xref: /aosp_15_r20/external/googleapis/google/cloud/dialogflow/v2/audio_config.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.dialogflow.v2;
18
19import "google/api/field_behavior.proto";
20import "google/api/resource.proto";
21import "google/protobuf/duration.proto";
22
23option cc_enable_arenas = true;
24option csharp_namespace = "Google.Cloud.Dialogflow.V2";
25option go_package = "cloud.google.com/go/dialogflow/apiv2/dialogflowpb;dialogflowpb";
26option java_multiple_files = true;
27option java_outer_classname = "AudioConfigProto";
28option java_package = "com.google.cloud.dialogflow.v2";
29option objc_class_prefix = "DF";
30option (google.api.resource_definition) = {
31  type: "automl.googleapis.com/Model"
32  pattern: "projects/{project}/locations/{location}/models/{model}"
33};
34option (google.api.resource_definition) = {
35  type: "speech.googleapis.com/PhraseSet"
36  pattern: "projects/{project}/locations/{location}/phraseSets/{phrase_set}"
37};
38
39// Hints for the speech recognizer to help with recognition in a specific
40// conversation state.
41message SpeechContext {
42  // Optional. A list of strings containing words and phrases that the speech
43  // recognizer should recognize with higher likelihood.
44  //
45  // This list can be used to:
46  //
47  // * improve accuracy for words and phrases you expect the user to say,
48  //   e.g. typical commands for your Dialogflow agent
49  // * add additional words to the speech recognizer vocabulary
50  // * ...
51  //
52  // See the [Cloud Speech
53  // documentation](https://cloud.google.com/speech-to-text/quotas) for usage
54  // limits.
55  repeated string phrases = 1 [(google.api.field_behavior) = OPTIONAL];
56
57  // Optional. Boost for this context compared to other contexts:
58  //
59  // * If the boost is positive, Dialogflow will increase the probability that
60  //   the phrases in this context are recognized over similar sounding phrases.
61  // * If the boost is unspecified or non-positive, Dialogflow will not apply
62  //   any boost.
63  //
64  // Dialogflow recommends that you use boosts in the range (0, 20] and that you
65  // find a value that fits your use case with binary search.
66  float boost = 2 [(google.api.field_behavior) = OPTIONAL];
67}
68
69// Information for a word recognized by the speech recognizer.
70message SpeechWordInfo {
71  // The word this info is for.
72  string word = 3;
73
74  // Time offset relative to the beginning of the audio that corresponds to the
75  // start of the spoken word. This is an experimental feature and the accuracy
76  // of the time offset can vary.
77  google.protobuf.Duration start_offset = 1;
78
79  // Time offset relative to the beginning of the audio that corresponds to the
80  // end of the spoken word. This is an experimental feature and the accuracy of
81  // the time offset can vary.
82  google.protobuf.Duration end_offset = 2;
83
84  // The Speech confidence between 0.0 and 1.0 for this word. A higher number
85  // indicates an estimated greater likelihood that the recognized word is
86  // correct. The default of 0.0 is a sentinel value indicating that confidence
87  // was not set.
88  //
89  // This field is not guaranteed to be fully stable over time for the same
90  // audio input. Users should also not rely on it to always be provided.
91  float confidence = 4;
92}
93
94// Instructs the speech recognizer how to process the audio content.
95message InputAudioConfig {
96  // Required. Audio encoding of the audio content to process.
97  AudioEncoding audio_encoding = 1 [(google.api.field_behavior) = REQUIRED];
98
99  // Required. Sample rate (in Hertz) of the audio content sent in the query.
100  // Refer to [Cloud Speech API
101  // documentation](https://cloud.google.com/speech-to-text/docs/basics) for
102  // more details.
103  int32 sample_rate_hertz = 2 [(google.api.field_behavior) = REQUIRED];
104
105  // Required. The language of the supplied audio. Dialogflow does not do
106  // translations. See [Language
107  // Support](https://cloud.google.com/dialogflow/docs/reference/language)
108  // for a list of the currently supported language codes. Note that queries in
109  // the same session do not necessarily need to specify the same language.
