speechflow 1.4.4 → 1.5.0

package/CHANGELOG.md

@@ -2,6 +2,43 @@
 ChangeLog
 =========
 
+1.5.0 (2025-08-31)
+------------------
+
+- IMPROVEMENT: add improved dashboard infrastructure and allow nodes to publish dashboard info
+- IMPROVEMENT: add CLI option for exporting dashboard info via OSC
+- IMPROVEMENT: add new audio processing nodes (compressor with sidechain, expander, gain, filler)
+- IMPROVEMENT: add AWS integration nodes (Polly, Translate, Transcribe)
+- IMPROVEMENT: add OpenAI Transcribe node for speech-to-text
+- IMPROVEMENT: add noise suppression nodes (rnnoise, speex)
+- IMPROVEMENT: provide audio helper utilities and access bus functionality
+- IMPROVEMENT: improve types and error handling
+- IMPROVEMENT: switch to GPT-5 with improved error handling and timeout support
+- IMPROVEMENT: switch from native compressor to custom implementation
+- BUGFIX: fix usage of AudioIO quit and abort methods
+- BUGFIX: fix operator order in audio processing
+- BUGFIX: reset envelope array when channels change
+- BUGFIX: fix parameter configuration in audio nodes
+- BUGFIX: fix private field access and remove unnecessary casts
+- UPDATE: upgrade NPM dependencies
+- UPDATE: update OxLint rules and configuration
+- CLEANUP: cleanup and simplify code throughout project
+- CLEANUP: cleanup expander node implementation and remove stereoLink feature
+- CLEANUP: cleanup gender, ffmpeg, filler, and AWS nodes
+- CLEANUP: reduce code depth in multiple components
+- CLEANUP: align identifiers with remaining code
+- CLEANUP: make code compliant with updated linter rules
+- CLEANUP: fix indentation and remove duplicate entries
+
+1.4.5 (2025-08-07)
+------------------
+
+- IMPROVEMENT: better CLI option handling
+- IMPROVEMENT: better optical appearance of dashboard
+- BUGFIX: do not complain if no .env file is found
+- BUGFIX: avoid read-timeouts in "deepgram" node
+- CLEANUP: output stack traces only for "info" and "debug" verbosity levels
+
 1.4.4 (2025-08-07)
 ------------------
 

package/README.md

@@ -31,10 +31,20 @@ remote MQTT network I/O,
 local Voice Activity Detection (VAD),
 local voice gender recognition,
 local audio LUFS-S/RMS metering,
+local audio Speex noise suppression,
+local audio RNNoise noise suppression,
+local audio compressor processing,
+local audio expander processing,
+local audio gain processing,
+local audio filler processing,
 remote-controlable local audio muting,
+cloud-based [Amazon Transcribe](https://aws.amazon.com/transcribe/) speech-to-text conversion,
+cloud-based [OpenAI GPT Transcribe](https://platform.openai.com/docs/models/gpt-4o-mini-transcribe) speech-to-text conversion,
 cloud-based [Deepgram](https://deepgram.com) speech-to-text conversion,
 cloud-based [ElevenLabs](https://elevenlabs.io/) text-to-speech conversion,
+cloud-based [Amazon Polly](https://aws.amazon.com/polly/) text-to-speech conversion,
 cloud-based [DeepL](https://deepl.com) text-to-text translation,
+cloud-based [Amazon Translate](https://aws.amazon.com/translate/) text-to-text translation,
 cloud-based [OpenAI/GPT](https://openai.com) text-to-text translation (or spelling correction),
 local [Ollama/Gemma](https://ollama.com) text-to-text translation (or spelling correction),
 local [OPUS/ONNX](https://github.com/Helsinki-NLP/Opus-MT) text-to-text translation,
@@ -53,6 +63,59 @@ derived from the exported `SpeechFlowNode` class of the `speechflow` package.
 **SpeechFlow** is written in TypeScript and
 ships as an installable package for the Node Package Manager (NPM).
 
