New continuous recognition option

In latest Voicegain release (1.16.0) we have added a new option to our /asr/recognize/async API for ASR/speech-to-text. It is called continuousRecognition and if enabled modifies the default behavior of the grammar-based recognition.

Normally when /asr/recognize/async API is used the recognizer will return when the grammar is matched and the complete timeout expires. That means that it is only possible to get a single recognition in one /asr/recognize/async API request. If a no-match or no-input is detected the recognition will terminate.

However, sometimes there are use cases which demand that the recognizer e.g. ignores all no-matches until a match is found. This is what the continuousRecognition option is for.

With continuousRecognition you have fine control over which of the 4 events - no-input, no-match, match, and error - will be returned in a callback and which (if any) event will terminate recognition. If you do not set any event to terminate recogntion, the recognition session can be stopped by closing the audio stream or by returning stop:true from the callback.

What is it good for?

An example might be a use case where a voicemail is being played to a caller and during the playback we want to interpret caller commands like: stop, next, previous, save, delete. If we used normal recognition we would encounter situations where what is said was not understood. Stopping recognition on no-match would not make much sense because either: (1) re-prompting would mess up the flow of the call, or (2) restarting recognition  might introduce a gap in recognition that may result in missing a part what the caller said.

In scenario like this it is best to ignore no-match and continue to listen, the caller will notice no response to what he said and will naturally repeat that.

The settings for continuous recognition that would work in this case would be:

• stopOn : match, error
• noCallbackFor : no-input, no-match - notes: (1) in this case we suggest setting a noinputTimeout very long so that internally no no-inputs are generated, (2) application could also decide to accept no-match callbacks - they could be tracked and if too numerous acted upon.
  

Continuous Recognition is supported in Voicegain integration for Twilio Media Streams - either TwiML <Stream> or <Connect><Stream> in Twilio Programmable Voice

It is not yet supported in Voicegain Telephony Bot APIs.

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Many of our customers have been asking us for help in benchmarking Voicegain speech-to-text recognizer (ASR) on their specific audio files. To make this benchmarking easier we have released a python script that accomplishes just that. With a single command line you can transcribe all audio files from the input directory and compare them against reference transcripts - calculating the WER for each file. You can also do a 2-way comparison of reference vs Voicegain transcript vs Google Speech-to-Text transcript.

The script and the documentation is available at: https://github.com/voicegain/platform/tree/master/utility-scripts/test-transcribe

See our benchmark blog post to give you an idea of what kind of accuracy to expect from the Voicegain recognizer.

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Updated: Feb 28 2022

In this blog post we describe two case studies to illustrate improvements in speech-to-text or ASR recognition accuracy that can be expected from training of the underlying acoustic models.  We trained our acoustic model to recognize Indian and Irish English better.

Case study setup

The Voicegain out-of-the-box Acoustic Model which is available as default on the Voicegain Platform had been trained to recognize mainly US English although our training data set did contain some British English audio. The training data did not contain Indian and Irish English, except for maybe accidental occurrences.

Both case studies were performed in an identical manner:

• Training data contained about 300 hours of transcribed speech audio.
• Training was done to get improved accuracy on the new type of data but at the same time to also retain the baseline accuracy. An alternative would have been to aim for maximum improvement on the new data at expense of accuracy of the baseline model.
• Training was stopped after significant improvement was achieved. It could have been continued to achieve further improvement, although that might have been marginal.
• Benchmarks presented here were done on data that was not included in the training set.

Case Study 1: Indian English

Here are the parameters of this study.

• We had 250 hours of audio containing male and female speakers, each speaker reading about 50 minutes worth of speech audio.
• We separated 6 speakers for the benchmark, selecting 3 male and 3 female samples. Samples were selected to contain both easy, medium, and difficult test cases.

Here are the results of the benchmark before and after training. For comparison. we also include results from Google Enhanced Speech-to-Text.