110  string language_code = 3 [(google.api.field_behavior) = REQUIRED];
111
112  // If `true`, Dialogflow returns
113  // [SpeechWordInfo][google.cloud.dialogflow.v2.SpeechWordInfo] in
114  // [StreamingRecognitionResult][google.cloud.dialogflow.v2.StreamingRecognitionResult]
115  // with information about the recognized speech words, e.g. start and end time
116  // offsets. If false or unspecified, Speech doesn't return any word-level
117  // information.
118  bool enable_word_info = 13;
119
120  // A list of strings containing words and phrases that the speech
121  // recognizer should recognize with higher likelihood.
122  //
123  // See [the Cloud Speech
124  // documentation](https://cloud.google.com/speech-to-text/docs/basics#phrase-hints)
125  // for more details.
126  //
127  // This field is deprecated. Please use [`speech_contexts`]() instead. If you
128  // specify both [`phrase_hints`]() and [`speech_contexts`](), Dialogflow will
129  // treat the [`phrase_hints`]() as a single additional [`SpeechContext`]().
130  repeated string phrase_hints = 4 [deprecated = true];
131
132  // Context information to assist speech recognition.
133  //
134  // See [the Cloud Speech
135  // documentation](https://cloud.google.com/speech-to-text/docs/basics#phrase-hints)
136  // for more details.
137  repeated SpeechContext speech_contexts = 11;
138
139  // Optional. Which Speech model to select for the given request.
140  // For more information, see
141  // [Speech models](https://cloud.google.com/dialogflow/es/docs/speech-models).
142  string model = 7;
143
144  // Which variant of the [Speech
145  // model][google.cloud.dialogflow.v2.InputAudioConfig.model] to use.
146  SpeechModelVariant model_variant = 10;
147
148  // If `false` (default), recognition does not cease until the
149  // client closes the stream.
150  // If `true`, the recognizer will detect a single spoken utterance in input
151  // audio. Recognition ceases when it detects the audio's voice has
152  // stopped or paused. In this case, once a detected intent is received, the
153  // client should close the stream and start a new request with a new stream as
154  // needed.
155  // Note: This setting is relevant only for streaming methods.
156  // Note: When specified, InputAudioConfig.single_utterance takes precedence
157  // over StreamingDetectIntentRequest.single_utterance.
158  bool single_utterance = 8;
159
160  // Only used in
161  // [Participants.AnalyzeContent][google.cloud.dialogflow.v2.Participants.AnalyzeContent]
162  // and
163  // [Participants.StreamingAnalyzeContent][google.cloud.dialogflow.v2.Participants.StreamingAnalyzeContent].
164  // If `false` and recognition doesn't return any result, trigger
165  // `NO_SPEECH_RECOGNIZED` event to Dialogflow agent.
166  bool disable_no_speech_recognized_event = 14;
167
168  // Enable automatic punctuation option at the speech backend.
169  bool enable_automatic_punctuation = 17;
170
171  // If `true`, the request will opt out for STT conformer model migration.
172  // This field will be deprecated once force migration takes place in June
173  // 2024. Please refer to [Dialogflow ES Speech model
174  // migration](https://cloud.google.com/dialogflow/es/docs/speech-model-migration).
175  bool opt_out_conformer_model_migration = 26;
176}
177
178// Description of which voice to use for speech synthesis.
179message VoiceSelectionParams {
180  // Optional. The name of the voice. If not set, the service will choose a
181  // voice based on the other parameters such as language_code and
182  // [ssml_gender][google.cloud.dialogflow.v2.VoiceSelectionParams.ssml_gender].
183  string name = 1 [(google.api.field_behavior) = OPTIONAL];
184
185  // Optional. The preferred gender of the voice. If not set, the service will
186  // choose a voice based on the other parameters such as language_code and
187  // [name][google.cloud.dialogflow.v2.VoiceSelectionParams.name]. Note that
188  // this is only a preference, not requirement. If a voice of the appropriate
189  // gender is not available, the synthesizer should substitute a voice with a
190  // different gender rather than failing the request.
191  SsmlVoiceGender ssml_gender = 2 [(google.api.field_behavior) = OPTIONAL];
192}
193
194// Configuration of how speech should be synthesized.
195message SynthesizeSpeechConfig {
196  // Optional. Speaking rate/speed, in the range [0.25, 4.0]. 1.0 is the normal
197  // native speed supported by the specific voice. 2.0 is twice as fast, and 0.5
198  // is half as fast. If unset(0.0), defaults to the native 1.0 speed. Any other
199  // values < 0.25 or > 4.0 will return an error.
