This guide describes how to configure and use the Yandex Speech Recognition (SR) plugin to the UniMRCP server. The document is intended for users having a certain knowledge of Yandex SpeechKit Speech-to-Text API and UniMRCP.
For installation instructions, use one of the guides below.
Instructions provided in this guide are applicable to the following versions.
UniMRCP 1.6.0 and above
UniMRCP Yandex SR Plugin 1.0.0 and above
This is a brief check list of the features currently supported by the UniMRCP server running with the Yandex SR plugin.
DEFINE-GRAMMAR
RECOGNIZE
START-INPUT-TIMERS
STOP
SET-PARAMS
GET-PARAMS
RECOGNITION-COMPLETE
START-OF-INPUT
Input-Type
No-Input-Timeout
Recognition-Timeout
Speech-Complete-Timeout
Speech-Incomplete-Timeout
Waveform-URI
Media-Type
Completion-Cause
Confidence-Threshold
Start-Input-Timers
DTMF-Interdigit-Timeout
DTMF-Term-Timeout
DTMF-Term-Char
Save-Waveform
Speech-Language
Cancel-If-Queue
Sensitivity-Level
Built-in and dynamic speech contexts
Built-in/embedded DTMF grammar
SRGS XML (limited support)
The configuration file of the Yandex SR plugin is located in /opt/unimrcp/conf/umsyandexsr.xml. The configuration file is written in XML.
The root element of the XML document must be <umsyandexsr>
.
Attributes
Name | Unit | Description |
---|---|---|
license-file | File path | Specifies the license file. File name may include patterns containing '*' sign. If multiple files match the pattern, the most recent one gets used. |
subscription-key-file | File path | Specifies the Yandex SpeechKit subscription key file to use. File name may include patterns containing '*' sign. If multiple files match the pattern, the most recent one gets used. |
Parent
Children
Name | Unit | Description |
---|---|---|
streaming-recognition | String | Specifies parameters of streaming recognition employed via gRPC. |
speech-contexts | String | Contains a list of speech contexts. |
speech-dtmf-input-detector | String | Specifies parameters of the speech and DTMF input detector. |
utterance-manager | String | Specifies parameters of the utterance manager. |
rdr-manager | String | Specifies parameters of the Recognition Details Record (RDR) manager. |
monitoring-agent | String | Specifies parameters of the monitoring manager. |
license-server | String | Specifies parameters used to connect to the license server. The use of the license server is optional. |
Example
This is an example of a bare document.
<umsyandexsr license-file="umsyandexsr_*.lic"
subscription-key-file ="yandex.subscription.key">
</umsyandexsr>
This element specifies parameters of streaming recognition.
Attributes
Name | Unit | Description |
---|---|---|
folder-id | String | Specifies the Yandex SpeechKit folder identifier. Required. |
language | String | Specifies the default language to use, if not set by the client. |
model | String | Specifies the language model to use. Can be overridden by client. Available since YSR 1.4.0. |
raw-results | Boolean | Specifies whether to provide raw ("nine applies") or normalized ("9 apples") results. Can be overridden by client. Available since YSR 1.4.0. |
data-logging | Boolean | Specifies whether to enable data logging at the service side. Can be overridden by client. Available since YSR 1.5.0. |
interim-results | Boolean | Specifies whether to request interim results or not. |
start-of-input | String | Specifies the source of start of input event sent to the client (use "service-originated" for an event originated based on a first-received interim result and "internal" for an event determined by plugin). |
max-alternatives | Integer | Specifies the maximum number of speech recognition result alternatives to be returned. Can be overridden by client by means of the header field N-Best-List-Length. |
alternatives-below-threshold | Boolean | Specifies whether to return speech recognition result alternatives with the confidence score below the confidence threshold. |
confidence-format | String | Specifies the format of the confidence score to be returned (use "auto" for a format based on protocol version, "mrcpv2" for a float value in the range of 0..1, "mrcpv1" for an integer value in the range of 0..100). |
tag-format | String | Specifies the format of the instance to be returned (use "extended" to have the recorded utterance name concatenated to the result with a sign '|'). Can be overridden by client. Available since YSR 1.5.0. |
single-utterance | Boolean | Specifies whether to detect a single spoken utterance or perform continuous recognition. |