+Impression
+----------
+
+**SpeechFlow** is a command-line interface (CLI) based tool, so there
+is no exciting screenshot possible from its CLI appearance, of course.
+Instead, here is a sample of a fictive training which is held in German
+and real-time translated to English.
+
+First, the used configuration was a straight linear pipeline in file `sample.conf`:
+
+```txt
+device(device: "coreaudio:Elgato Wave:3", mode: "r") |
+meter(interval: 50, dashboard: "meter1") |
+deepgram(language: "de", model: "nova-2", interim: true) |
+trace(type: "text", dashboard: "text1") |
+filter(name: "final", type: "text", var: "kind", op: "==", val: "final") |
+sentence() |
+trace(type: "text", dashboard: "text2") |
+deepl(src: "de", dst: "en") |
+trace(type: "text", dashboard: "text3") |
+elevenlabs(voice: "Mark", optimize: "latency", speed: 1.05, language: "en") |
+meter(interval: 50, dashboard: "meter2") |
+device(device: "coreaudio:USBAudio2.0", mode: "w")
+```
+
+Second, the corresponding **SpeechFlow** command was:
+
+```sh
+$ speechflow -v info -c sample.conf \
+  -d audio:meter1:DE,text:text1:DE-Interim,text:text2:DE-Final,text:text3:EN,audio:meter2:EN
+```
+
+Finally, the resulting dashboard under URL `http://127.0.0.1:8484/` was:
+
+![dashboard](etc/speechflow.png)
+
+On the left you can see the volume meter of the microphone (`device`),
+followed by the German result of the speech-to-text conversion
+(`deepgram`), followed by the still German results of the text-to-text
+sentence splitting/aggregation (`sentence`), followed by the English
+results of the text-to-text translation (`deepl`) and then finally on
+the right you can see the volume meter of the text-to-speech conversion
+(`elevenlabs`).
+
+The entire **SpeechFlow** processing pipeline runs in real-time and
+the latency between input and output audio is about 2-3 seconds, very
+similar to the usual latency human live translators also cause. The
+latency primarily comes from the speech-to-text part in the pipeline,
+as the end of sentences have to be awaited -- especially in the German
+language where the verb can come very late in a sentence. So, the
+latency is primarily not caused by any technical aspects, but by the
+nature of live translation.
+
 Installation
 ------------
 
@@ -235,18 +298,29 @@ First a short overview of the available processing nodes:
   **mute**,
   **meter**,
   **vad**,
-  **gender**.
+  **gender**,
+  **speex**,
+  **rrnoise**,
+  **compressor**,
+  **expander**,
+  **gain**,
+  **filler**.
 - Audio-to-Text nodes:
+  **openaitranscribe**,
+  **awstranscribe**,
   **deepgram**.
 - Text-to-Text nodes:
   **deepl**,
+  **awstranslate**,
   **openai**,
   **ollama**,
   **transformers**,
   **subtitle**,
   **format**.
 - Text-to-Audio nodes:
+  **awspolly**.
   **elevenlabs**.
+  **kokoro**.
 - Any-to-Any nodes:
   **filter**,
   **trace**.
@@ -450,10 +524,160 @@ The following nodes process audio chunks only.
   | ----------- | --------- | -------- | ------------------------ |
   | **window**  | 0         | 500      | *none*                   |
 
