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public interface RecognitionConfigOrBuilder extends MessageOrBuilder
Implements
MessageOrBuilderMethods
getAdaptation()
public abstract SpeechAdaptation getAdaptation()
Speech adaptation configuration improves the accuracy of speech
recognition. For more information, see the speech
adaptation
documentation.
When speech adaptation is set it supersedes the speech_contexts
field.
.google.cloud.speech.v1p1beta1.SpeechAdaptation adaptation = 20;
Type | Description |
SpeechAdaptation | The adaptation. |
getAdaptationOrBuilder()
public abstract SpeechAdaptationOrBuilder getAdaptationOrBuilder()
Speech adaptation configuration improves the accuracy of speech
recognition. For more information, see the speech
adaptation
documentation.
When speech adaptation is set it supersedes the speech_contexts
field.
.google.cloud.speech.v1p1beta1.SpeechAdaptation adaptation = 20;
Type | Description |
SpeechAdaptationOrBuilder |
getAlternativeLanguageCodes(int index)
public abstract String getAlternativeLanguageCodes(int index)
A list of up to 3 additional BCP-47 language tags, listing possible alternative languages of the supplied audio. See Language Support for a list of the currently supported language codes. If alternative languages are listed, recognition result will contain recognition in the most likely language detected including the main language_code. The recognition result will include the language tag of the language detected in the audio. Note: This feature is only supported for Voice Command and Voice Search use cases and performance may vary for other use cases (e.g., phone call transcription).
repeated string alternative_language_codes = 18;
Name | Description |
index | int The index of the element to return. |
Type | Description |
String | The alternativeLanguageCodes at the given index. |
getAlternativeLanguageCodesBytes(int index)
public abstract ByteString getAlternativeLanguageCodesBytes(int index)
A list of up to 3 additional BCP-47 language tags, listing possible alternative languages of the supplied audio. See Language Support for a list of the currently supported language codes. If alternative languages are listed, recognition result will contain recognition in the most likely language detected including the main language_code. The recognition result will include the language tag of the language detected in the audio. Note: This feature is only supported for Voice Command and Voice Search use cases and performance may vary for other use cases (e.g., phone call transcription).
repeated string alternative_language_codes = 18;
Name | Description |
index | int The index of the value to return. |
Type | Description |
ByteString | The bytes of the alternativeLanguageCodes at the given index. |
getAlternativeLanguageCodesCount()
public abstract int getAlternativeLanguageCodesCount()
A list of up to 3 additional BCP-47 language tags, listing possible alternative languages of the supplied audio. See Language Support for a list of the currently supported language codes. If alternative languages are listed, recognition result will contain recognition in the most likely language detected including the main language_code. The recognition result will include the language tag of the language detected in the audio. Note: This feature is only supported for Voice Command and Voice Search use cases and performance may vary for other use cases (e.g., phone call transcription).
repeated string alternative_language_codes = 18;
Type | Description |
int | The count of alternativeLanguageCodes. |
getAlternativeLanguageCodesList()
public abstract List<String> getAlternativeLanguageCodesList()
A list of up to 3 additional BCP-47 language tags, listing possible alternative languages of the supplied audio. See Language Support for a list of the currently supported language codes. If alternative languages are listed, recognition result will contain recognition in the most likely language detected including the main language_code. The recognition result will include the language tag of the language detected in the audio. Note: This feature is only supported for Voice Command and Voice Search use cases and performance may vary for other use cases (e.g., phone call transcription).
repeated string alternative_language_codes = 18;
Type | Description |
List<String> | A list containing the alternativeLanguageCodes. |
getAudioChannelCount()
public abstract int getAudioChannelCount()
The number of channels in the input audio data.
ONLY set this for MULTI-CHANNEL recognition.
Valid values for LINEAR16 and FLAC are 1
-8
.
Valid values for OGG_OPUS are '1'-'254'.
Valid value for MULAW, AMR, AMR_WB and SPEEX_WITH_HEADER_BYTE is only 1
.
If 0
or omitted, defaults to one channel (mono).
Note: We only recognize the first channel by default.
To perform independent recognition on each channel set
enable_separate_recognition_per_channel
to 'true'.
int32 audio_channel_count = 7;
Type | Description |
int | The audioChannelCount. |
getDiarizationConfig()
public abstract SpeakerDiarizationConfig getDiarizationConfig()
Config to enable speaker diarization and set additional parameters to make diarization better suited for your application. Note: When this is enabled, we send all the words from the beginning of the audio for the top alternative in every consecutive STREAMING responses. This is done in order to improve our speaker tags as our models learn to identify the speakers in the conversation over time. For non-streaming requests, the diarization results will be provided only in the top alternative of the FINAL SpeechRecognitionResult.
