Merge pull request #403 from eric-urban/eur/speech-refresh

remove luis doc and refresh

Author: Stacyrch140

articles/ai-services/.openpublishing.redirection.ai-services.json

@@ -30,6 +30,11 @@
       "redirect_url": "/azure/ai-services/language-service/conversational-language-understanding/how-to/migrate-from-luis",
       "redirect_document_id": false
     },
+    {
+      "source_path_from_root": "/articles/ai-services/luis/luis-concept-data-conversion.md",
+      "redirect_url": "/azure/ai-services/language-service/conversational-language-understanding/how-to/migrate-from-luis",
+      "redirect_document_id": false
+    },
     {
       "source_path_from_root": "/articles/ai-services/custom-vision-service/update-application-to-3.0-sdk.md",
       "redirect_url": "/azure/ai-services/custom-vision-service/overview",
@@ -405,6 +410,11 @@
       "redirect_url": "/azure/ai-services/speech-service/release-notes",
       "redirect_document_id": false
     },
+    {
+      "source_path_from_root": "/articles/ai-services/speech-service/how-to-recognize-intents-from-speech-csharp.md",
+      "redirect_url": "/azure/ai-services/speech-service/intent-recognition",
+      "redirect_document_id": false
+    },
     {
       "source_path_from_root": "/articles/ai-services/anomaly-detector/how-to/postman.md",
       "redirect_url": "/azure/ai-services/anomaly-detector/overview",

articles/ai-services/luis/faq.md

@@ -24,10 +24,6 @@ LUIS has several limit areas. The first is the model limit, which controls inten
 
 An authoring resource lets you create, manage, train, test, and publish your applications. A prediction resource lets you query your prediction endpoint beyond the 1,000 requests provided by the authoring resource. See [Authoring and query prediction endpoint keys in LUIS](luis-how-to-azure-subscription.md) to learn about the differences between the authoring key and the prediction runtime key.
 
-## Does LUIS support speech to text?
-
-Yes, [Speech](../speech-service/how-to-recognize-intents-from-speech-csharp.md#luis-and-speech) to text is provided as an integration with LUIS.
-
 ## What are Synonyms and word variations?
 
 LUIS has little or no knowledge of the broader _NLP_ aspects, such as semantic similarity, without explicit identification in examples. For example, the following tokens (words) are three different things until they're used in similar contexts in the examples provided:

articles/ai-services/luis/luis-concept-data-conversion.md

@@ -95,10 +95,6 @@ Use the _kind_, `LUIS`, when filtering resources in the Azure portal.The LUIS qu
 
 [Sentiment analysis integration](how-to/publish.md), which provides sentiment information, is provided without requiring another Azure resource.
 
-### Speech integration
-
-[Speech integration](../speech-service/how-to-recognize-intents-from-speech-csharp.md) provides 1 thousand endpoint requests per unit cost.
-
 [Learn more about pricing.][pricing]
 
 ## Keyboard controls

articles/ai-services/luis/luis-limits.md

@@ -74,8 +74,6 @@ items:
     items:
     - name: With Bing Spell Check v7
       href: luis-tutorial-bing-spellcheck.md
-    - name: With Speech service
-      href: ../speech-service/how-to-recognize-intents-from-speech-csharp.md?toc=/azure/ai-services/luis/toc.json&bc=/azure/ai-services/luis/breadcrumb/toc.json
     - name: With LUIS and question answering using orchestration
       href: how-to/orchestration-projects.md
   - name: Migrate to conversational language understanding
@@ -157,8 +155,6 @@ items:
       href: luis-concept-data-alteration.md
     - name: Data retention
       href: luis-concept-data-storage.md
-    - name: Data conversion
-      href: luis-concept-data-conversion.md
     - name: Data extraction
       href: luis-concept-data-extraction.md
   - name: Security

articles/ai-services/luis/toc.yml

@@ -6,8 +6,9 @@ author: eric-urban
 manager: nitinme
 ms.service: azure-ai-speech
 ms.topic: how-to
-ms.date: 1/21/2024
+ms.date: 9/20/2024
 ms.author: eur
+#Customer intent: As a developer, I want to learn about the data types that I can use to train a custom neural voice.
 ---
 