+- Node: **speex**<br/>
+  Purpose: **Speex Noise Suppression node**<br/>
+  Example: `speex(attentuate: -18)`
+
+  > This node uses the Speex DSP pre-processor to perform noise
+  > suppression, i.e., it detects and attenuates (by a certain level of
+  > dB) the noise in the audio stream.
+
+  | Port    | Payload     |
+  | ------- | ----------- |
+  | input   | audio       |
+  | output  | audio       |
+
+  | Parameter   | Position  | Default  | Requirement              |
+  | ----------- | --------- | -------- | ------------------------ |
+  | **attentuate** | 0 | -18  | *none* | `-60 <= n <= 0` |
+
+- Node: **rnnoise**<br/>
+  Purpose: **RNNoise Noise Suppression node**<br/>
+  Example: `rnnoise()`
+
+  > This node uses RNNoise to perform noise suppression, i.e., it
+  > detects and attenuates the noise in the audio stream.
+
+  | Port    | Payload     |
+  | ------- | ----------- |
+  | input   | audio       |
+  | output  | audio       |
+
+  | Parameter   | Position  | Default  | Requirement              |
+  | ----------- | --------- | -------- | ------------------------ |
+
+- Node: **compressor**<br/>
+  Purpose: **audio compressor node**<br/>
+  Example: `compressor(thresholdDb: -18)`
+
+  > This node applies a dynamics compressor, i.e., it attenuates the
+  > volume by a certain ratio whenever the volume is above the threshold.
+
+  | Port    | Payload     |
+  | ------- | ----------- |
+  | input   | audio       |
+  | output  | audio       |
+
+  | Parameter   | Position  | Default  | Requirement              |
+  | ----------- | --------- | -------- | ------------------------ |
+  | **thresholdDb** | *none* | -18 | `n <= 0 && n >= -60` |
+  | **ratio**       | *none* | 4   | `n >= 1 && n <= 20`  |
+  | **attackMs**    | *none* | 10  | `n >= 0 && n <= 100` |
+  | **releaseMs**   | *none* | 50  | `n >= 0 && n <= 100` |
+  | **kneeDb**      | *none* | 6   | `n >= 0 && n <= 100` |
+  | **makeupDb**    | *none* | 0   | `n >= 0 && n <= 100` |
+
+- Node: **expander**<br/>
+  Purpose: **audio expander node**<br/>
+  Example: `expander(thresholdDb: -46)`
+
+  > This node applies a dynamics expander, i.e., it attenuates the
+  > volume by a certain ratio whenever the volume is below the threshold.
+
+  | Port    | Payload     |
+  | ------- | ----------- |
+  | input   | audio       |
+  | output  | audio       |
+
+  | Parameter   | Position  | Default  | Requirement              |
+  | ----------- | --------- | -------- | ------------------------ |
+  | **thresholdDb** | *none* | -45 | `n <= 0 && n >= -60` |
+  | **ratio**       | *none* | 4   | `n >= 1 && n <= 20`  |
+  | **attackMs**    | *none* | 10  | `n >= 0 && n <= 100` |
+  | **releaseMs**   | *none* | 50  | `n >= 0 && n <= 100` |
+  | **kneeDb**      | *none* | 6   | `n >= 0 && n <= 100` |
+  | **makeupDb**    | *none* | 0   | `n >= 0 && n <= 100` |
+
+- Node: **gain**<br/>
+  Purpose: **audio gain adjustment node**<br/>
+  Example: `gain(db: 12)`
+
+  > This node applies a gain adjustment to audio, i.e., it increases or
+  > decreases the volume by certain decibels
+
+  | Port    | Payload     |
+  | ------- | ----------- |
+  | input   | audio       |
+  | output  | audio       |
+
+  | Parameter   | Position  | Default  | Requirement              |
+  | ----------- | --------- | -------- | ------------------------ |
+  | **db** | *none* | 12 | `n >= -60 && n <= -60` |
+
+- Node: **filler**<br/>
+  Purpose: **audio filler node**<br/>
+  Example: `filler()`
+
+  > This node adds missing audio frames of silence in order to fill
+  > the chronological gaps between generated audio frames (from
+  > text-to-speech).
+
+  | Port    | Payload     |
+  | ------- | ----------- |
+  | input   | audio       |
+  | output  | audio       |
+
+  | Parameter   | Position  | Default  | Requirement              |
+  | ----------- | --------- | -------- | ------------------------ |
+
 ### Audio-to-Text Nodes
 
 The following nodes convert audio to text chunks.
 