.google.cloud.speech.v1p1beta1.SpeakerDiarizationConfig diarization_config = 19;
Type | Description |
SpeakerDiarizationConfig | The diarizationConfig. |
getDiarizationConfigOrBuilder()
public abstract SpeakerDiarizationConfigOrBuilder getDiarizationConfigOrBuilder()
Config to enable speaker diarization and set additional parameters to make diarization better suited for your application. Note: When this is enabled, we send all the words from the beginning of the audio for the top alternative in every consecutive STREAMING responses. This is done in order to improve our speaker tags as our models learn to identify the speakers in the conversation over time. For non-streaming requests, the diarization results will be provided only in the top alternative of the FINAL SpeechRecognitionResult.
.google.cloud.speech.v1p1beta1.SpeakerDiarizationConfig diarization_config = 19;
Type | Description |
SpeakerDiarizationConfigOrBuilder |
getDiarizationSpeakerCount() (deprecated)
public abstract int getDiarizationSpeakerCount()
Deprecated. google.cloud.speech.v1p1beta1.RecognitionConfig.diarization_speaker_count is deprecated. See google/cloud/speech/v1p1beta1/cloud_speech.proto;l=383
If set, specifies the estimated number of speakers in the conversation. Defaults to '2'. Ignored unless enable_speaker_diarization is set to true. Note: Use diarization_config instead.
int32 diarization_speaker_count = 17 [deprecated = true];
Type | Description |
int | The diarizationSpeakerCount. |
getEnableAutomaticPunctuation()
public abstract boolean getEnableAutomaticPunctuation()
If 'true', adds punctuation to recognition result hypotheses. This feature is only available in select languages. Setting this for requests in other languages has no effect at all. The default 'false' value does not add punctuation to result hypotheses.
bool enable_automatic_punctuation = 11;
Type | Description |
boolean | The enableAutomaticPunctuation. |
getEnableSeparateRecognitionPerChannel()
public abstract boolean getEnableSeparateRecognitionPerChannel()
This needs to be set to true
explicitly and audio_channel_count
> 1
to get each channel recognized separately. The recognition result will
contain a channel_tag
field to state which channel that result belongs
to. If this is not true, we will only recognize the first channel. The
request is billed cumulatively for all channels recognized:
audio_channel_count
multiplied by the length of the audio.
bool enable_separate_recognition_per_channel = 12;
Type | Description |
boolean | The enableSeparateRecognitionPerChannel. |
getEnableSpeakerDiarization() (deprecated)
public abstract boolean getEnableSpeakerDiarization()
Deprecated. google.cloud.speech.v1p1beta1.RecognitionConfig.enable_speaker_diarization is deprecated. See google/cloud/speech/v1p1beta1/cloud_speech.proto;l=378
If 'true', enables speaker detection for each recognized word in the top alternative of the recognition result using a speaker_tag provided in the WordInfo. Note: Use diarization_config instead.
bool enable_speaker_diarization = 16 [deprecated = true];
Type | Description |
boolean | The enableSpeakerDiarization. |
getEnableSpokenEmojis()
public abstract BoolValue getEnableSpokenEmojis()
The spoken emoji behavior for the call If not set, uses default behavior based on model of choice If 'true', adds spoken emoji formatting for the request. This will replace spoken emojis with the corresponding Unicode symbols in the final transcript. If 'false', spoken emojis are not replaced.
.google.protobuf.BoolValue enable_spoken_emojis = 23;
Type | Description |
BoolValue | The enableSpokenEmojis. |
getEnableSpokenEmojisOrBuilder()
public abstract BoolValueOrBuilder getEnableSpokenEmojisOrBuilder()
The spoken emoji behavior for the call If not set, uses default behavior based on model of choice If 'true', adds spoken emoji formatting for the request. This will replace spoken emojis with the corresponding Unicode symbols in the final transcript. If 'false', spoken emojis are not replaced.
.google.protobuf.BoolValue enable_spoken_emojis = 23;
Type | Description |
BoolValueOrBuilder |
getEnableSpokenPunctuation()
public abstract BoolValue getEnableSpokenPunctuation()
The spoken punctuation behavior for the call If not set, uses default behavior based on model of choice e.g. command_and_search will enable spoken punctuation by default If 'true', replaces spoken punctuation with the corresponding symbols in the request. For example, "how are you question mark" becomes "how are you?". See https://cloud.google.com/speech-to-text/docs/spoken-punctuation for support. If 'false', spoken punctuation is not replaced.