 # Training data for custom neural voice
@@ -28,8 +29,8 @@ This table lists data types and how each is used to create a custom Text to spee
 | Data type | Description | When to use | Extra processing required |
 | --------- | ----------- | ----------- | ------------------------------ |
 | [Individual utterances + matching transcript](#individual-utterances--matching-transcript) | A collection (.zip) of audio files (.wav) as individual utterances. Each audio file should be 15 seconds or less in length, paired with a formatted transcript (.txt). | Professional recordings with matching transcripts | Ready for training. |
-| [Long audio + transcript](#long-audio--transcript-preview) | A collection (.zip) of long, unsegmented audio files (.wav or .mp3, longer than 20 seconds, at most 1000 audio files), paired with a collection (.zip) of transcripts that contains all spoken words. | You have audio files and matching transcripts, but they aren't segmented into utterances. | Segmentation (using batch transcription).<br>Audio format transformation wherever required. |
-| [Audio only (Preview)](#audio-only-preview) | A collection (.zip) of audio files (.wav or .mp3, at most 1000 audio files) without a transcript. | You only have audio files available, without transcripts. | Segmentation + transcript generation (using batch transcription).<br>Audio format transformation wherever required.|
+| [Long audio + transcript](#long-audio--transcript-preview) | A collection (.zip) of long, unsegmented audio files (.wav or .mp3, longer than 20 seconds, at most 1,000 audio files), paired with a collection (.zip) of transcripts that contains all spoken words. | You have audio files and matching transcripts, but they aren't segmented into utterances. | Segmentation (using batch transcription).<br>Audio format transformation wherever required. |
+| [Audio only (Preview)](#audio-only-preview) | A collection (.zip) of audio files (.wav or .mp3, at most 1,000 audio files) without a transcript. | You only have audio files available, without transcripts. | Segmentation + transcript generation (using batch transcription).<br>Audio format transformation wherever required.|
 
 Files should be grouped by type into a dataset and uploaded as a zip file. Each dataset can only contain a single data type.
 
@@ -107,12 +108,12 @@ Follow these guidelines when preparing audio for segmentation.
 | Sample format |RIFF(.wav): PCM, at least 16-bit.<br/><br/>mp3: At least 256 KBps bit rate.|
 | Audio length | Longer than 20 seconds |
 | Archive format | .zip |
-| Maximum archive size | 2048 MB, at most 1000 audio files included |
+| Maximum archive size | 2048 MB, at most 1,000 audio files included |
 
 > [!NOTE]
 > The default sampling rate for a custom neural voice is 24,000 Hz. Audio files with a sampling rate lower than 16,000 Hz will be rejected. Your audio files with a sampling rate higher than 16,000 Hz and lower than 24,000 Hz will be up-sampled to 24,000 Hz to train a neural voice. It's recommended that you should use a sample rate of 24,000 Hz for your training data.
 
-All audio files should be grouped into a zip file. It's OK to put .wav files and .mp3 files into the same zip file. For example, you can upload a 45 second audio file named 'kingstory.wav' and a 200 second long audio file named 'queenstory.mp3' in the same zip file. All .mp3 files will be transformed into the .wav format after processing.
+All audio files should be grouped into a zip file. It's OK to put .wav files and .mp3 files into the same zip file. For example, you can upload a 45-second audio file named 'kingstory.wav' and a 200-second long audio file named 'queenstory.mp3' in the same zip file. All .mp3 files will be transformed into the .wav format after processing.
 