results-indent | Integer | Specifies the indentation used to compose NLSML results. |
skip-unsupported-grammars | Boolean | Specifies whether to skip or raise an error while referencing a malformed or not supported grammar. |
transcription-grammar | String | Specifies the name of the built-in speech transcription grammar. The grammar can be referenced as builtin:speech/transcribe or builtin:grammar/transcribe, where transcribe is the default value of this parameter. |
http-proxy | String | Specifies the URI of HTTP proxy, if used. Available since YSR 1.2.0. |
profanity-filter | Boolean | Specifies whether to filter out words containing profanity. Can be overridden by client. Available since YSR 1.6.0. |
match-srgs | Boolean | Specifies whether to match transcribed text specified SRGS grammar. Can be overridden by the ‘scope’ attribute in the grammar. Available since YSR 1.6.0. |
stream-creation-timeout | Time interval [msec] | Specifies how long to wait for gRPC stream creation. If timeout is set 0, no timer is used. Otherwise, if timeout is elapsed, gRPC stream creation is cancelled. Available since YSR 1.6.0. |
inter-result-timeout | Time interval [msec] | Specifies a timeout between interim results containing transcribed speech. If the timeout is elapsed, input is considered complete. The timeout defaults to 0 (disabled). Available since YSR 1.6.0. |
grpc-log-redirection | Boolean | Specifies whether to enable gRPC log redirection. Available since YSR 1.6.0. |
grpc-log-verbosity | String | Specifies gRPC logging verbosity. One of DEBUG, INFO, ERROR. See GRPC_VERBOSITY for more info. Available since YSR 1.6.0. |
grpc-log-trace | String | Specifies a comma separated list of tracers producing gRPC logs. Use 'all' to turn all tracers on. See GRPC_TRACE for more info. Available since YSR 1.6.0. |
auth-request-timeout | Time interval (sec) | Specifies a timeout in seconds set on HTTP requests placed to re-validate access token. Available since YSR 1.7.0. |
Parent
<umsyandexsr>
Children
Example
This is an example of streaming recognition element.
<streaming-recognition
folder-id="abcd1234edfg"
interim-results="true"
start-of-input="service-originated"
language="en-US"
max-alternatives="1"
alternatives-below-threshold="false"
confidence-format="auto"
single-utterance="true"
results-indent="2"
skip-unsupported-grammars="true"
transcription-grammar="transcribe"
/>
This element specifies a list of speech contexts.
Attributes
Parent
<umsyandexsr>
Children
<speech-context>
Example
The example below defines two speech contexts booking and directory.
<speech-contexts>
<speech-context id="booking" enable="true">
<phrase>I would like to book a flight from New York to Rome with a ticket eligible for free cancellation</phrase>
<phrase>I would like to book a one-way flight from New York to Rome</phrase>
</speech-context>
<speech-context id="directory" enable="true">
<phrase>call Steve</phrase>
<phrase>call John</phrase>
<phrase>dial 5</phrase>
<phrase>dial 6</phrase>
</speech-context>
</speech-contexts>
This element specifies a speech context.
Attributes
Name | Unit | Description |
---|---|---|
id | String | Specifies a unique string identifier of the speech context to be referenced by the MRCP client. |
enable | Boolean | Specifies whether the speech context is enabled or disabled. |
speech-complete | Boolean | Specifies whether to complete input as soon as an interim result matches one of the specified phrases. |
language | String | The language the phrases are defined for. |
Parent
<speech-contexts>
Children
<phrase>
Example
This is an example of speech context element.
<speech-context id="directory" enable="true">
<phrase>call Steve</phrase>
<phrase>call John</phrase>
<phrase>dial 5</phrase>
<phrase>dial 6</phrase>
</speech-context>
This element specifies a phrase in the speech context.
Attributes
Name | Unit | Description |
---|---|---|
tag | String | Specifies an optional arbitrary string identifier to be returned as an instance in the NLSML result, if the transcription result matches the phrase. |
Parent
<speech-context>
Children
This is an example of a speech context with phrases having tags specified.
<speech-context id="boolean" speech-complete="true" enable="true">
<phrase tag="true">yes</phrase>
<phrase tag="true">sure</phrase>
<phrase tag="true">correct</phrase>
<phrase tag="false">no</phrase>
<phrase tag="false">not sure</phrase>
<phrase tag="false">incorrect </phrase>
</speech-context>
This element specifies parameters of the speech and DTMF input detector.