.google.protobuf.BoolValue enable_spoken_punctuation = 22;
Type | Description |
BoolValue | The enableSpokenPunctuation. |
getEnableSpokenPunctuationOrBuilder()
public abstract BoolValueOrBuilder getEnableSpokenPunctuationOrBuilder()
The spoken punctuation behavior for the call If not set, uses default behavior based on model of choice e.g. command_and_search will enable spoken punctuation by default If 'true', replaces spoken punctuation with the corresponding symbols in the request. For example, "how are you question mark" becomes "how are you?". See https://cloud.google.com/speech-to-text/docs/spoken-punctuation for support. If 'false', spoken punctuation is not replaced.
.google.protobuf.BoolValue enable_spoken_punctuation = 22;
Type | Description |
BoolValueOrBuilder |
getEnableWordConfidence()
public abstract boolean getEnableWordConfidence()
If true
, the top result includes a list of words and the
confidence for those words. If false
, no word-level confidence
information is returned. The default is false
.
bool enable_word_confidence = 15;
Type | Description |
boolean | The enableWordConfidence. |
getEnableWordTimeOffsets()
public abstract boolean getEnableWordTimeOffsets()
If true
, the top result includes a list of words and
the start and end time offsets (timestamps) for those words. If
false
, no word-level time offset information is returned. The default is
false
.
bool enable_word_time_offsets = 8;
Type | Description |
boolean | The enableWordTimeOffsets. |
getEncoding()
public abstract RecognitionConfig.AudioEncoding getEncoding()
Encoding of audio data sent in all RecognitionAudio
messages.
This field is optional for FLAC
and WAV
audio files and required
for all other audio formats. For details, see AudioEncoding.
.google.cloud.speech.v1p1beta1.RecognitionConfig.AudioEncoding encoding = 1;
Type | Description |
RecognitionConfig.AudioEncoding | The encoding. |
getEncodingValue()
public abstract int getEncodingValue()
Encoding of audio data sent in all RecognitionAudio
messages.
This field is optional for FLAC
and WAV
audio files and required
for all other audio formats. For details, see AudioEncoding.
.google.cloud.speech.v1p1beta1.RecognitionConfig.AudioEncoding encoding = 1;
Type | Description |
int | The enum numeric value on the wire for encoding. |
getLanguageCode()
public abstract String getLanguageCode()
Required. The language of the supplied audio as a BCP-47 language tag. Example: "en-US". See Language Support for a list of the currently supported language codes.
string language_code = 3 [(.google.api.field_behavior) = REQUIRED];
Type | Description |
String | The languageCode. |
getLanguageCodeBytes()
public abstract ByteString getLanguageCodeBytes()
Required. The language of the supplied audio as a BCP-47 language tag. Example: "en-US". See Language Support for a list of the currently supported language codes.
string language_code = 3 [(.google.api.field_behavior) = REQUIRED];
Type | Description |
ByteString | The bytes for languageCode. |
getMaxAlternatives()
public abstract int getMaxAlternatives()
Maximum number of recognition hypotheses to be returned.
Specifically, the maximum number of SpeechRecognitionAlternative
messages
within each SpeechRecognitionResult
.
The server may return fewer than max_alternatives
.
Valid values are 0
-30
. A value of 0
or 1
will return a maximum of
one. If omitted, will return a maximum of one.
int32 max_alternatives = 4;
Type | Description |
int | The maxAlternatives. |
getMetadata()
public abstract RecognitionMetadata getMetadata()
Metadata regarding this request.
.google.cloud.speech.v1p1beta1.RecognitionMetadata metadata = 9;
Type | Description |
RecognitionMetadata | The metadata. |
getMetadataOrBuilder()
public abstract RecognitionMetadataOrBuilder getMetadataOrBuilder()
Metadata regarding this request.
