/// Provides information to the speech translation that specifies how to process /// the request. #[derive(Clone, PartialEq, ::prost::Message)] pub struct 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 \[Ogg\]() /// 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. #[prost(string, tag = "1")] pub audio_encoding: ::prost::alloc::string::String, /// Required. Source language code (BCP-47) of the input audio. #[prost(string, tag = "2")] pub source_language_code: ::prost::alloc::string::String, /// Required. Target language code (BCP-47) of the output. #[prost(string, tag = "3")] pub target_language_code: ::prost::alloc::string::String, /// 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). #[prost(int32, tag = "4")] pub sample_rate_hertz: i32, /// Optional. `google-provided-model/video` and /// `google-provided-model/enhanced-phone-call` are premium models. /// `google-provided-model/phone-call` is not premium model. #[prost(string, tag = "5")] pub model: ::prost::alloc::string::String, } /// Config used for streaming translation. #[derive(Clone, PartialEq, ::prost::Message)] pub struct StreamingTranslateSpeechConfig { /// Required. The common config for all the following audio contents. #[prost(message, optional, tag = "1")] pub audio_config: ::core::option::Option, /// 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). #[prost(bool, tag = "2")] pub single_utterance: bool, } /// 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. #[derive(Clone, PartialEq, ::prost::Message)] pub struct StreamingTranslateSpeechRequest { /// The streaming request, which is either a streaming config or content. #[prost(oneof = "streaming_translate_speech_request::StreamingRequest", tags = "1, 2")] pub streaming_request: ::core::option::Option, } /// Nested message and enum types in `StreamingTranslateSpeechRequest`. pub mod streaming_translate_speech_request { /// The streaming request, which is either a streaming config or content. #[derive(Clone, PartialEq, ::prost::Oneof)] pub enum StreamingRequest { /// Provides information to the recognizer that specifies how to process the /// request. The first `StreamingTranslateSpeechRequest` message must contain /// a `streaming_config` message. #[prost(message, tag = "1")] StreamingConfig(super::StreamingTranslateSpeechConfig), /// 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). #[prost(bytes, tag = "2")] AudioContent(::prost::alloc::vec::Vec), } } /// A streaming speech translation result corresponding to a portion of the audio /// that is currently being processed. #[derive(Clone, PartialEq, ::prost::Message)] pub struct StreamingTranslateSpeechResult { /// Translation result. #[prost(oneof = "streaming_translate_speech_result::Result", tags = "1")] pub result: ::core::option::Option, } /// Nested message and enum types in `StreamingTranslateSpeechResult`. pub mod streaming_translate_speech_result { /// Text translation result. #[derive(Clone, PartialEq, ::prost::Message)] pub struct TextTranslationResult { /// Output only. The translated sentence. #[prost(string, tag = "1")] pub translation: ::prost::alloc::string::String, /// 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. #[prost(bool, tag = "2")] pub is_final: bool, } /// Translation result. #[derive(Clone, PartialEq, ::prost::Oneof)] pub enum Result { /// Text translation result. #[prost(message, tag = "1")] TextTranslationResult(TextTranslationResult), } } /// A streaming speech translation response corresponding to a portion of /// the audio currently processed. #[derive(Clone, PartialEq, ::prost::Message)] pub struct StreamingTranslateSpeechResponse { /// Output only. If set, returns a \[google.rpc.Status][google.rpc.Status\] message that /// specifies the error for the operation. #[prost(message, optional, tag = "1")] pub error: ::core::option::Option, /// Output only. The translation result that is currently being processed (is_final could be /// true or false). #[prost(message, optional, tag = "2")] pub result: ::core::option::Option, /// Output only. Indicates the type of speech event. #[prost(enumeration = "streaming_translate_speech_response::SpeechEventType", tag = "3")] pub speech_event_type: i32, } /// Nested message and enum types in `StreamingTranslateSpeechResponse`. pub mod streaming_translate_speech_response { /// Indicates the type of speech event. #[derive(Clone, Copy, Debug, PartialEq, Eq, Hash, PartialOrd, Ord, ::prost::Enumeration)] #[repr(i32)] pub enum SpeechEventType { /// No speech event specified. 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. EndOfSingleUtterance = 1, } } #[doc = r" Generated client implementations."] pub mod speech_translation_service_client { #![allow(unused_variables, dead_code, missing_docs, clippy::let_unit_value)] use tonic::codegen::*; #[doc = " Provides translation from/to media types."] #[derive(Debug, Clone)] pub struct SpeechTranslationServiceClient { inner: tonic::client::Grpc, } impl SpeechTranslationServiceClient where T: tonic::client::GrpcService, T::ResponseBody: Body + Send + 'static, T::Error: Into, ::Error: Into + Send, { pub fn new(inner: T) -> Self { let inner = tonic::client::Grpc::new(inner); Self { inner } } pub fn with_interceptor( inner: T, interceptor: F, ) -> SpeechTranslationServiceClient> where F: tonic::service::Interceptor, T: tonic::codegen::Service< http::Request, Response = http::Response< >::ResponseBody, >, >, >>::Error: Into + Send + Sync, { SpeechTranslationServiceClient::new(InterceptedService::new(inner, interceptor)) } #[doc = r" Compress requests with `gzip`."] #[doc = r""] #[doc = r" This requires the server to support it otherwise it might respond with an"] #[doc = r" error."] pub fn send_gzip(mut self) -> Self { self.inner = self.inner.send_gzip(); self } #[doc = r" Enable decompressing responses with `gzip`."] pub fn accept_gzip(mut self) -> Self { self.inner = self.inner.accept_gzip(); self } #[doc = " Performs bidirectional streaming speech translation: receive results while"] #[doc = " sending audio. This method is only available via the gRPC API (not REST)."] pub async fn streaming_translate_speech( &mut self, request: impl tonic::IntoStreamingRequest, ) -> Result< tonic::Response>, tonic::Status, > { self.inner.ready().await.map_err(|e| { tonic::Status::new( tonic::Code::Unknown, format!("Service was not ready: {}", e.into()), ) })?; let codec = tonic::codec::ProstCodec::default(); let path = http :: uri :: PathAndQuery :: from_static ("/google.cloud.mediatranslation.v1beta1.SpeechTranslationService/StreamingTranslateSpeech") ; self.inner.streaming(request.into_streaming_request(), path, codec).await } } }