// or 48000.
//
// - `mp3`
//
// MP3 audio. Support all standard MP3 bitrates (which range from 32-320
// kbps). When using this encoding, `sample_rate_hertz` has to match the
// sample rate of the file being used.
string audio_encoding = 1 [(google.api.field_behavior) = REQUIRED];
// Required. Source language code (BCP-47) of the input audio.
string source_language_code = 2 [(google.api.field_behavior) = REQUIRED];
// Required. Target language code (BCP-47) of the output.
string target_language_code = 3 [(google.api.field_behavior) = REQUIRED];
// Optional. Sample rate in Hertz of the audio data. 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 = 4 [(google.api.field_behavior) = OPTIONAL];
// Optional. `google-provided-model/video` and
// `google-provided-model/enhanced-phone-call` are premium models.
// `google-provided-model/phone-call` is not premium model.
string model = 5 [(google.api.field_behavior) = OPTIONAL];
}
// Config used for streaming translation.
message StreamingTranslateSpeechConfig {
// Required. The common config for all the following audio contents.
TranslateSpeechConfig audio_config = 1 [(google.api.field_behavior) = REQUIRED];
// Optional. If `false` or omitted, the system performs
// continuous translation (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
// `StreamingTranslateSpeechResult`s with the `is_final` flag set to `true`.
//
// If `true`, the speech translator 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 translation.
// When the client receives 'END_OF_SINGLE_UTTERANCE' event, the client should
// stop sending the requests. However, clients should keep receiving remaining
// responses until the stream is terminated. To construct the complete
// sentence in a streaming way, one should override (if 'is_final' of previous
// response is false), or append (if 'is_final' of previous response is true).
bool single_utterance = 2 [(google.api.field_behavior) = OPTIONAL];
}
// The top-level message sent by the client for the `StreamingTranslateSpeech`
// method. Multiple `StreamingTranslateSpeechRequest` messages are sent. The
// first message must contain a `streaming_config` message and must not contain
// `audio_content` data. All subsequent messages must contain `audio_content`
// data and must not contain a `streaming_config` message.
message StreamingTranslateSpeechRequest {
// The streaming request, which is either a streaming config or content.
oneof streaming_request {
// Provides information to the recognizer that specifies how to process the
// request. The first `StreamingTranslateSpeechRequest` message must contain
// a `streaming_config` message.
StreamingTranslateSpeechConfig streaming_config = 1;
// The audio data to be translated. Sequential chunks of audio data are sent
// in sequential `StreamingTranslateSpeechRequest` messages. The first
// `StreamingTranslateSpeechRequest` message must not contain
// `audio_content` data and all subsequent `StreamingTranslateSpeechRequest`
// messages must contain `audio_content` data. The audio bytes must be
// encoded as specified in `StreamingTranslateSpeechConfig`. Note: as with
// all bytes fields, protobuffers use a pure binary representation (not
// base64).
bytes audio_content = 2;
}
}
// A streaming speech translation result corresponding to a portion of the audio
// that is currently being processed.
message StreamingTranslateSpeechResult {
// Text translation result.
message TextTranslationResult {
// Output only. The translated sentence.
string translation = 1 [(google.api.field_behavior) = OUTPUT_ONLY];
// Output only. If `false`, this `StreamingTranslateSpeechResult` represents
// an interim result that may change. If `true`, this is the final time the
// translation service will return this particular
// `StreamingTranslateSpeechResult`, the streaming translator will not
// return any further hypotheses for this portion of the transcript and
// corresponding audio.
bool is_final = 2 [(google.api.field_behavior) = OUTPUT_ONLY];
}
// Translation result.
oneof result {
// Text translation result.
TextTranslationResult text_translation_result = 1;
}
}
// A streaming speech translation response corresponding to a portion of
// the audio currently processed.
message StreamingTranslateSpeechResponse {
// Indicates the type of speech event.
enum SpeechEventType {
// No speech event specified.
SPEECH_EVENT_TYPE_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). When the client receives 'END_OF_SINGLE_UTTERANCE'
// event, the client should stop sending the requests. However, clients
// should keep receiving remaining responses until the stream is terminated.
// To construct the complete sentence in a streaming way, one should
// override (if 'is_final' of previous response is false), or append (if
// 'is_final' of previous response is true). 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 [(google.api.field_behavior) = OUTPUT_ONLY];
// Output only. The translation result that is currently being processed (is_final could be
// true or false).
StreamingTranslateSpeechResult result = 2 [(google.api.field_behavior) = OUTPUT_ONLY];
// Output only. Indicates the type of speech event.
SpeechEventType speech_event_type = 3 [(google.api.field_behavior) = OUTPUT_ONLY];
}
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