// 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.texttospeech.v1beta1; import "google/api/annotations.proto"; import "google/api/client.proto"; import "google/api/field_behavior.proto"; option cc_enable_arenas = true; option csharp_namespace = "Google.Cloud.TextToSpeech.V1Beta1"; option go_package = "google.golang.org/genproto/googleapis/cloud/texttospeech/v1beta1;texttospeech"; option java_multiple_files = true; option java_outer_classname = "TextToSpeechProto"; option java_package = "com.google.cloud.texttospeech.v1beta1"; option php_namespace = "Google\\Cloud\\TextToSpeech\\V1beta1"; option ruby_package = "Google::Cloud::TextToSpeech::V1beta1"; // Service that implements Google Cloud Text-to-Speech API. service TextToSpeech { option (google.api.default_host) = "texttospeech.googleapis.com"; option (google.api.oauth_scopes) = "https://www.googleapis.com/auth/cloud-platform"; // Returns a list of Voice supported for synthesis. rpc ListVoices(ListVoicesRequest) returns (ListVoicesResponse) { option (google.api.http) = { get: "/v1beta1/voices" }; option (google.api.method_signature) = "language_code"; } // Synthesizes speech synchronously: receive results after all text input // has been processed. rpc SynthesizeSpeech(SynthesizeSpeechRequest) returns (SynthesizeSpeechResponse) { option (google.api.http) = { post: "/v1beta1/text:synthesize" body: "*" }; option (google.api.method_signature) = "input,voice,audio_config"; } } // The top-level message sent by the client for the `ListVoices` method. message ListVoicesRequest { // Optional. Recommended. // [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag. // If not specified, the API will return all supported voices. // If specified, the ListVoices call will only return voices that can be used // to synthesize this language_code. E.g. when specifying "en-NZ", you will // get supported "en-NZ" voices; when specifying "no", you will get supported // "no-\*" (Norwegian) and "nb-\*" (Norwegian Bokmal) voices; specifying "zh" // will also get supported "cmn-\*" voices; specifying "zh-hk" will also get // supported "yue-hk" voices. string language_code = 1 [(google.api.field_behavior) = OPTIONAL]; } // Gender of the voice as described in // [SSML voice element](https://www.w3.org/TR/speech-synthesis11/#edef_voice). enum SsmlVoiceGender { // An unspecified gender. // In VoiceSelectionParams, this means that the client doesn't care which // gender the selected voice will have. In the Voice field of // ListVoicesResponse, this may mean that the voice doesn't fit any of the // other categories in this enum, or that the gender of the voice isn't known. SSML_VOICE_GENDER_UNSPECIFIED = 0; // A male voice. MALE = 1; // A female voice. FEMALE = 2; // A gender-neutral voice. This voice is not yet supported. NEUTRAL = 3; } // Configuration to set up audio encoder. The encoding determines the output // audio format that we'd like. enum AudioEncoding { // Not specified. Will return result [google.rpc.Code.INVALID_ARGUMENT][]. AUDIO_ENCODING_UNSPECIFIED = 0; // Uncompressed 16-bit signed little-endian samples (Linear PCM). // Audio content returned as LINEAR16 also contains a WAV header. LINEAR16 = 1; // MP3 audio at 32kbps. MP3 = 2; // MP3 at 64kbps. MP3_64_KBPS = 4; // Opus encoded audio wrapped in an ogg container. The result will be a // file which can be played natively on Android, and in browsers (at least // Chrome and Firefox). The quality of the encoding is considerably higher // than MP3 while using approximately the same bitrate. OGG_OPUS = 3; // 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law. // Audio content returned as MULAW also contains a WAV header. MULAW = 5; // 8-bit samples that compand 14-bit audio samples using G.711 PCMU/A-law. // Audio content returned as ALAW also contains a WAV header. ALAW = 6; } // The message returned to the client by the `ListVoices` method. message ListVoicesResponse { // The list of voices. repeated Voice voices = 1; } // Description of a voice supported by the TTS service. message Voice { // The languages that this voice supports, expressed as // [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tags (e.g. // "en-US", "es-419", "cmn-tw"). repeated string language_codes = 1; // The name of this voice. Each distinct voice has a unique name. string name = 2; // The gender of this voice. SsmlVoiceGender ssml_gender = 3; // The natural sample rate (in hertz) for this voice. int32 natural_sample_rate_hertz = 4; } // The top-level message sent by the client for the `SynthesizeSpeech` method. message SynthesizeSpeechRequest { // The type of timepoint information that is returned in the response. enum TimepointType { // Not specified. No timepoint information will be returned. TIMEPOINT_TYPE_UNSPECIFIED = 0; // Timepoint information of `` tags in SSML input will be returned. SSML_MARK = 1; } // Required. The Synthesizer requires either plain text or SSML as input. SynthesisInput input = 1 [(google.api.field_behavior) = REQUIRED]; // Required. The desired voice of the synthesized audio. VoiceSelectionParams voice = 2 [(google.api.field_behavior) = REQUIRED]; // Required. The configuration of the synthesized audio. AudioConfig audio_config = 3 [(google.api.field_behavior) = REQUIRED]; // Whether and what timepoints are returned in the response. repeated TimepointType enable_time_pointing = 4; } // Contains text input to be synthesized. Either `text` or `ssml` must be // supplied. Supplying both or neither returns // [google.rpc.Code.INVALID_ARGUMENT][]. The input size is