xref: /aosp_15_r20/external/googleapis/google/cloud/texttospeech/v1beta1/cloud_tts.proto (revision d5c09012810ac0c9f33fe448fb6da8260d444cc9)
1// Copyright 2023 Google LLC
2//
3// Licensed under the Apache License, Version 2.0 (the "License");
4// you may not use this file except in compliance with the License.
5// You may obtain a copy of the License at
6//
7//     http://www.apache.org/licenses/LICENSE-2.0
8//
9// Unless required by applicable law or agreed to in writing, software
10// distributed under the License is distributed on an "AS IS" BASIS,
11// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
12// See the License for the specific language governing permissions and
13// limitations under the License.
14
15syntax = "proto3";
16
17package google.cloud.texttospeech.v1beta1;
18
19import "google/api/annotations.proto";
20import "google/api/client.proto";
21import "google/api/field_behavior.proto";
22import "google/api/resource.proto";
23
24option cc_enable_arenas = true;
25option csharp_namespace = "Google.Cloud.TextToSpeech.V1Beta1";
26option go_package = "cloud.google.com/go/texttospeech/apiv1beta1/texttospeechpb;texttospeechpb";
27option java_multiple_files = true;
28option java_outer_classname = "TextToSpeechProto";
29option java_package = "com.google.cloud.texttospeech.v1beta1";
30option php_namespace = "Google\\Cloud\\TextToSpeech\\V1beta1";
31option ruby_package = "Google::Cloud::TextToSpeech::V1beta1";
32option (google.api.resource_definition) = {
33  type: "automl.googleapis.com/Model"
34  pattern: "projects/{project}/locations/{location}/models/{model}"
35};
36
37// Service that implements Google Cloud Text-to-Speech API.
38service TextToSpeech {
39  option (google.api.default_host) = "texttospeech.googleapis.com";
40  option (google.api.oauth_scopes) =
41      "https://www.googleapis.com/auth/cloud-platform";
42
43  // Returns a list of Voice supported for synthesis.
44  rpc ListVoices(ListVoicesRequest) returns (ListVoicesResponse) {
45    option (google.api.http) = {
46      get: "/v1beta1/voices"
47    };
48    option (google.api.method_signature) = "language_code";
49  }
50
51  // Synthesizes speech synchronously: receive results after all text input
52  // has been processed.
53  rpc SynthesizeSpeech(SynthesizeSpeechRequest)
54      returns (SynthesizeSpeechResponse) {
55    option (google.api.http) = {
56      post: "/v1beta1/text:synthesize"
57      body: "*"
58    };
59    option (google.api.method_signature) = "input,voice,audio_config";
60  }
61}
62
63// Gender of the voice as described in
64// [SSML voice element](https://www.w3.org/TR/speech-synthesis11/#edef_voice).
65enum SsmlVoiceGender {
66  // An unspecified gender.
67  // In VoiceSelectionParams, this means that the client doesn't care which
68  // gender the selected voice will have. In the Voice field of
69  // ListVoicesResponse, this may mean that the voice doesn't fit any of the
70  // other categories in this enum, or that the gender of the voice isn't known.
71  SSML_VOICE_GENDER_UNSPECIFIED = 0;
72
73  // A male voice.
74  MALE = 1;
75
76  // A female voice.
77  FEMALE = 2;
78
79  // A gender-neutral voice. This voice is not yet supported.
80  NEUTRAL = 3;
81}
82
83// Configuration to set up audio encoder. The encoding determines the output
84// audio format that we'd like.
85enum AudioEncoding {
86  // Not specified. Will return result
87  // [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT].
88  AUDIO_ENCODING_UNSPECIFIED = 0;
89
90  // Uncompressed 16-bit signed little-endian samples (Linear PCM).
91  // Audio content returned as LINEAR16 also contains a WAV header.
92  LINEAR16 = 1;
93
94  // MP3 audio at 32kbps.
95  MP3 = 2;
96
97  // MP3 at 64kbps.
98  MP3_64_KBPS = 4;
99
100  // Opus encoded audio wrapped in an ogg container. The result will be a
101  // file which can be played natively on Android, and in browsers (at least
102  // Chrome and Firefox). The quality of the encoding is considerably higher
103  // than MP3 while using approximately the same bitrate.
104  OGG_OPUS = 3;
105
106  // 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
107  // Audio content returned as MULAW also contains a WAV header.
108  MULAW = 5;
109
110  // 8-bit samples that compand 14-bit audio samples using G.711 PCMU/A-law.
111  // Audio content returned as ALAW also contains a WAV header.
112  ALAW = 6;
113}
114
115// The top-level message sent by the client for the `ListVoices` method.
116message ListVoicesRequest {
117  // Optional. Recommended.
118  // [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tag.
119  // If not specified, the API will return all supported voices.
