#[non_exhaustive]
pub struct StartCallAnalyticsStreamTranscriptionInput {
Show 14 fields pub language_code: Option<CallAnalyticsLanguageCode>, pub media_sample_rate_hertz: Option<i32>, pub media_encoding: Option<MediaEncoding>, pub vocabulary_name: Option<String>, pub session_id: Option<String>, pub audio_stream: EventStreamSender<AudioStream, AudioStreamError>, pub vocabulary_filter_name: Option<String>, pub vocabulary_filter_method: Option<VocabularyFilterMethod>, pub language_model_name: Option<String>, pub enable_partial_results_stabilization: Option<bool>, pub partial_results_stability: Option<PartialResultsStability>, pub content_identification_type: Option<ContentIdentificationType>, pub content_redaction_type: Option<ContentRedactionType>, pub pii_entity_types: Option<String>,
}

Fields (Non-exhaustive)§

This struct is marked as non-exhaustive
Non-exhaustive structs could have additional fields added in future. Therefore, non-exhaustive structs cannot be constructed in external crates using the traditional Struct { .. } syntax; cannot be matched against without a wildcard ..; and struct update syntax will not work.
§language_code: Option<CallAnalyticsLanguageCode>

Specify the language code that represents the language spoken in your audio.

If you're unsure of the language spoken in your audio, consider using IdentifyLanguage to enable automatic language identification.

For a list of languages supported with streaming Call Analytics, refer to the Supported languages table.

§media_sample_rate_hertz: Option<i32>

The sample rate of the input audio (in hertz). Low-quality audio, such as telephone audio, is typically around 8,000 Hz. High-quality audio typically ranges from 16,000 Hz to 48,000 Hz. Note that the sample rate you specify must match that of your audio.

§media_encoding: Option<MediaEncoding>

Specify the encoding of your input audio. Supported formats are:

  • FLAC

  • OPUS-encoded audio in an Ogg container

  • PCM (only signed 16-bit little-endian audio formats, which does not include WAV)

For more information, see Media formats.

§vocabulary_name: Option<String>

Specify the name of the custom vocabulary that you want to use when processing your transcription. Note that vocabulary names are case sensitive.

If the language of the specified custom vocabulary doesn't match the language identified in your media, the custom vocabulary is not applied to your transcription.

For more information, see Custom vocabularies.

§session_id: Option<String>

Specify a name for your Call Analytics transcription session. If you don't include this parameter in your request, Amazon Transcribe generates an ID and returns it in the response.

You can use a session ID to retry a streaming session.

§audio_stream: EventStreamSender<AudioStream, AudioStreamError>

An encoded stream of audio blobs. Audio streams are encoded as either HTTP/2 or WebSocket data frames.

For more information, see Transcribing streaming audio.

§vocabulary_filter_name: Option<String>

Specify the name of the custom vocabulary filter that you want to use when processing your transcription. Note that vocabulary filter names are case sensitive.

If the language of the specified custom vocabulary filter doesn't match the language identified in your media, the vocabulary filter is not applied to your transcription.

For more information, see Using vocabulary filtering with unwanted words.

§vocabulary_filter_method: Option<VocabularyFilterMethod>

Specify how you want your vocabulary filter applied to your transcript.

To replace words with ***, choose mask.

To delete words, choose remove.

To flag words without changing them, choose tag.

§language_model_name: Option<String>

Specify the name of the custom language model that you want to use when processing your transcription. Note that language model names are case sensitive.

The language of the specified language model must match the language code you specify in your transcription request. If the languages don't match, the custom language model isn't applied. There are no errors or warnings associated with a language mismatch.

For more information, see Custom language models.

§enable_partial_results_stabilization: Option<bool>

Enables partial result stabilization for your transcription. Partial result stabilization can reduce latency in your output, but may impact accuracy. For more information, see Partial-result stabilization.

§partial_results_stability: Option<PartialResultsStability>

Specify the level of stability to use when you enable partial results stabilization (EnablePartialResultsStabilization).

Low stability provides the highest accuracy. High stability transcribes faster, but with slightly lower accuracy.

For more information, see Partial-result stabilization.

§content_identification_type: Option<ContentIdentificationType>

Labels all personally identifiable information (PII) identified in your transcript.

Content identification is performed at the segment level; PII specified in PiiEntityTypes is flagged upon complete transcription of an audio segment.

You can’t set ContentIdentificationType and ContentRedactionType in the same request. If you set both, your request returns a BadRequestException.

For more information, see Redacting or identifying personally identifiable information.

§content_redaction_type: Option<ContentRedactionType>

Redacts all personally identifiable information (PII) identified in your transcript.

Content redaction is performed at the segment level; PII specified in PiiEntityTypes is redacted upon complete transcription of an audio segment.

You can’t set ContentRedactionType and ContentIdentificationType in the same request. If you set both, your request returns a BadRequestException.

For more information, see Redacting or identifying personally identifiable information.

§pii_entity_types: Option<String>

Specify which types of personally identifiable information (PII) you want to redact in your transcript. You can include as many types as you'd like, or you can select ALL.

To include PiiEntityTypes in your Call Analytics request, you must also include either ContentIdentificationType or ContentRedactionType.