limited to 5000 // characters. message SynthesisInput { // The input source, which is either plain text or SSML. oneof input_source { // The raw text to be synthesized. string text = 1; // The SSML document to be synthesized. The SSML document must be valid // and well-formed. Otherwise the RPC will fail and return // [google.rpc.Code.INVALID_ARGUMENT][]. For more information, see // [SSML](https://cloud.google.com/text-to-speech/docs/ssml). string ssml = 2; } } // Description of which voice to use for a synthesis request. message VoiceSelectionParams { // Required. The language (and potentially also the region) of the voice expressed as a // [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag, e.g. // "en-US". This should not include a script tag (e.g. use // "cmn-cn" rather than "cmn-Hant-cn"), because the script will be inferred // from the input provided in the SynthesisInput. The TTS service // will use this parameter to help choose an appropriate voice. Note that // the TTS service may choose a voice with a slightly different language code // than the one selected; it may substitute a different region // (e.g. using en-US rather than en-CA if there isn't a Canadian voice // available), or even a different language, e.g. using "nb" (Norwegian // Bokmal) instead of "no" (Norwegian)". string language_code = 1 [(google.api.field_behavior) = REQUIRED]; // The name of the voice. If not set, the service will choose a // voice based on the other parameters such as language_code and gender. string name = 2; // The preferred gender of the voice. If not set, the service will // choose a voice based on the other parameters such as language_code and // name. Note that this is only a preference, not requirement; if a // voice of the appropriate gender is not available, the synthesizer should // substitute a voice with a different gender rather than failing the request. SsmlVoiceGender ssml_gender = 3; } // Description of audio data to be synthesized. message AudioConfig { // Required. The format of the audio byte stream. AudioEncoding audio_encoding = 1 [(google.api.field_behavior) = REQUIRED]; // Optional. Input only. Speaking rate/speed, in the range [0.25, 4.0]. 1.0 is // the normal native speed supported by the specific voice. 2.0 is twice as // fast, and 0.5 is half as fast. If unset(0.0), defaults to the native 1.0 // speed. Any other values < 0.25 or > 4.0 will return an error. double speaking_rate = 2 [ (google.api.field_behavior) = INPUT_ONLY, (google.api.field_behavior) = OPTIONAL ]; // Optional. Input only. Speaking pitch, in the range [-20.0, 20.0]. 20 means // increase 20 semitones from the original pitch. -20 means decrease 20 // semitones from the original pitch. double pitch = 3 [ (google.api.field_behavior) = INPUT_ONLY, (google.api.field_behavior) = OPTIONAL ]; // Optional. Input only. Volume gain (in dB) of the normal native volume // supported by the specific voice, in the range [-96.0, 16.0]. If unset, or // set to a value of 0.0 (dB), will play at normal native signal amplitude. A // value of -6.0 (dB) will play at approximately half the amplitude of the // normal native signal amplitude. A value of +6.0 (dB) will play at // approximately twice the amplitude of the normal native signal amplitude. // Strongly recommend not to exceed +10 (dB) as there's usually no effective // increase in loudness for any value greater than that. double volume_gain_db = 4 [ (google.api.field_behavior) = INPUT_ONLY, (google.api.field_behavior) = OPTIONAL ]; // Optional. The synthesis sample rate (in hertz) for this audio. When this is // specified in SynthesizeSpeechRequest, if this is different from the voice's // natural sample rate, then the synthesizer will honor this request by // converting to the desired sample rate (which might result in worse audio // quality), unless the specified sample rate is not supported for the // encoding chosen, in which case it will fail the request and return // [google.rpc.Code.INVALID_ARGUMENT][]. int32 sample_rate_hertz = 5 [(google.api.field_behavior) = OPTIONAL]; // Optional. Input only. An identifier which selects 'audio effects' profiles // that are applied on (post synthesized) text to speech. Effects are applied // on top of each other in the order they are given. See // [audio // profiles](https://cloud.google.com/text-to-speech/docs/audio-profiles) for // current supported profile ids. repeated string effects_profile_id = 6 [ (google.api.field_behavior) = INPUT_ONLY, (google.api.field_behavior) = OPTIONAL ]; } // The message returned to the client by the `SynthesizeSpeech` method. message SynthesizeSpeechResponse { // The audio data bytes encoded as specified in the request, including the // header for encodings that are wrapped in containers (e.g. MP3, OGG_OPUS). // For LINEAR16 audio, we include the WAV header. Note: as // with all bytes fields, protobuffers use a pure binary representation, // whereas JSON representations use base64. bytes audio_content = 1; // A link between a position in the original request input and a corresponding // time in the output audio. It's only supported via `` of SSML input. repeated Timepoint timepoints = 2; // The audio metadata of `audio_content`. AudioConfig audio_config = 4; } // This contains a mapping between a certain point in the input text and a // corresponding time in the output audio. message Timepoint { // Timepoint name as received from the client within `` tag. string mark_name = 4; // Time offset in seconds from the start of the synthesized audio. double time_seconds = 3; }