--- /dev/null
+// 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
+//
+// 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.v1;
+
+import "google/api/annotations.proto";
+
+option cc_enable_arenas = true;
+option csharp_namespace = "Google.Cloud.TextToSpeech.V1";
+option go_package = "google.golang.org/genproto/googleapis/cloud/texttospeech/v1;texttospeech";
+option java_multiple_files = true;
+option java_outer_classname = "TextToSpeechProto";
+option java_package = "com.google.cloud.texttospeech.v1";
+option php_namespace = "Google\\Cloud\\TextToSpeech\\V1";
+
+// Service that implements Google Cloud Text-to-Speech API.
+service TextToSpeech {
+ // Returns a list of Voice supported for synthesis.
+ rpc ListVoices(ListVoicesRequest) returns (ListVoicesResponse) {
+ option (google.api.http) = {
+ get: "/v1/voices"
+ };
+ }
+
+ // Synthesizes speech synchronously: receive results after all text input
+ // has been processed.
+ rpc SynthesizeSpeech(SynthesizeSpeechRequest)
+ returns (SynthesizeSpeechResponse) {
+ option (google.api.http) = {
+ post: "/v1/text:synthesize"
+ body: "*"
+ };
+ }
+}
+
+// The top-level message sent by the client for the `ListVoices` method.
+message ListVoicesRequest {
+ // Optional (but recommended)
+ // [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag. 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-*" 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-*" voices.
+ string language_code = 1;
+}
+
+// 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 {
+ // Required. The Synthesizer requires either plain text or SSML as input.
+ SynthesisInput input = 1;
+
+ // Required. The desired voice of the synthesized audio.
+ VoiceSelectionParams voice = 2;
+
+ // Required. The configuration of the synthesized audio.
+ AudioConfig audio_config = 3;
+}
+
+// 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](/speech/text-to-speech/docs/ssml).
+ string ssml = 2;
+ }
+}
+
+// Description of which voice to use for a synthesis request.
+message VoiceSelectionParams {
+ // The language (and optionally 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". Required. 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;
+
+ // The name of the voice. Optional; 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. Optional; 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 requested audio byte stream.
+ AudioEncoding audio_encoding = 1;
+
+ // Optional 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;
+
+ // Optional 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;
+
+ // Optional 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;
+
+ // The synthesis sample rate (in hertz) for this audio. Optional. 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;
+
+ // 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;
+}
+
+// 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 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;
+}
+
+// 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.
+ 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.
+ MP3 = 2;
+
+ // 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;
+}
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