120  // If specified, the ListVoices call will only return voices that can be used
121  // to synthesize this language_code. For example, if you specify `"en-NZ"`,
122  // all `"en-NZ"` voices will be returned. If you specify `"no"`, both
123  // `"no-\*"` (Norwegian) and `"nb-\*"` (Norwegian Bokmal) voices will be
124  // returned.
125  string language_code = 1 [(google.api.field_behavior) = OPTIONAL];
126}
127
128// The message returned to the client by the `ListVoices` method.
129message ListVoicesResponse {
130  // The list of voices.
131  repeated Voice voices = 1;
132}
133
134// Description of a voice supported by the TTS service.
135message Voice {
136  // The languages that this voice supports, expressed as
137  // [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt) language tags (e.g.
138  // "en-US", "es-419", "cmn-tw").
139  repeated string language_codes = 1;
140
141  // The name of this voice.  Each distinct voice has a unique name.
142  string name = 2;
143
144  // The gender of this voice.
145  SsmlVoiceGender ssml_gender = 3;
146
147  // The natural sample rate (in hertz) for this voice.
148  int32 natural_sample_rate_hertz = 4;
149}
150
151// The top-level message sent by the client for the `SynthesizeSpeech` method.
152message SynthesizeSpeechRequest {
153  // The type of timepoint information that is returned in the response.
154  enum TimepointType {
155    // Not specified. No timepoint information will be returned.
156    TIMEPOINT_TYPE_UNSPECIFIED = 0;
157
158    // Timepoint information of `<mark>` tags in SSML input will be returned.
159    SSML_MARK = 1;
160  }
161
162  // Required. The Synthesizer requires either plain text or SSML as input.
163  SynthesisInput input = 1 [(google.api.field_behavior) = REQUIRED];
164
165  // Required. The desired voice of the synthesized audio.
166  VoiceSelectionParams voice = 2 [(google.api.field_behavior) = REQUIRED];
167
168  // Required. The configuration of the synthesized audio.
169  AudioConfig audio_config = 3 [(google.api.field_behavior) = REQUIRED];
170
171  // Whether and what timepoints are returned in the response.
172  repeated TimepointType enable_time_pointing = 4;
173}
174
175// Contains text input to be synthesized. Either `text` or `ssml` must be
176// supplied. Supplying both or neither returns
177// [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT]. The
178// input size is limited to 5000 bytes.
179message SynthesisInput {
180  // The input source, which is either plain text or SSML.
181  oneof input_source {
182    // The raw text to be synthesized.
183    string text = 1;
184
185    // The SSML document to be synthesized. The SSML document must be valid
186    // and well-formed. Otherwise the RPC will fail and return
187    // [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT]. For
188    // more information, see
189    // [SSML](https://cloud.google.com/text-to-speech/docs/ssml).
190    string ssml = 2;
191  }
192}
193
194// Description of which voice to use for a synthesis request.
195message VoiceSelectionParams {
196  // Required. The language (and potentially also the region) of the voice
197  // expressed as a [BCP-47](https://www.rfc-editor.org/rfc/bcp/bcp47.txt)
198  // language tag, e.g. "en-US". This should not include a script tag (e.g. use
199  // "cmn-cn" rather than "cmn-Hant-cn"), because the script will be inferred
200  // from the input provided in the SynthesisInput.  The TTS service
201  // will use this parameter to help choose an appropriate voice.  Note that
202  // the TTS service may choose a voice with a slightly different language code
203  // than the one selected; it may substitute a different region
204  // (e.g. using en-US rather than en-CA if there isn't a Canadian voice
205  // available), or even a different language, e.g. using "nb" (Norwegian
206  // Bokmal) instead of "no" (Norwegian)".
207  string language_code = 1 [(google.api.field_behavior) = REQUIRED];
208
209  // The name of the voice. If not set, the service will choose a
210  // voice based on the other parameters such as language_code and gender.
211  string name = 2;
212
213  // The preferred gender of the voice. If not set, the service will
214  // choose a voice based on the other parameters such as language_code and
215  // name. Note that this is only a preference, not requirement; if a
216  // voice of the appropriate gender is not available, the synthesizer should
217  // substitute a voice with a different gender rather than failing the request.
218  SsmlVoiceGender ssml_gender = 3;
219
220  // The configuration for a custom voice. If [CustomVoiceParams.model] is set,
221  // the service will choose the custom voice matching the specified
222  // configuration.
223  CustomVoiceParams custom_voice = 4;
224}
225
226// Description of audio data to be synthesized.
227message AudioConfig {
228  // Required. The format of the audio byte stream.
229  AudioEncoding audio_encoding = 1 [(google.api.field_behavior) = REQUIRED];
230
231  // Optional. Input only. Speaking rate/speed, in the range [0.25, 4.0]. 1.0 is
232  // the normal native speed supported by the specific voice. 2.0 is twice as
233  // fast, and 0.5 is half as fast. If unset(0.0), defaults to the native 1.0
234  // speed. Any other values < 0.25 or > 4.0 will return an error.