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Some observations:

• All 6 test speakers show significant improvement over the original accuracy.
• After training the accuracy of 5 speakers is better than Google Enhanced Speech-to-Text. The one remaining speaker improved a lot (from 62% to 76%) but the accuracy was still not as good as Google. We examined the audio and it turns out that it was not recorded properly. The speaker was speaking very quietly and the microphone gain was set very high - this resulted in the audio containing a lot of strange artifacts, like e.g. tongue clicking. The speaker also ready the text in a very unnatural "mechanical" way. Kudos to Google for doing so well on such a bad recording.
• On average custom-trained Voicegain speech-to-text was better by about 2% on our Indian English benchmark compared to Google Enhanced recognizer.

Case Study 2: Irish English

Here are the parameters of this study.

• We collected about 350 hours of transcribed speech audio from one speaker from Northern Ireland.
• For the benchmark we retained some audio from that speaker that was not used for training plus we found audio from 5 other speakers with various types of Irish English accents.  

Here are the results of the benchmark before and after training. We also include results from Google Enhanced Speech-to-Text.

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Some observations:

• The speaker that was used for training is labeled here as 'Legge'. We see huge improvement after training from 76.2% to 88.5% which is significantly above Google Enhanced with 83.9%
• The other speaker with over 10% improvement is 'Lucas' which has a very similar accent to 'Legge'.
• We looked in detail at the audio  of the speaker labeled 'Cairns' who had the least improvement and for whom Google was better than our custom trained recognizer. The audio has significantly lower quality that the other samples plus it contains noticeable echo. Its audio characteristics are quite different from that audio characteristics of the training data used.
• On average custom trained Voicegain speech-to-text was better by about 1% on our Irish English benchmark compared to Google Enhanced recognizer.

Further Observations

• The amount of data used in training at 250-350 hours was not large given that normally Acoustic Models for speech recognition are trained on 10s of thousands of hours of audio.
• The large improvement on 'Legge' speaker suggest that if the goal is to improve recognition on very specific type of speech or speaker the training set could be lower, maybe 50 to 100 hours, to achieve significant improvement.  
• Bigger training set may be needed - 500 hours or more - in cases where the variability of speech and other audio characteristics is large.

UPDATE Feb 2022

We have published 2 additional studies showing the benefits of Acoustic Model training:

• Getting high Speech Recognition Accuracy on Alphanumeric Sequences: A Case Study with UK Zip Codes
• Model Training Delivers Huge Gains in Accuracy - Case Study: Indian Food Bot

Interested in Voicegain? Take us for a test drive!

1. Click here for instructions to access our live demo site.

2. If you are building a cool voice app and you are looking to test our APIs, click hereto sign up for a developer account  and receive $50 in free credits

3. If you want to take Voicegain as your own AI Transcription Assistant to meetings, click here.

In our previous post we described how Voicegain is providing grammar-based speech recognition to Twilio Programmable Voice platform via the Twilio Media Stream Feature.

Starting from release 1.16.0 of Voicegain Platform and API it possible to use Voicegain speech-to-text for speech transcription (without grammars) to achieve functionality like using TwiML <Gather>.

The reasons we think it will be attractive to Twilio users are:

• lower cost per each speech-to-text capture
• higher accuracy for customers who choose Acoustic Model customization
• access to all speech-to-text hypotheses in word-tree output mode
  

Using Voicegain as an alternative to <Gather> will have similar steps to using Voicegain for grammar-based recognition - these are listed below.

Initiating Speech Transcription with Voicegain

This is done by invoking Voicegain async transcribe API: /asr/transcribe/async

Below is an example of the payload needed to start a new transcription session:

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Some notes about the content of the request:

• we are requesting the callback to return transcript in text form - other options are possible like words (individual words with confidences) and word-tree (words organized in a tree of recognition hypotheses)
• startInputTimers tells ASR to delay start of timers - they will be started later when the question prompt finishes playing
• TWIML is set as the streaming protocol with the format set to PCMU (u-law) and sample rate of 8kHz
• asr settings include the two timeouts used in transcription - no-input, and complete timeouts.