200  double speaking_rate = 1 [(google.api.field_behavior) = OPTIONAL];
201
202  // Optional. Speaking pitch, in the range [-20.0, 20.0]. 20 means increase 20
203  // semitones from the original pitch. -20 means decrease 20 semitones from the
204  // original pitch.
205  double pitch = 2 [(google.api.field_behavior) = OPTIONAL];
206
207  // Optional. Volume gain (in dB) of the normal native volume supported by the
208  // specific voice, in the range [-96.0, 16.0]. If unset, or set to a value of
209  // 0.0 (dB), will play at normal native signal amplitude. A value of -6.0 (dB)
210  // will play at approximately half the amplitude of the normal native signal
211  // amplitude. A value of +6.0 (dB) will play at approximately twice the
212  // amplitude of the normal native signal amplitude. We strongly recommend not
213  // to exceed +10 (dB) as there's usually no effective increase in loudness for
214  // any value greater than that.
215  double volume_gain_db = 3 [(google.api.field_behavior) = OPTIONAL];
216
217  // Optional. An identifier which selects 'audio effects' profiles that are
218  // applied on (post synthesized) text to speech. Effects are applied on top of
219  // each other in the order they are given.
220  repeated string effects_profile_id = 5
221      [(google.api.field_behavior) = OPTIONAL];
222
223  // Optional. The desired voice of the synthesized audio.
224  VoiceSelectionParams voice = 4 [(google.api.field_behavior) = OPTIONAL];
225}
226
227// Instructs the speech synthesizer on how to generate the output audio content.
228// If this audio config is supplied in a request, it overrides all existing
229// text-to-speech settings applied to the agent.
230message OutputAudioConfig {
231  // Required. Audio encoding of the synthesized audio content.
232  OutputAudioEncoding audio_encoding = 1
233      [(google.api.field_behavior) = REQUIRED];
234
235  // The synthesis sample rate (in hertz) for this audio. If not
236  // provided, then the synthesizer will use the default sample rate based on
237  // the audio encoding. If this is different from the voice's natural sample
238  // rate, then the synthesizer will honor this request by converting to the
239  // desired sample rate (which might result in worse audio quality).
240  int32 sample_rate_hertz = 2;
241
242  // Configuration of how speech should be synthesized.
243  SynthesizeSpeechConfig synthesize_speech_config = 3;
244}
245
246// A wrapper of repeated TelephonyDtmf digits.
247message TelephonyDtmfEvents {
248  // A sequence of TelephonyDtmf digits.
249  repeated TelephonyDtmf dtmf_events = 1;
250}
251
252// Configures speech transcription for
253// [ConversationProfile][google.cloud.dialogflow.v2.ConversationProfile].
254message SpeechToTextConfig {
255  // The speech model used in speech to text.
256  // `SPEECH_MODEL_VARIANT_UNSPECIFIED`, `USE_BEST_AVAILABLE` will be treated as
257  // `USE_ENHANCED`. It can be overridden in
258  // [AnalyzeContentRequest][google.cloud.dialogflow.v2.AnalyzeContentRequest]
259  // and
260  // [StreamingAnalyzeContentRequest][google.cloud.dialogflow.v2.StreamingAnalyzeContentRequest]
261  // request. If enhanced model variant is specified and an enhanced version of
262  // the specified model for the language does not exist, then it would emit an
263  // error.
264  SpeechModelVariant speech_model_variant = 1;
265
266  // Which Speech model to select. Select the
267  // model best suited to your domain to get best results. If a model is not
268  // explicitly specified, then Dialogflow auto-selects a model based on other
269  // parameters in the SpeechToTextConfig and Agent settings.
270  // If enhanced speech model is enabled for the agent and an enhanced
271  // version of the specified model for the language does not exist, then the
272  // speech is recognized using the standard version of the specified model.
273  // Refer to
274  // [Cloud Speech API
275  // documentation](https://cloud.google.com/speech-to-text/docs/basics#select-model)
276  // for more details.