.google.cloud.speech.v1p1beta1.RecognitionMetadata metadata = 9;
Type | Description |
RecognitionMetadataOrBuilder |
getModel()
public abstract String getModel()
Which model to select for the given request. Select the model best suited to your domain to get best results. If a model is not explicitly specified, then we auto-select a model based on the parameters in the RecognitionConfig. <table> <tr> <td><b>Model</b></td> <td><b>Description</b></td> </tr> <tr> <td><code>latest_long</code></td> <td>Best for long form content like media or conversation.</td> </tr> <tr> <td><code>latest_short</code></td> <td>Best for short form content like commands or single shot directed speech.</td> </tr> <tr> <td><code>command_and_search</code></td> <td>Best for short queries such as voice commands or voice search.</td> </tr> <tr> <td><code>phone_call</code></td> <td>Best for audio that originated from a phone call (typically recorded at an 8khz sampling rate).</td> </tr> <tr> <td><code>video</code></td> <td>Best for audio that originated from video or includes multiple speakers. Ideally the audio is recorded at a 16khz or greater sampling rate. This is a premium model that costs more than the standard rate.</td> </tr> <tr> <td><code>default</code></td> <td>Best for audio that is not one of the specific audio models. For example, long-form audio. Ideally the audio is high-fidelity, recorded at a 16khz or greater sampling rate.</td> </tr> <tr> <td><code>medical_conversation</code></td> <td>Best for audio that originated from a conversation between a medical provider and patient.</td> </tr> <tr> <td><code>medical_dictation</code></td> <td>Best for audio that originated from dictation notes by a medical provider.</td> </tr> </table>
string model = 13;
Type | Description |
String | The model. |
getModelBytes()
public abstract ByteString getModelBytes()
Which model to select for the given request. Select the model best suited to your domain to get best results. If a model is not explicitly specified, then we auto-select a model based on the parameters in the RecognitionConfig. <table> <tr> <td><b>Model</b></td> <td><b>Description</b></td> </tr> <tr> <td><code>latest_long</code></td> <td>Best for long form content like media or conversation.</td> </tr> <tr> <td><code>latest_short</code></td> <td>Best for short form content like commands or single shot directed speech.</td> </tr> <tr> <td><code>command_and_search</code></td> <td>Best for short queries such as voice commands or voice search.</td> </tr> <tr> <td><code>phone_call</code></td> <td>Best for audio that originated from a phone call (typically recorded at an 8khz sampling rate).</td> </tr> <tr> <td><code>video</code></td> <td>Best for audio that originated from video or includes multiple speakers. Ideally the audio is recorded at a 16khz or greater sampling rate. This is a premium model that costs more than the standard rate.</td> </tr> <tr> <td><code>default</code></td> <td>Best for audio that is not one of the specific audio models. For example, long-form audio. Ideally the audio is high-fidelity, recorded at a 16khz or greater sampling rate.</td> </tr> <tr> <td><code>medical_conversation</code></td> <td>Best for audio that originated from a conversation between a medical provider and patient.</td> </tr> <tr> <td><code>medical_dictation</code></td> <td>Best for audio that originated from dictation notes by a medical provider.</td> </tr> </table>
string model = 13;
Type | Description |
ByteString | The bytes for model. |
getProfanityFilter()
public abstract boolean getProfanityFilter()
If set to true
, the server will attempt to filter out
profanities, replacing all but the initial character in each filtered word
with asterisks, e.g. "f***". If set to false
or omitted, profanities
won't be filtered out.
bool profanity_filter = 5;
Type | Description |
boolean | The profanityFilter. |
getSampleRateHertz()
public abstract int getSampleRateHertz()
Sample rate in Hertz of the audio data sent in all
RecognitionAudio
messages. Valid values are: 8000-48000.
16000 is optimal. For best results, set the sampling rate of the audio
source to 16000 Hz. If that's not possible, use the native sample rate of
the audio source (instead of re-sampling).
This field is optional for FLAC and WAV audio files, but is
required for all other audio formats. For details, see AudioEncoding.
int32 sample_rate_hertz = 2;
Type | Description |
int | The sampleRateHertz. |
getSpeechContexts(int index)
public abstract SpeechContext getSpeechContexts(int index)
Array of SpeechContext. A means to provide context to assist the speech recognition. For more information, see speech adaptation.
repeated .google.cloud.speech.v1p1beta1.SpeechContext speech_contexts = 6;
Name | Description |
index | int |
Type | Description |
SpeechContext |
getSpeechContextsCount()
public abstract int getSpeechContextsCount()
Array of SpeechContext. A means to provide context to assist the speech recognition. For more information, see speech adaptation.
repeated .google.cloud.speech.v1p1beta1.SpeechContext speech_contexts = 6;
Type | Description |
int |
getSpeechContextsList()
public abstract List<SpeechContext> getSpeechContextsList()
Array of SpeechContext. A means to provide context to assist the speech recognition. For more information, see speech adaptation.
repeated .google.cloud.speech.v1p1beta1.SpeechContext speech_contexts = 6;
Type | Description |
List<SpeechContext> |
getSpeechContextsOrBuilder(int index)
public abstract SpeechContextOrBuilder getSpeechContextsOrBuilder(int index)
Array of SpeechContext. A means to provide context to assist the speech recognition. For more information, see speech adaptation.
repeated .google.cloud.speech.v1p1beta1.SpeechContext speech_contexts = 6;
Name | Description |
index | int |
Type | Description |
SpeechContextOrBuilder |
getSpeechContextsOrBuilderList()
public abstract List<? extends SpeechContextOrBuilder> getSpeechContextsOrBuilderList()
Array of SpeechContext. A means to provide context to assist the speech recognition. For more information, see speech adaptation.