This request, if successful, will return the websocket url in the audio.stream.websocketUrl field. This value will be used in making a TwiML request.

Note, in the transcribe mode DTMF detection is currently not possible. Please let us know if this is something that would be critical to your use case.

TwiML <Connect><Stream> request

After we have initiated a Voicegain ASR session, we can tell Twilio to open Media Streams connection to Voicegain. This is done by means of the following TwiML request:

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Some notes about the content of the TwiML request:

• the websocket URL is the one returned from Voicegain /asr/transcribe/async request
• more than one question prompt is supported - they will be played one after another
• three types of prompts are supported: 01) recording retrieved from a URL, 02) TTS prompt (several voices are available), 03) 'clip:' prompt generated   using Voicegain Prompt Manager which supports dynamic concatenation of prerecorded prompts
• bargeIn is enabled - prompt playback will stop as soon as caller starts speaking

Returned Transcription Response

Below is an example response from the transcription  in case where "content" : {"full" : ["transcript"] } .

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We want to share a short video showing live transcription in action at CBC. This one is using our baseline Acoustic Model. No customizations were made, no hints used. This video gives an idea of what latency is achievable with real-time transcription.

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Automated real-time transcription is a great solution for accommodating hearing impaired if no sign-language interpreter is available. I can be used, e.g., at churches to transcribe sermons, at conventions and meetings to transcribe talks, at educational institutions (schools, universities) to live transcribe lessons and lectures, etc.

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Voicegain Platform provides a complete stack to support live transcription:

• Utility for audio capture at source
• Cloud based or On-Prem transcription engine and API
• Web portal for controlling multiple simultaneous live transcriptions
• Web-based viewer app to enable following the transcription on any device with web browser. This app can also be embedded into any web page.

Very high accuracy - above that provided by Google, Amazon, and Microsoft Cloud speech-to-text - can be achieved through Acoustic Model customization.

Voicegain adds grammar-based speech recognition to Twilio Programmable Voice platform via the Twilio Media Stream Feature.

The difference between Voicegain speech recognition and Twilio TwiML <Gather> is:

1. Voicegain supports grammars with semantic tags (GRXML or JSGF) while <Gather> is a large vocabulary recognizer that just returns text, and
2. Voicegain is  significantly cheaper (we will describe the price difference in an upcoming blog post).

When using Voicegain with Twilio, your application logic will need to handle callback requests from both Twilio and Voicegain.

Each recognition will involve two main steps described below:

Initiating Speech Recognition with Voicegain

This is done by invoking Voicegain async recognition API: /asr/recognize/async

Below is an example of the payload needed to start a new recognition session:

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Some notes about the content of the request:

• startInputTimers tells ASR to delay start of timers - they will be started later when the question prompt finishes playing
• TWIML is set as the streaming protocol with the format set to PCMU (u-law) and sample rate of 8kHz
• asr settings include the three standard timeouts used in grammar based recognition - no-input, complete, and incomplete timeouts
• grammar is set to GRXML grammar loaded from an external URL

This request, if successful, will return the websocket url in the audio.stream.websocketUrl field. This value will be used in making a TwiML request.

Note, if the grammar is specified to recognize DTMF, the Voicegain recognizer will recognize DTMF signals included in the audio sent from Twilio Platform.

TwiML <Connect><Stream> request

After we have initiated a Voicegain ASR session, we can tell Twilio to open Media Streams connection to Voicegain. This is done by means of the following TwiML request:

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Some notes about the content of the TwiML request:

• the websocket URL is the one returned from Voicegain /asr/recognize/async request
• more than one question prompt is supported - they will be played one after another
• three types of prompts are supported: 01) recording retrieved from a URL, 02) TTS prompt (several voices are available), 03) 'clip:' prompt generated   using Voicegain Prompt Manager which supports dynamic concatenation of prerecorded prompts
• bargeIn is enabled - prompt playback will stop as soon as caller starts speaking

Returned Recognition Response

Below is an example response from the recognition. This response is from built-in phone grammar.

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