Values must be comma-separated and can include: BANK_ACCOUNT_NUMBER, BANK_ROUTING, CREDIT_DEBIT_NUMBER, CREDIT_DEBIT_CVV, CREDIT_DEBIT_EXPIRY, PIN, EMAIL, ADDRESS, NAME, PHONE, SSN, or ALL.

Implementations§

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impl StartCallAnalyticsStreamTranscriptionInput

source

pub fn language_code(&self) -> Option<&CallAnalyticsLanguageCode>

Specify the language code that represents the language spoken in your audio.

If you're unsure of the language spoken in your audio, consider using IdentifyLanguage to enable automatic language identification.

For a list of languages supported with streaming Call Analytics, refer to the Supported languages table.

source

pub fn media_sample_rate_hertz(&self) -> Option<i32>

The sample rate of the input audio (in hertz). Low-quality audio, such as telephone audio, is typically around 8,000 Hz. High-quality audio typically ranges from 16,000 Hz to 48,000 Hz. Note that the sample rate you specify must match that of your audio.

source

pub fn media_encoding(&self) -> Option<&MediaEncoding>

Specify the encoding of your input audio. Supported formats are:

  • FLAC

  • OPUS-encoded audio in an Ogg container

  • PCM (only signed 16-bit little-endian audio formats, which does not include WAV)

For more information, see Media formats.

source

pub fn vocabulary_name(&self) -> Option<&str>

Specify the name of the custom vocabulary that you want to use when processing your transcription. Note that vocabulary names are case sensitive.

If the language of the specified custom vocabulary doesn't match the language identified in your media, the custom vocabulary is not applied to your transcription.

For more information, see Custom vocabularies.

source

pub fn session_id(&self) -> Option<&str>

Specify a name for your Call Analytics transcription session. If you don't include this parameter in your request, Amazon Transcribe generates an ID and returns it in the response.

You can use a session ID to retry a streaming session.

source

pub fn audio_stream(&self) -> &EventStreamSender<AudioStream, AudioStreamError>

An encoded stream of audio blobs. Audio streams are encoded as either HTTP/2 or WebSocket data frames.

For more information, see Transcribing streaming audio.

source

pub fn vocabulary_filter_name(&self) -> Option<&str>

Specify the name of the custom vocabulary filter that you want to use when processing your transcription. Note that vocabulary filter names are case sensitive.

If the language of the specified custom vocabulary filter doesn't match the language identified in your media, the vocabulary filter is not applied to your transcription.

For more information, see Using vocabulary filtering with unwanted words.

source

pub fn vocabulary_filter_method(&self) -> Option<&VocabularyFilterMethod>

Specify how you want your vocabulary filter applied to your transcript.

To replace words with ***, choose mask.

To delete words, choose remove.

To flag words without changing them, choose tag.

source

pub fn language_model_name(&self) -> Option<&str>

Specify the name of the custom language model that you want to use when processing your transcription. Note that language model names are case sensitive.

The language of the specified language model must match the language code you specify in your transcription request. If the languages don't match, the custom language model isn't applied. There are no errors or warnings associated with a language mismatch.

For more information, see Custom language models.

source

pub fn enable_partial_results_stabilization(&self) -> Option<bool>

Enables partial result stabilization for your transcription. Partial result stabilization can reduce latency in your output, but may impact accuracy. For more information, see Partial-result stabilization.

source

pub fn partial_results_stability(&self) -> Option<&PartialResultsStability>

Specify the level of stability to use when you enable partial results stabilization (EnablePartialResultsStabilization).

Low stability provides the highest accuracy. High stability transcribes faster, but with slightly lower accuracy.

For more information, see Partial-result stabilization.

source

pub fn content_identification_type(&self) -> Option<&ContentIdentificationType>

Labels all personally identifiable information (PII) identified in your transcript.

Content identification is performed at the segment level; PII specified in PiiEntityTypes is flagged upon complete transcription of an audio segment.

You can’t set ContentIdentificationType and ContentRedactionType in the same request. If you set both, your request returns a BadRequestException.

For more information, see Redacting or identifying personally identifiable information.

source

pub fn content_redaction_type(&self) -> Option<&ContentRedactionType>

Redacts all personally identifiable information (PII) identified in your transcript.

Content redaction is performed at the segment level; PII specified in PiiEntityTypes is redacted upon complete transcription of an audio segment.

You can’t set ContentRedactionType and ContentIdentificationType in the same request. If you set both, your request returns a BadRequestException.

For more information, see Redacting or identifying personally identifiable information.

source

pub fn pii_entity_types(&self) -> Option<&str>

Specify which types of personally identifiable information (PII) you want to redact in your transcript. You can include as many types as you'd like, or you can select ALL.

To include PiiEntityTypes in your Call Analytics request, you must also include either ContentIdentificationType or ContentRedactionType.

Values must be comma-separated and can include: BANK_ACCOUNT_NUMBER, BANK_ROUTING, CREDIT_DEBIT_NUMBER, CREDIT_DEBIT_CVV, CREDIT_DEBIT_EXPIRY, PIN, EMAIL, ADDRESS, NAME, PHONE, SSN, or ALL.

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impl StartCallAnalyticsStreamTranscriptionInput

Trait Implementations§

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impl Debug for StartCallAnalyticsStreamTranscriptionInput

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fn fmt(&self, f: &mut Formatter<'_>) -> Result

Formats the value using the given formatter. Read more

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