Attributes
Name | Unit | Description |
---|---|---|
vad-mode | Integer | Specifies an operating mode of VAD in the range of [0 ... 3]. Default is 1. |
speech-start-timeout | Time interval [msec] | Specifies how long to wait in transition mode before triggering a start of speech input event. |
speech-complete-timeout | Time interval [msec] | Specifies how long to wait in transition mode before triggering an end of speech input event. The complete timeout is used when there is an interim result available. |
speech-incomplete-timeout | Time interval [msec] | Specifies how long to wait in transition mode before triggering an end of speech input event. The incomplete timeout is used as long as there is no interim result available. Afterwards, the complete timeout is used. |
noinput-timeout | Time interval [msec] | Specifies how long to wait before triggering a no-input event. |
input-timeout | Time interval [msec] | Specifies how long to wait for input to complete. |
dtmf-interdigit-timeout | Time interval [msec] | Specifies a DTMF inter-digit timeout. |
dtmf-term-timeout | Time interval [msec] | Specifies a DTMF input termination timeout. |
dtmf-term-char | Character | Specifies a DTMF input termination character. |
speech-leading-silence | Time interval [msec] | Specifies desired silence interval preceding spoken input. |
speech-trailing-silence | Time interval [msec] | Specifies desired silence interval following spoken input. |
speech-output-period | Time interval [msec] | Specifies an interval used to send speech frames to the recognizer. |
Parent
<umsyandexsr>
Children
Example
The example below defines a typical speech and DTMF input detector having the default parameters set.
<speech-dtmf-input-detector
vad-mode="2"
speech-start-timeout="300"
speech-complete-timeout="1000"
speech-incomplete-timeout="3000"
noinput-timeout="5000"
input-timeout="10000"
dtmf-interdigit-timeout="5000"
dtmf-term-timeout="10000"
dtmf-term-char=""
speech-leading-silence="300"
speech-trailing-silence="300"
speech-output-period="200"
/>
This element specifies parameters of the utterance manager.
Attributes
Name | Unit | Description |
---|---|---|
save-waveforms | Boolean | Specifies whether to save waveforms or not. |
purge-existing | Boolean | Specifies whether to delete existing records on start-up. |
max-file-age | Time interval [min] | Specifies a time interval in minutes after expiration of which a waveform is deleted. Set 0 for infinite. |
max-file-count | Integer | Specifies the max number of waveforms to store. If reached, the oldest waveform is deleted. Set 0 for infinite. |
waveform-base-uri | String | Specifies the base URI used to compose an absolute waveform URI. |
waveform-folder | Dir path | Specifies a folder the waveforms should be stored in. |
file-prefix | String | Specifies a prefix used to compose the name of the file to be stored. Defaults to 'umsyandexsr-', if not specified. |
use-logging-tag | Boolean | Specifies whether to use the MRCP header field Logging-Tag, if present, to compose the name of the file to be stored. Available since YSR 1.6.0. |
Parent
<umsyandexsr>
Children
Example
The example below defines a typical utterance manager having the default parameters set.
<utterance-manager
save-waveforms="false"
purge-existing="false"
max-file-age="60"
max-file-count="100"
waveform-base-uri="http://localhost/utterances/"
waveform-folder=""
/>
This element specifies parameters of the Recognition Details Record (RDR) manager.
Attributes
Name | Unit | Description |
---|---|---|
save-records | Boolean | Specifies whether to save recognition details records or not. |
purge-existing | Boolean | Specifies whether to delete existing records on start-up. |
max-file-age | Time interval [min] | Specifies a time interval in minutes after expiration of which a record is deleted. Set 0 for infinite. |
max-file-count | Integer | Specifies the max number of records to store. If reached, the oldest record is deleted. Set 0 for infinite. |
record-folder | Dir path | Specifies a folder to store recognition details records in. Defaults to ${UniMRCPInstallDir}/var. |
file-prefix | String | Specifies a prefix used to compose the name of the file to be stored. Defaults to 'umsyandexsr-', if not specified. |
use-logging-tag | Boolean | Specifies whether to use the MRCP header field Logging-Tag, if present, to compose the name of the file to be stored. Available since YSR 1.6.0. |
Parent
<umsyandexsr>
Children
Example
The example below defines a typical utterance manager having the default parameters set.
<rdr-manager
save-records="false"
purge-existing="false"
max-file-age="60"
max-file-count="100"
waveform-folder=""
/>
This element specifies parameters of the monitoring agent.
Attributes
Name | Unit | Description |
---|---|---|
refresh-period | Time interval [sec] | Specifies a time interval in seconds used to periodically refresh usage details. See <usage-refresh-handler> . |
Parent
<umsyandexsr>
Children
<usage-change-handler>
<usage-refresh-handler>
Example
The example below defines a monitoring agent with usage change and refresh handlers.
<monitoring-agent refresh-period="60">
<usage-change-handler>
<log-usage enable="true" priority="NOTICE"/>
</usage-change-handler>
<usage-refresh-handler>
<dump-channels enable="true" status-file="umsyandexsr-channels.status"/>
</usage-refresh-handler >
</monitoring-agent>
This element specifies an event handler called on every usage change.