235  double speaking_rate = 2 [
236    (google.api.field_behavior) = INPUT_ONLY,
237    (google.api.field_behavior) = OPTIONAL
238  ];
239
240  // Optional. Input only. Speaking pitch, in the range [-20.0, 20.0]. 20 means
241  // increase 20 semitones from the original pitch. -20 means decrease 20
242  // semitones from the original pitch.
243  double pitch = 3 [
244    (google.api.field_behavior) = INPUT_ONLY,
245    (google.api.field_behavior) = OPTIONAL
246  ];
247
248  // Optional. Input only. Volume gain (in dB) of the normal native volume
249  // supported by the specific voice, in the range [-96.0, 16.0]. If unset, or
250  // set to a value of 0.0 (dB), will play at normal native signal amplitude. A
251  // value of -6.0 (dB) will play at approximately half the amplitude of the
252  // normal native signal amplitude. A value of +6.0 (dB) will play at
253  // approximately twice the amplitude of the normal native signal amplitude.
254  // Strongly recommend not to exceed +10 (dB) as there's usually no effective
255  // increase in loudness for any value greater than that.
256  double volume_gain_db = 4 [
257    (google.api.field_behavior) = INPUT_ONLY,
258    (google.api.field_behavior) = OPTIONAL
259  ];
260
261  // Optional. The synthesis sample rate (in hertz) for this audio. When this is
262  // specified in SynthesizeSpeechRequest, if this is different from the voice's
263  // natural sample rate, then the synthesizer will honor this request by
264  // converting to the desired sample rate (which might result in worse audio
265  // quality), unless the specified sample rate is not supported for the
266  // encoding chosen, in which case it will fail the request and return
267  // [google.rpc.Code.INVALID_ARGUMENT][google.rpc.Code.INVALID_ARGUMENT].
268  int32 sample_rate_hertz = 5 [(google.api.field_behavior) = OPTIONAL];
269
270  // Optional. Input only. An identifier which selects 'audio effects' profiles
271  // that are applied on (post synthesized) text to speech. Effects are applied
272  // on top of each other in the order they are given. See
273  // [audio
274  // profiles](https://cloud.google.com/text-to-speech/docs/audio-profiles) for
275  // current supported profile ids.
276  repeated string effects_profile_id = 6 [
277    (google.api.field_behavior) = INPUT_ONLY,
278    (google.api.field_behavior) = OPTIONAL
279  ];
280}
281
282// Description of the custom voice to be synthesized.
283message CustomVoiceParams {
284  // Deprecated. The usage of the synthesized audio. Usage does not affect
285  // billing.
286  enum ReportedUsage {
287    // Request with reported usage unspecified will be rejected.
288    REPORTED_USAGE_UNSPECIFIED = 0;
289
290    // For scenarios where the synthesized audio is not downloadable and can
291    // only be used once. For example, real-time request in IVR system.
292    REALTIME = 1;
293
294    // For scenarios where the synthesized audio is downloadable and can be
295    // reused. For example, the synthesized audio is downloaded, stored in
296    // customer service system and played repeatedly.
297    OFFLINE = 2;
298  }
299
300  // Required. The name of the AutoML model that synthesizes the custom voice.
301  string model = 1 [
302    (google.api.field_behavior) = REQUIRED,
303    (google.api.resource_reference) = { type: "automl.googleapis.com/Model" }
304  ];
305
306  // Optional. Deprecated. The usage of the synthesized audio to be reported.
307  ReportedUsage reported_usage = 3
308      [deprecated = true, (google.api.field_behavior) = OPTIONAL];
309}
310
311// The message returned to the client by the `SynthesizeSpeech` method.
312message SynthesizeSpeechResponse {
313  // The audio data bytes encoded as specified in the request, including the
314  // header for encodings that are wrapped in containers (e.g. MP3, OGG_OPUS).
315  // For LINEAR16 audio, we include the WAV header. Note: as
316  // with all bytes fields, protobuffers use a pure binary representation,
317  // whereas JSON representations use base64.
318  bytes audio_content = 1;
319
320  // A link between a position in the original request input and a corresponding
321  // time in the output audio. It's only supported via `<mark>` of SSML input.
322  repeated Timepoint timepoints = 2;
323
324  // The audio metadata of `audio_content`.
325  AudioConfig audio_config = 4;
326}
327
328// This contains a mapping between a certain point in the input text and a
329// corresponding time in the output audio.
330message Timepoint {
331  // Timepoint name as received from the client within `<mark>` tag.
332  string mark_name = 4;
333
334  // Time offset in seconds from the start of the synthesized audio.
335  double time_seconds = 3;
336}
337