grpc 第三方依赖 就是grpc的 third_party 文件夹
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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.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 `<mark>` 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 `<mark>` 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 `<mark>` tag.
string mark_name = 4;
// Time offset in seconds from the start of the synthesized audio.
double time_seconds = 3;
}