Attributes
Parent
<monitoring-agent>
Children
<log-usage>
<update-usage>
<dump-channels>
Example
This is an example of the usage change event handler.
<usage-change-handler>
<log-usage enable="true" priority="NOTICE"/>
<update-usage enable="false" status-file="umsyandexsr-usage.status"/>
<dump-channels enable="false" status-file="umsyandexsr-channels.status"/>
</usage-change-handler>
This element specifies an event handler called periodically to update usage details.
Attributes
Parent
<monitoring-agent>
Children
<log-usage>
<update-usage>
<dump-channels>
Example
This is an example of the usage change event handler.
<usage-refresh-handler>
<log-usage enable="true" priority="NOTICE"/>
<update-usage enable="false" status-file="umsyandexsr-usage.status"/>
<dump-channels enable="false" status-file="umsyandexsr-channels.status"/>
</usage-refresh-handler>
This element specifies parameters used to connect to the license server.
Attributes
Name | Unit | Description |
---|---|---|
enable | Boolean | Specifies whether the use of license server is enabled or not. If enabled, the license-file attribute is not honored. |
server-address | String | Specifies the IP address or host name of the license server. |
certificate-file | File path | Specifies the client certificate used to connect to the license server. File name may include patterns containing a '*' sign. If multiple files match the pattern, the most recent one gets used. |
ca-file | File path | Specifies the certificate authority used to validate the license server. |
channel-count | Integer | Specifies the number of channels to check out from the license server. If not specified or set to 0, either all available channels or a pool of channels will be checked based on the configuration of the license server. |
http-proxy-address | String | Specifies the IP address or host name of the HTTP proxy server, if used. Available since YSR 1.6.0. |
http-proxy-port | Integer | Specifies the port number of the HTTP proxy server, if used. Available since YSR 1.6.0. |
Parent
<umsyandexsr>
Children
Example
The example below defines a typical configuration which can be used to connect to a license server located, for example, at 10.0.0.1.
<license-server
enable="true"
server-address="10.0.0.1"
certificate-file="unilic_client_*.crt"
ca-file="unilic_ca.crt"
/>
For further reference to the license server, visit
This section outlines common configuration steps.
The default configuration should be sufficient for the general use.
The Yandex SpeechKit folder identifier must be set in the configuration file umsyandexsr.xml.
Recognition language can be specified by the client per MRCP session by means of the header field Speech-Language set in a SET-PARAMS or RECOGNIZE request. Otherwise, the parameter language set in the configuration file umsyandexsr.xml is used. The parameter defaults to en-US.
Since YSR 1.6.0, the recognition language can also be set by the attribute xml:lang specified in the SRGS grammar.
<?xml version="1.0" encoding="UTF-8"?>
<grammar mode="voice" root="transcribe" version="1.0"
xml:lang="en-US"
xmlns="http://www.w3.org/2001/06/grammar">
<meta name="scope" content="builtin"/>
<rule id="transcribe"><one-of/></rule>
</grammar>
Since YSR 1.6.0, the recognition language can also be set by the optional parameter language passed to a built-in grammar.
builtin:speech/transcribe?language=en-US
Sampling rate is determined based on the SDP negotiation. Refer to the configuration guide of the UniMRCP server on how to specify supported encodings and sampling rates to be used in communication between the client and server.
The native sampling rate with the linear16 audio encoding is used in gRPC streaming to the Yandex Cloud SpeechKit service.
While the default parameters specified for the speech input detector are sufficient for the general use, various parameters can be adjusted to better suit a particular requirement.
This parameter is used to trigger a start of speech input. The shorter is the timeout, the sooner a START-OF-INPUT event is delivered to the client. However, a short timeout may also lead to a false positive.
This parameter is used to trigger an end of speech input. The shorter is the timeout, the shorter is the response time. However, a short timeout may also lead to a false positive.
Note that both events, an expiration of the speech complete timeout and an END-OF-SINGLE-UTTERANCE response delivered from the Yandex Cloud SpeechKit service, are monitored to trigger an end of speech input, on whichever comes first basis. In order to rely solely on an event delivered from the speech service, the parameter speech-complete-timeout needs to be set to a higher value.
This parameter is used to specify an operating mode of the Voice Activity Detector (VAD) within an integer range of [0 … 3]. A higher mode is more aggressive and, as a result, is more restrictive in reporting speech. The parameter can be overridden per MRCP session by setting the header field Sensitivity-Level in a SET-PARAMS or RECOGNIZE request. The following table shows how the Sensitivity-Level is mapped to the vad-mode.