 ### Transcription data for Long audio + transcript
 
@@ -126,7 +127,7 @@ Transcripts must be prepared to the specifications listed in this table. Each au
 | # of utterances per line | No limit |
 | Maximum file size | 2048 MB |
 
-All transcripts files in this data type should be grouped into a zip file. For example, you might upload a 45 second audio file named 'kingstory.wav' and a 200 second long audio file named 'queenstory.mp3' in the same zip file. You need to upload another zip file containing the corresponding two transcripts--one named 'kingstory.txt' and the other one named 'queenstory.txt'. Within each plain text file, you provide the full correct transcription for the matching audio.
+All transcripts files in this data type should be grouped into a zip file. For example, you might upload a 45-second audio file named 'kingstory.wav' and a 200-second long audio file named 'queenstory.mp3' in the same zip file. You need to upload another zip file containing the corresponding two transcripts--one named 'kingstory.txt' and the other one named 'queenstory.txt'. Within each plain text file, you provide the full correct transcription for the matching audio.
 
 After your dataset is successfully uploaded, we'll help you segment the audio file into utterances based on the transcript provided. You can check the segmented utterances and the matching transcripts by downloading the dataset. Unique IDs are assigned to the segmented utterances automatically. It's important that you make sure the transcripts you provide are 100% accurate. Errors in the transcripts can reduce the accuracy during the audio segmentation and further introduce quality loss in the training phase that comes later.
 
@@ -150,7 +151,7 @@ Follow these guidelines when preparing audio.
 | Sample format |RIFF(.wav): PCM, at least 16-bit<br>mp3: At least 256 KBps bit rate.|
 | Audio length | No limit |
 | Archive format | .zip |
-| Maximum archive size | 2048 MB, at most 1000 audio files included |
+| Maximum archive size | 2048 MB, at most 1,000 audio files included |
 
 > [!NOTE]
 > The default sampling rate for a custom neural voice is 24,000 Hz. Your audio files with a sampling rate higher than 16,000 Hz and lower than 24,000 Hz will be up-sampled to 24,000 Hz to train a neural voice. It's recommended that you should use a sample rate of 24,000 Hz for your training data.

articles/ai-services/speech-service/how-to-custom-voice-training-data.md

@@ -2,12 +2,14 @@
 title: How to get speech to text session ID and transcription ID
 titleSuffix: Azure AI services
 description: Learn how to get speech to text session ID and transcription ID
-author: alexeyo26
+author: eric-urban
+ms.author: eur
 manager: nitinme
 ms.service: azure-ai-speech
 ms.topic: how-to
-ms.date: 1/21/2024
-ms.author: alexeyo 
+ms.date: 9/20/2024
+ms.reviewer: alexeyo
+#Customer intent: As a developer, I need to know how to get the session ID and transcription ID for speech to text so that I can debug issues with my application.
 ---
 
 # How to get speech to text session ID and transcription ID
@@ -68,7 +70,7 @@ spx help translate log
 
 Unlike Speech SDK, [Speech to text REST API for short audio](rest-speech-to-text-short.md) doesn't automatically generate a Session ID. You need to generate it yourself and provide it within the REST request.
 
-Generate a GUID inside your code or using any standard tool. Use the GUID value *without dashes or other dividers*. As an example we'll use `9f4ffa5113a846eba289aa98b28e766f`.
+Generate a GUID inside your code or using any standard tool. Use the GUID value *without dashes or other dividers*. As an example we use `9f4ffa5113a846eba289aa98b28e766f`.
 