277  // If you specify a model, the following models typically have the best
278  // performance:
279  //
280  // - phone_call (best for Agent Assist and telephony)
281  // - latest_short (best for Dialogflow non-telephony)
282  // - command_and_search
283  //
284  // Leave this field unspecified to use
285  // [Agent Speech
286  // settings](https://cloud.google.com/dialogflow/cx/docs/concept/agent#settings-speech)
287  // for model selection.
288  string model = 2;
289
290  // Use timeout based endpointing, interpreting endpointer sensitivy as
291  // seconds of timeout value.
292  bool use_timeout_based_endpointing = 11;
293}
294
295// [DTMF](https://en.wikipedia.org/wiki/Dual-tone_multi-frequency_signaling)
296// digit in Telephony Gateway.
297enum TelephonyDtmf {
298  // Not specified. This value may be used to indicate an absent digit.
299  TELEPHONY_DTMF_UNSPECIFIED = 0;
300
301  // Number: '1'.
302  DTMF_ONE = 1;
303
304  // Number: '2'.
305  DTMF_TWO = 2;
306
307  // Number: '3'.
308  DTMF_THREE = 3;
309
310  // Number: '4'.
311  DTMF_FOUR = 4;
312
313  // Number: '5'.
314  DTMF_FIVE = 5;
315
316  // Number: '6'.
317  DTMF_SIX = 6;
318
319  // Number: '7'.
320  DTMF_SEVEN = 7;
321
322  // Number: '8'.
323  DTMF_EIGHT = 8;
324
325  // Number: '9'.
326  DTMF_NINE = 9;
327
328  // Number: '0'.
329  DTMF_ZERO = 10;
330
331  // Letter: 'A'.
332  DTMF_A = 11;
333
334  // Letter: 'B'.
335  DTMF_B = 12;
336
337  // Letter: 'C'.
338  DTMF_C = 13;
339
340  // Letter: 'D'.
341  DTMF_D = 14;
342
343  // Asterisk/star: '*'.
344  DTMF_STAR = 15;
345
346  // Pound/diamond/hash/square/gate/octothorpe: '#'.
347  DTMF_POUND = 16;
348}
349
350// Audio encoding of the audio content sent in the conversational query request.
351// Refer to the
352// [Cloud Speech API
353// documentation](https://cloud.google.com/speech-to-text/docs/basics) for more
354// details.
355enum AudioEncoding {
356  // Not specified.
357  AUDIO_ENCODING_UNSPECIFIED = 0;
358
359  // Uncompressed 16-bit signed little-endian samples (Linear PCM).
360  AUDIO_ENCODING_LINEAR_16 = 1;
361
362  // [`FLAC`](https://xiph.org/flac/documentation.html) (Free Lossless Audio
363  // Codec) is the recommended encoding because it is lossless (therefore
364  // recognition is not compromised) and requires only about half the
365  // bandwidth of `LINEAR16`. `FLAC` stream encoding supports 16-bit and
366  // 24-bit samples, however, not all fields in `STREAMINFO` are supported.
367  AUDIO_ENCODING_FLAC = 2;
368
369  // 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
370  AUDIO_ENCODING_MULAW = 3;
371
372  // Adaptive Multi-Rate Narrowband codec. `sample_rate_hertz` must be 8000.
373  AUDIO_ENCODING_AMR = 4;
374
375  // Adaptive Multi-Rate Wideband codec. `sample_rate_hertz` must be 16000.
376  AUDIO_ENCODING_AMR_WB = 5;
377
378  // Opus encoded audio frames in Ogg container
379  // ([OggOpus](https://wiki.xiph.org/OggOpus)).
380  // `sample_rate_hertz` must be 16000.
381  AUDIO_ENCODING_OGG_OPUS = 6;
382
383  // Although the use of lossy encodings is not recommended, if a very low
384  // bitrate encoding is required, `OGG_OPUS` is highly preferred over
385  // Speex encoding. The [Speex](https://speex.org/) encoding supported by
386  // Dialogflow API has a header byte in each block, as in MIME type
387  // `audio/x-speex-with-header-byte`.
388  // It is a variant of the RTP Speex encoding defined in
389  // [RFC 5574](https://tools.ietf.org/html/rfc5574).