Sensitivity-Level | Vad-Mode |
---|---|
[0.00 ... 0.25) | 0 |
[0.25 … 0.50) | 1 |
[0.50 ... 0.75) | 2 |
[0.75 ... 1.00] | 3 |
While the default parameters specified for the DTMF input detector are sufficient for the general use, various parameters can be adjusted to better suit a particular requirement.
This parameter is used to set an inter-digit timeout on DTMF input. The parameter can be overridden per MRCP session by setting the header field DTMF-Interdigit-Timeout in a SET-PARAMS or RECOGNIZE request.
This parameter is used to set a termination timeout on DTMF input and is in effect when dtmf-term-char is set and there is a match for an input grammar. The parameter can be overridden per MRCP session by setting the header field DTMF-Term-Timeout in a SET-PARAMS or RECOGNIZE request.
This parameter is used to set a character terminating DTMF input. The parameter can be overridden per MRCP session by setting the header field DTMF-Term-Char in a SET-PARAMS or RECOGNIZE request.
This parameter is used to trigger a no-input event. The parameter can be overridden per MRCP session by setting the header field No-Input-Timeout in a SET-PARAMS or RECOGNIZE request.
This parameter is used to limit input (recognition) time. The parameter can be overridden per MRCP session by setting the header field Recognition-Timeout in a SET-PARAMS or RECOGNIZE request.
Single Utterance Mode
By default, if the configuration parameter single-utterance is set to true, recognition is performed in the single utterance mode and is terminated upon an expiration of the speech complete timeout or an END-OF-SINGLE-UTTERANCE response delivered from the service.
Continuous Recognition Mode
In the continuous speech recognition mode, when the configuration parameter single-utterance is set to false, recognition is terminated upon an expiration of the speech complete timeout, which is recommended to be set in the range of 1500 msec to 3000 msec. The service may return multiple results (sub utterances), which are concatenated and sent back to the MRCP client in a single RECOGNITION-COMPLETE event.
The parameter single-utterance can be overridden per MRCP session by setting the header field Vendor-Specific-Parameters in a SET-PARAMS or RECOGNIZE request, where the parameter name is single-utterance and acceptable values are true and false.
The following parameters can optionally be specified by the MRCP client in SET-PARAMS, DEFINE-GRAMMAR and RECOGNIZE requests via the MRCP header field Vendor-Specific-Parameters.
Name | Unit | Description |
---|---|---|
folder-id | String | Specifies the Yandex SpeechKit folder identifier. Available since YSR 1.4.0. |
model | String | Specifies the language model to use. Available since YSR 1.4.0. |
raw-results | Boolean | Specifies whether to provide raw ("nine applies") or normalized ("9 apples") results. Available since YSR 1.4.0. |
start-of-input | String | Specifies the source of start of input event sent to the client (use "service-originated" for an event originated based on a first-received interim result and "internal" for an event determined by plugin). Available since YSR 1.4.0. |
alternatives-below-threshold | Boolean | Specifies whether to return speech recognition result alternatives with the confidence score below the confidence threshold. Available since YSR 1.4.0. |
single-utterance | Boolean | Specifies whether to detect a single spoken utterance or perform continuous recognition. Available since YSR 1.4.0. |
speech-start-timeout | Time interval [msec] | Specifies how long to wait in transition mode before triggering a start of speech input event. Available since YSR 1.4.0. |
tag-format | String | Specifies the format of the instance to be returned (use "extended" to have the recorded utterance name concatenated to the result with a sign '|'). Available since YSR 1.5.0. |
data-logging | Boolean | Specifies whether to enable data logging at the service side. Available since YSR 1.5.0. |
profanity-filter | Boolean | Specifies whether to attempt to filter out profanities. Available since YSR 1.6.0. |
interim-result-timeout | Time interval [msec] | Specifies a timeout between interim results containing transcribed speech. If the timeout is elapsed, input is considered complete. Available since YSR 1. 6.0. |
logging-tag | String | Specifies the logging tag. Available since YSR 1.6.0. |
All the vendor-specific parameters can also be specified at the grammar-level via a built-in or SRGS XML grammar.
The following example demonstrates the use of a built-in grammar with the vendor-specific parameters model and raw-results set to general and true correspondingly.
builtin:speech/transcribe?model=general;raw-results=true
The following example demonstrates the use of an SRGS XML grammar with the vendor-specific parameters model and raw-results set to general and true correspondingly.