 As a part of your REST request use `X-ConnectionId=<GUID>` expression. For our example, a sample request looks like this:
 ```http

articles/ai-services/speech-service/how-to-get-speech-session-id.md

@@ -7,16 +7,16 @@ ms.author: eur
 manager: nitinme
 ms.service: azure-ai-speech
 ms.topic: how-to
-ms.date: 1/21/2024
+ms.date: 9/20/2024
 ms.reviewer: yulili
 ms.custom: references_regions, devx-track-extended-java, devx-track-python
 zone_pivot_groups: programming-languages-set-nineteen
+#Customer intent: As a developer, I need to know how to lower speech synthesis latency using Speech SDK so that I can improve the performance of my application.
 ---
 
 # Lower speech synthesis latency using Speech SDK
 
-The synthesis latency is critical to your applications.
-In this article, we'll introduce the best practices to lower the latency and bring the best performance to your end users.
+In this article, we introduce the best practices to lower the text to speech synthesis latency and bring the best performance to your end users.
 
 Normally, we measure the latency by `first byte latency` and `finish latency`, as follows:
 
@@ -298,12 +298,12 @@ SPXConnection* connection = [[SPXConnection alloc]initFromSpeechSynthesizer:synt
 ::: zone-end
 
 > [!NOTE]
-> If the synthesize text is available, just call `SpeakTextAsync` to synthesize the audio. The SDK will handle the connection.
+> If the text is available, just call `SpeakTextAsync` to synthesize the audio. The SDK will handle the connection.
 
 ### Reuse SpeechSynthesizer
 
 Another way to reduce the connection latency is to reuse the `SpeechSynthesizer` so you don't need to create a new `SpeechSynthesizer` for each synthesis.
-We recommend using object pool in service scenario, see our sample code for [C#](https://github.com/Azure-Samples/cognitive-services-speech-sdk/blob/master/samples/csharp/sharedcontent/console/speech_synthesis_server_scenario_sample.cs) and [Java](https://github.com/Azure-Samples/cognitive-services-speech-sdk/blob/master/samples/java/jre/console/src/com/microsoft/cognitiveservices/speech/samples/console/SpeechSynthesisScenarioSamples.java).
+We recommend using object pool in service scenario. See our sample code for [C#](https://github.com/Azure-Samples/cognitive-services-speech-sdk/blob/master/samples/csharp/sharedcontent/console/speech_synthesis_server_scenario_sample.cs) and [Java](https://github.com/Azure-Samples/cognitive-services-speech-sdk/blob/master/samples/java/jre/console/src/com/microsoft/cognitiveservices/speech/samples/console/SpeechSynthesisScenarioSamples.java).
 
 
 ## Transmit compressed audio over the network
@@ -313,10 +313,10 @@ Meanwhile, a compressed audio format helps to save the users' network bandwidth,
 
 We support many compressed formats including `opus`, `webm`, `mp3`, `silk`, and so on, see the full list in [SpeechSynthesisOutputFormat](/cpp/cognitive-services/speech/microsoft-cognitiveservices-speech-namespace#speechsynthesisoutputformat).
 For example, the bitrate of `Riff24Khz16BitMonoPcm` format is 384 kbps, while `Audio24Khz48KBitRateMonoMp3` only costs 48 kbps.
-Our Speech SDK will automatically use a compressed format for transmission when a `pcm` output format is set.
+The Speech SDK automatically uses a compressed format for transmission when a `pcm` output format is set.
 For Linux and Windows, `GStreamer` is required to enable this feature.
 Refer [this instruction](how-to-use-codec-compressed-audio-input-streams.md) to install and configure `GStreamer` for Speech SDK.
-For Android, iOS and macOS, no extra configuration is needed starting version 1.20.
+For Android, iOS, and macOS, no extra configuration is needed starting version 1.20.
 
 ## Input text streaming
 

articles/ai-services/speech-service/how-to-lower-speech-synthesis-latency.md

@@ -7,8 +7,9 @@ ms.author: eur
 manager: nitinme
 ms.service: azure-ai-speech
 ms.topic: how-to
-ms.date: 1/21/2024
+ms.date: 9/20/2024
 ms.reviewer: v-baolianzou
+#Customer intent: As a developer, I need to know how to migrate from custom voice to custom neural voice so that I can use the latest technology in my applications.
 ---
 
 # Migrate from custom voice to custom neural voice

articles/ai-services/speech-service/how-to-migrate-to-custom-neural-voice.md

@@ -6,8 +6,9 @@ author: eric-urban
 manager: nitinme
 ms.service: azure-ai-speech
 ms.topic: how-to
-ms.date: 1/21/2024
+ms.date: 9/20/2024
 ms.author: eur
+#Customer intent: As a developer, I need to know how to migrate from prebuilt standard voice to prebuilt neural voice so that I can use the latest technology in my applications.
 ---
 
 # Migrate from prebuilt standard voice to prebuilt neural voice