390  // The stream is a sequence of blocks, one block per RTP packet. Each block
391  // starts with a byte containing the length of the block, in bytes, followed
392  // by one or more frames of Speex data, padded to an integral number of
393  // bytes (octets) as specified in RFC 5574. In other words, each RTP header
394  // is replaced with a single byte containing the block length. Only Speex
395  // wideband is supported. `sample_rate_hertz` must be 16000.
396  AUDIO_ENCODING_SPEEX_WITH_HEADER_BYTE = 7;
397}
398
399// Variant of the specified [Speech
400// model][google.cloud.dialogflow.v2.InputAudioConfig.model] to use.
401//
402// See the [Cloud Speech
403// documentation](https://cloud.google.com/speech-to-text/docs/enhanced-models)
404// for which models have different variants. For example, the "phone_call" model
405// has both a standard and an enhanced variant. When you use an enhanced model,
406// you will generally receive higher quality results than for a standard model.
407enum SpeechModelVariant {
408  // No model variant specified. In this case Dialogflow defaults to
409  // USE_BEST_AVAILABLE.
410  SPEECH_MODEL_VARIANT_UNSPECIFIED = 0;
411
412  // Use the best available variant of the [Speech
413  // model][InputAudioConfig.model] that the caller is eligible for.
414  //
415  // Please see the [Dialogflow
416  // docs](https://cloud.google.com/dialogflow/docs/data-logging) for
417  // how to make your project eligible for enhanced models.
418  USE_BEST_AVAILABLE = 1;
419
420  // Use standard model variant even if an enhanced model is available.  See the
421  // [Cloud Speech
422  // documentation](https://cloud.google.com/speech-to-text/docs/enhanced-models)
423  // for details about enhanced models.
424  USE_STANDARD = 2;
425
426  // Use an enhanced model variant:
427  //
428  // * If an enhanced variant does not exist for the given
429  //   [model][google.cloud.dialogflow.v2.InputAudioConfig.model] and request
430  //   language, Dialogflow falls back to the standard variant.
431  //
432  //   The [Cloud Speech
433  //   documentation](https://cloud.google.com/speech-to-text/docs/enhanced-models)
434  //   describes which models have enhanced variants.
435  //
436  // * If the API caller isn't eligible for enhanced models, Dialogflow returns
437  //   an error. Please see the [Dialogflow
438  //   docs](https://cloud.google.com/dialogflow/docs/data-logging)
439  //   for how to make your project eligible.
440  USE_ENHANCED = 3;
441}
442
443// Gender of the voice as described in
444// [SSML voice element](https://www.w3.org/TR/speech-synthesis11/#edef_voice).
445enum SsmlVoiceGender {
446  // An unspecified gender, which means that the client doesn't care which
447  // gender the selected voice will have.
448  SSML_VOICE_GENDER_UNSPECIFIED = 0;
449
450  // A male voice.
451  SSML_VOICE_GENDER_MALE = 1;
452
453  // A female voice.
454  SSML_VOICE_GENDER_FEMALE = 2;
455
456  // A gender-neutral voice.
457  SSML_VOICE_GENDER_NEUTRAL = 3;
458}
459
460// Audio encoding of the output audio format in Text-To-Speech.
461enum OutputAudioEncoding {
462  // Not specified.
463  OUTPUT_AUDIO_ENCODING_UNSPECIFIED = 0;
464
465  // Uncompressed 16-bit signed little-endian samples (Linear PCM).
466  // Audio content returned as LINEAR16 also contains a WAV header.
467  OUTPUT_AUDIO_ENCODING_LINEAR_16 = 1;
468
469  // MP3 audio at 32kbps.
470  OUTPUT_AUDIO_ENCODING_MP3 = 2;
471
472  // MP3 audio at 64kbps.
473  OUTPUT_AUDIO_ENCODING_MP3_64_KBPS = 4;
474
475  // Opus encoded audio wrapped in an ogg container. The result will be a
476  // file which can be played natively on Android, and in browsers (at least
477  // Chrome and Firefox). The quality of the encoding is considerably higher
478  // than MP3 while using approximately the same bitrate.
479  OUTPUT_AUDIO_ENCODING_OGG_OPUS = 3;
480
481  // 8-bit samples that compand 14-bit audio samples using G.711 PCMU/mu-law.
482  OUTPUT_AUDIO_ENCODING_MULAW = 5;
483}
484