<grammar mode="voice" root="transcribe" version="1.0" xml:lang="en-US" xmlns="http://www.w3.org/2001/06/grammar">
<meta name="scope" content="builtin"/>
<meta name="model" content="general"/>
<meta name="raw-results" content="true"/>
<rule id="transcribe">
<one-of ><item>blank</item></one-of>
</rule>
</grammar>
Saving of utterances is not required for regular operation and is disabled by default. However, enabling this functionality allows to save utterances sent to the service and later listen to them offline.
The relevant settings can be specified via the element utterance-manager.
Utterances can optionally be recorded and stored if the configuration parameter save-waveforms is set to true. The parameter can be overridden per MRCP session by setting the header field Save-Waveforms in a SET-PARAMS or RECOGNIZE request.
This parameter specifies whether to delete existing waveforms on start-up.
This parameter specifies a time interval in minutes after expiration of which a waveform is deleted. If set to 0, there is no expiration time specified.
This parameter specifies the maximum number of waveforms to store. If the specified number is reached, the oldest waveform is deleted. If set to 0, there is no limit specified.
This parameter specifies the base URI used to compose an absolute waveform URI returned in the header field Waveform-Uri in response to a RECOGNIZE request.
This parameter specifies a path to the directory used to store waveforms in. The directory defaults to ${UniMRCPInstallDir}/var.
Producing of recognition details records (RDR) is not required for regular operation and is disabled by default. However, enabling this functionality allows to store details of each recognition attempt in a separate file and analyze them later offline. The RDRs ate stored in the JSON format.
The relevant settings can be specified via the element rdr-manager.
This parameter specifies whether to save recognition details records or not.
This parameter specifies whether to delete existing records on start-up.
This parameter specifies a time interval in minutes after expiration of which a record is deleted. If set to 0, there is no expiration time specified.
This parameter specifies the maximum number of records to store. If the specified number is reached, the oldest record is deleted. If set to 0, there is no limit specified.
This parameter specifies a path to the directory used to store records in. The directory defaults to ${UniMRCPInstallDir}/var.
Pre-set built-in speech contexts can be referenced by the MRCP client in a RECOGNIZE request as follows:
builtin:speech/$id
Where $id is a unique string identifier of built-in speech context.
Speech contexts are defined in the configuration file umsyandexsr.xml. A speech context is assigned a unique string identifier and holds a list of phrases.
Below is a definition of a sample speech context directory:
<speech-context id="directory">
<phrase>call Steve</phrase>
<phrase>call John</phrase>
<phrase>dial 5</phrase>
<phrase>dial 6</phrase>
</speech-context>
Which can be referenced in a RECOGNIZE request as follows:
builtin:speech/directory
The prefixes builtin:speech and builtin:grammar can be used interchangeably as follows:
builtin:grammar/directory
For generic speech transcription, having no speech contexts defined, a pre-set identifier transcribe must be used.
builtin:speech/transcribe
The name of the identifier transcribe can be changed from the configuration file umsyandexsr.xml.
The MRCP client can also dynamically specify a speech context either
in a DEFINE-GRAMMAR request by further referencing the defined speech context in a RECOGNIZE request using the session URI scheme
or inline in a RECOGNIZE request
While composing a DEFINE-GRAMMAR or RECOGNIZE request containing speech context definition, the following should be considered.
The value of the header field Content-Id must be used as a unique string identifier of the speech context being defined.
The value of the header field Content-Type must be set to application/xml.
The message body must contain a definition of the speech context, composed based on the XML format of the element <speech-context>, specified in the configuration file umsyandexsr.xml. Note that the unique identifier of the speech context is set based on the header field Content-Id, as opposed to the attribute Id when loading from configuration.
Pre-set built-in DTMF grammars can be referenced by the MRCP client in a RECOGNIZE request as follows:
builtin:dtmf/$id
Where $id is a unique string identifier of the built-in DTMF grammar. For example:
builtin:dtmf/digits
Note that only a DTMF grammar identifier digits is currently supported.
Results received from the Yandex Cloud SpeechKit service are transformed to the NLSML format with no semantic interpretation performed and sent to the MRCP client in a RECOGNITION-COMPLETE event.
The number of in-use and total licensed channels can be monitored in several alternate ways. There is a set of actions which can take place on certain events. The behavior is configurable via the element monitoring-agent, which contains two event handlers: usage-change-handler and usage-refresh-handler.
While the usage-change-handler is invoked on every acquisition and release of a licensed channel, the usage-refresh-handler is invoked periodically on expiration of a timeout specified by the attribute refresh-period.