repeated .google.cloud.speech.v1p1beta1.SpeechContext speech_contexts = 6;
Type | Description |
List<? extends com.google.cloud.speech.v1p1beta1.SpeechContextOrBuilder> |
getTranscriptNormalization()
public abstract TranscriptNormalization getTranscriptNormalization()
Use transcription normalization to automatically replace parts of the transcript with phrases of your choosing. For StreamingRecognize, this normalization only applies to stable partial transcripts (stability > 0.8) and final transcripts.
.google.cloud.speech.v1p1beta1.TranscriptNormalization transcript_normalization = 24;
Type | Description |
TranscriptNormalization | The transcriptNormalization. |
getTranscriptNormalizationOrBuilder()
public abstract TranscriptNormalizationOrBuilder getTranscriptNormalizationOrBuilder()
Use transcription normalization to automatically replace parts of the transcript with phrases of your choosing. For StreamingRecognize, this normalization only applies to stable partial transcripts (stability > 0.8) and final transcripts.
.google.cloud.speech.v1p1beta1.TranscriptNormalization transcript_normalization = 24;
Type | Description |
TranscriptNormalizationOrBuilder |
getUseEnhanced()
public abstract boolean getUseEnhanced()
Set to true to use an enhanced model for speech recognition.
If use_enhanced
is set to true and the model
field is not set, then
an appropriate enhanced model is chosen if an enhanced model exists for
the audio.
If use_enhanced
is true and an enhanced version of the specified model
does not exist, then the speech is recognized using the standard version
of the specified model.
bool use_enhanced = 14;
Type | Description |
boolean | The useEnhanced. |
hasAdaptation()
public abstract boolean hasAdaptation()
Speech adaptation configuration improves the accuracy of speech
recognition. For more information, see the speech
adaptation
documentation.
When speech adaptation is set it supersedes the speech_contexts
field.
.google.cloud.speech.v1p1beta1.SpeechAdaptation adaptation = 20;
Type | Description |
boolean | Whether the adaptation field is set. |
hasDiarizationConfig()
public abstract boolean hasDiarizationConfig()
Config to enable speaker diarization and set additional parameters to make diarization better suited for your application. Note: When this is enabled, we send all the words from the beginning of the audio for the top alternative in every consecutive STREAMING responses. This is done in order to improve our speaker tags as our models learn to identify the speakers in the conversation over time. For non-streaming requests, the diarization results will be provided only in the top alternative of the FINAL SpeechRecognitionResult.
.google.cloud.speech.v1p1beta1.SpeakerDiarizationConfig diarization_config = 19;
Type | Description |
boolean | Whether the diarizationConfig field is set. |
hasEnableSpokenEmojis()
public abstract boolean hasEnableSpokenEmojis()
The spoken emoji behavior for the call If not set, uses default behavior based on model of choice If 'true', adds spoken emoji formatting for the request. This will replace spoken emojis with the corresponding Unicode symbols in the final transcript. If 'false', spoken emojis are not replaced.
.google.protobuf.BoolValue enable_spoken_emojis = 23;
Type | Description |
boolean | Whether the enableSpokenEmojis field is set. |
hasEnableSpokenPunctuation()
public abstract boolean hasEnableSpokenPunctuation()
The spoken punctuation behavior for the call If not set, uses default behavior based on model of choice e.g. command_and_search will enable spoken punctuation by default If 'true', replaces spoken punctuation with the corresponding symbols in the request. For example, "how are you question mark" becomes "how are you?". See https://cloud.google.com/speech-to-text/docs/spoken-punctuation for support. If 'false', spoken punctuation is not replaced.
.google.protobuf.BoolValue enable_spoken_punctuation = 22;
Type | Description |
boolean | Whether the enableSpokenPunctuation field is set. |
hasMetadata()
public abstract boolean hasMetadata()
Metadata regarding this request.
.google.cloud.speech.v1p1beta1.RecognitionMetadata metadata = 9;
Type | Description |
boolean | Whether the metadata field is set. |
hasTranscriptNormalization()
public abstract boolean hasTranscriptNormalization()
Use transcription normalization to automatically replace parts of the transcript with phrases of your choosing. For StreamingRecognize, this normalization only applies to stable partial transcripts (stability > 0.8) and final transcripts.
.google.cloud.speech.v1p1beta1.TranscriptNormalization transcript_normalization = 24;
Type | Description |
boolean | Whether the transcriptNormalization field is set. |