The following actions can be specified for either of the two handlers.
The action log-usage logs the following data in the order specified.
The number of currently in-use channels.
The maximum number of channels used concurrently.
The total number of licensed channels.
The following is a sample log statement, indicating 0 in-use, 0 max-used and 2 total channels.
[NOTICE] YandexSR Usage: 0/0/2
The action update-usage writes the following data to a status file umsyandexsr-usage.status, located by default in the directory ${UniMRCPInstallDir}/var/status.
The number of currently in-use channels.
The maximum number of channels used concurrently.
The total number of licensed channels.
The current status of the license permit.
The license server alarm. Set to on, if the license server is not available for more than one hour; otherwise, set to off. This parameter is maintained only if the license server is used. Available since YSR 1.2.0.
The following is a sample content of the status file.
in-use channels: 0
max used channels: 0
total channels: 2
license permit: true
licserver alarm: off
The action dump-channels writes the identifiers of in-use channels to a status file umsyandexsr-channels.status, located by default in the directory ${UniMRCPInstallDir}/var/status.
This example demonstrates how to perform speech recognition by using a RECOGNIZE request, having no speech contexts defined.
C->S:
MRCP/2.0 336 RECOGNIZE 1
Channel-Identifier: 6e1a2e4e54ae11e7@speechrecog
Content-Id: request1@form-level
Content-Type: text/uri-list
Cancel-If-Queue: false
No-Input-Timeout: 5000
Recognition-Timeout: 10000
Start-Input-Timers: true
Confidence-Threshold: 0.87
Save-Waveform: true
Content-Length: 25
builtin:speech/transcribe
S->C:
MRCP/2.0 83 1 200 IN-PROGRESS
Channel-Identifier: 6e1a2e4e54ae11e7@speechrecog
S->C:
MRCP/2.0 115 START-OF-INPUT 1 IN-PROGRESS
Channel-Identifier: 6e1a2e4e54ae11e7@speechrecog
Input-Type: speech
S->C:
MRCP/2.0 498 RECOGNITION-COMPLETE 1 COMPLETE
Channel-Identifier: 6e1a2e4e54ae11e7@speechrecog
Completion-Cause: 000 success
Waveform-Uri: <http://localhost/utterances/utter-6e1a2e4e54ae11e7-1.wav>;size=20480;duration=1280
Content-Type: application/x-nlsml
Content-Length: 214
<?xml version="1.0"?>
<result>
<interpretation grammar="builtin:speech/transcribe" confidence="1">
<instance>Dial 5</instance>
<input mode="speech">Dial 5</input>
</interpretation>
</result>
This example demonstrates how to perform speech recognition by using a RECOGNIZE request to reference a pre-set built-in speech context directory on the server.
C->S:
MRCP/2.0 335 RECOGNIZE 1
Channel-Identifier: 3ea18b9854af11e7@speechrecog
Content-Id: request1@form-level
Content-Type: text/uri-list
Cancel-If-Queue: false
No-Input-Timeout: 5000
Recognition-Timeout: 10000
Start-Input-Timers: true
Confidence-Threshold: 0.87
Save-Waveform: true
Content-Length: 24
builtin:speech/directory
S->C:
MRCP/2.0 83 1 200 IN-PROGRESS
Channel-Identifier: 3ea18b9854af11e7@speechrecog
S->C:
MRCP/2.0 115 START-OF-INPUT 1 IN-PROGRESS
Channel-Identifier: 3ea18b9854af11e7@speechrecog
Input-Type: speech
S->C:
MRCP/2.0 497 RECOGNITION-COMPLETE 1 COMPLETE
Channel-Identifier: 3ea18b9854af11e7@speechrecog
Completion-Cause: 000 success
Waveform-Uri: <http://localhost/utterances/utter-3ea18b9854af11e7-1.wav>;size=20480;duration=1280
Content-Type: application/x-nlsml
Content-Length: 213
<?xml version="1.0"?>
<result>
<interpretation grammar="builtin:speech/directory" confidence="1">
<instance>call Steve</instance>
<input mode="speech">call Steve</input>
</interpretation>
</result>
This examples demonstrates how to perform speech recognition, by using a DEFINE-GRAMMAR request to specify a speech context and further reference the defined speech context in a RECOGNIZE request.
C->S:
MRCP/2.0 314 DEFINE-GRAMMAR 1
Channel-Identifier: 25902c3a54b011e7@speechrecog
Content-Type: application/xml
Content-Id: request1@form-level
Content-Length: 146
<speech-context>
<phrase>call Steve</phrase>
<phrase>call John</phrase>
<phrase>dial 5</phrase>
<phrase>dial 6</phrase>
</speech-context>
S->C:
MRCP/2.0 112 1 200 COMPLETE
Channel-Identifier: 25902c3a54b011e7@speechrecog
Completion-Cause: 000 success
C->S:
MRCP/2.0 305 RECOGNIZE 2
Channel-Identifier: 25902c3a54b011e7@speechrecog
Content-Type: text/uri-list
Cancel-If-Queue: false
No-Input-Timeout: 5000
Recognition-Timeout: 10000
Start-Input-Timers: true
Confidence-Threshold: 0.87
Save-Waveform: true
Content-Length: 27
session:request1@form-level
S->C:
MRCP/2.0 83 2 200 IN-PROGRESS
Channel-Identifier: 25902c3a54b011e7@speechrecog
S->C:
MRCP/2.0 115 START-OF-INPUT 2 IN-PROGRESS
Channel-Identifier: 25902c3a54b011e7@speechrecog
Input-Type: speech
S->C:
MRCP/2.0 500 RECOGNITION-COMPLETE 2 COMPLETE
Channel-Identifier: 25902c3a54b011e7@speechrecog
Completion-Cause: 000 success
Waveform-Uri: <http://localhost/utterances/utter-25902c3a54b011e7-2.wav>;size=20480;duration=1280
Content-Type: application/x-nlsml
Content-Length: 216
<?xml version="1.0"?>
<result>
<interpretation grammar="session:request1@form-level" confidence="1">
<instance>call Steve</instance>
<input mode="speech">call Steve</input>
</interpretation>
</result>
This example demonstrates how to reference a built-in DTMF grammar in a RECOGNIZE request.
C->S:
MRCP/2.0 266 RECOGNIZE 1
Channel-Identifier: d26bef74091a174c@speechrecog
Content-Type: text/uri-list
Cancel-If-Queue: false
Start-Input-Timers: true
Confidence-Threshold: 0.7
Speech-Language: en-US
Dtmf-Term-Char: #
Content-Length: 19
builtin:dtmf/digits
S->C:
MRCP/2.0 83 1 200 IN-PROGRESS
Channel-Identifier: d26bef74091a174c@speechrecog
S->C:
MRCP/2.0 113 START-OF-INPUT 1 IN-PROGRESS
Channel-Identifier: d26bef74091a174c@speechrecog
Input-Type: dtmf
S->C:
MRCP/2.0 382 RECOGNITION-COMPLETE 1 COMPLETE
Channel-Identifier: d26bef74091a174c@speechrecog
Completion-Cause: 000 success
Content-Type: application/x-nlsml
Content-Length: 197
<?xml version="1.0"?>
<result>
<interpretation grammar="builtin:dtmf/digits" confidence="1.00">
<input mode="dtmf">1 2 3 4</input>
<instance>1234</instance>
</interpretation>
</result>
This example demonstrates how to perform recognition by activating both speech and DTMF grammars. In this example, the user is expected to input a 4-digit pin.
C->S:
MRCP/2.0 275 RECOGNIZE 1
Channel-Identifier: 6ae0f23e1b1e3d42@speechrecog
Content-Type: text/uri-list
Cancel-If-Queue: false
Start-Input-Timers: true
Confidence-Threshold: 0.7
Speech-Language: en-US
Content-Length: 47
builtin:dtmf/digits?length=4
builtin:speech/pin
S->C:
MRCP/2.0 83 2 200 IN-PROGRESS
Channel-Identifier: 6ae0f23e1b1e3d42@speechrecog
S->C:
MRCP/2.0 115 START-OF-INPUT 2 IN-PROGRESS
Channel-Identifier: 6ae0f23e1b1e3d42@speechrecog
Input-Type: speech
S->C:
MRCP/2.0 399 RECOGNITION-COMPLETE 2 COMPLETE
Channel-Identifier: 6ae0f23e1b1e3d42@speechrecog
Completion-Cause: 000 success
Content-Type: application/x-nlsml
Content-Length: 214
<?xml version="1.0"?>
<result>
<interpretation grammar=" builtin:speech/pin" confidence="1.00">
<instance>one two three four</instance>
<input mode="speech">one two three four</input>
</interpretation>
</result>
The following sequence diagrams outline common interactions between all the main components involved in a typical recognition session performed over MRCPv1 and MRCPv2 respectively.
All the data transmitted to and received from the Yandex Cloud SpeechKit API is carried over a secure TLS v1.2 connection via the gRPC streaming.
The standard TLS port 443 is used for the gRPC streaming.