+- Node: **openaitranscribe**<br/>
+  Purpose: **OpenAI/GPT Speech-to-Text conversion**<br/>
+  Example: `openaitranscribe(language: "de")`<br/>
+  Notice: this node requires an OpenAI API key!
+
+  > This node uses OpenAI GPT to perform Speech-to-Text (S2T)
+  > conversion, i.e., it recognizes speech in the input audio stream and
+  > outputs a corresponding text stream.
+
+  | Port    | Payload     |
+  | ------- | ----------- |
+  | input   | text        |
+  | output  | text        |
+
+  | Parameter    | Position  | Default  | Requirement        |
+  | ------------ | --------- | -------- | ------------------ |
+  | **key**      | *none*    | env.SPEECHFLOW\_OPENAI\_KEY | *none* |
+  | **api**      | *none*    | "https://api.openai.com" | `/^https?:\/\/.+?:\d+$/` |
+  | **model**    | *none*    | "gpt-4o-mini-transcribe" | *none* |
+  | **language** | *none*    | "en"     | `/^(?:de\|en)$/` |
+  | **interim**  | *none*    | false    | *none* |
+
+- Node: **awstranscribe**<br/>
+  Purpose: **Amazon Transcribe Speech-to-Text conversion**<br/>
+  Example: `awstranscribe(language: "de")`<br/>
+  Notice: this node requires an API key!
+
+  > This node uses Amazon Trancribe to perform Speech-to-Text (S2T)
+  > conversion, i.e., it recognizes speech in the input audio stream and
+  > outputs a corresponding text stream.
+
+  | Port    | Payload     |
+  | ------- | ----------- |
+  | input   | audio       |
+  | output  | text        |
+
+  | Parameter    | Position  | Default  | Requirement        |
+  | ------------ | --------- | -------- | ------------------ |
+  | **key**      | *none*    | env.SPEECHFLOW\_AMAZON\_KEY | *none* |
+  | **secKey**   | *none*    | env.SPEECHFLOW\_AMAZON\_KEY\_SEC | *none* |
+  | **region**   | *none*    | "eu-central-1" | *none* |
+  | **language** | *none*    | "en" | `/^(?:en|de)$/` |
+  | **interim**  | *none*    | false | *none* |
+
 - Node: **deepgram**<br/>
   Purpose: **Deepgram Speech-to-Text conversion**<br/>
   Example: `deepgram(language: "de")`<br/>
@@ -498,6 +722,26 @@ The following nodes process text chunks only.
   | **src**      | 0         | "de"     | `/^(?:de\|en)$/` |
   | **dst**      | 1         | "en"     | `/^(?:de\|en)$/` |
 
+- Node: **awstranslate**<br/>
+  Purpose: **AWS Translate Text-to-Text translation**<br/>
+  Example: `awstranslate(src: "de", dst: "en")`<br/>
+  Notice: this node requires an API key!
+
+  > This node performs translation between English and German languages.
+
+  | Port    | Payload     |
+  | ------- | ----------- |
+  | input   | text        |
+  | output  | text        |
+
+  | Parameter    | Position  | Default  | Requirement        |
+  | ------------ | --------- | -------- | ------------------ |
+  | **key**      | *none*    | env.SPEECHFLOW\_AMAZON\_KEY | *none* |
+  | **secKey**   | *none*    | env.SPEECHFLOW\_AMAZON\_KEY\_SEC | *none* |
+  | **region**   | *none*    | "eu-central-1" | *none* |
+  | **src**      | 0         | "de"     | `/^(?:de\|en)$/` |
+  | **dst**      | 1         | "en"     | `/^(?:de\|en)$/` |
+
 - Node: **openai**<br/>
   Purpose: **OpenAI/GPT Text-to-Text translation and spelling correction**<br/>
   Example: `openai(src: "de", dst: "en")`<br/>
@@ -618,14 +862,36 @@ The following nodes process text chunks only.
 
 The following nodes convert text chunks to audio chunks.
 
+- Node: **awspolly**<br/>
+  Purpose: **Amazon Polly Text-to-Speech conversion**<br/>
+  Example: `awspolly(language: "en", voice: "Danielle)`<br/>
+  Notice: this node requires an Amazon API key!
+
+  > This node uses Amazon Polly to perform Text-to-Speech (T2S)
+  > conversion, i.e., it converts the input text stream into an output
+  > audio stream. It is intended to generate speech.
+
+  | Port    | Payload     |
+  | ------- | ----------- |
+  | input   | text        |
+  | output  | audio       |
+
+  | Parameter      | Position  | Default   | Requirement        |
+  | -------------- | --------- | --------- | ------------------ |
+  | **key**        | *none*    | env.SPEECHFLOW\_AMAZON\_KEY | *none* |
+  | **secKey**     | *none*    | env.SPEECHFLOW\_AMAZON\_KEY\_SEC | *none* |
+  | **region**     | *none*    | "eu-central-1" | *none* |
+  | **voice**      | 0         | "Amy"     | `^(?:Amy|Danielle|Joanna|Matthew|Ruth|Stephen|Viki|Daniel)$/` |
+  | **language**   | 1         | "en"      | `/^(?:de\|en)$/`  |
+
 - Node: **elevenlabs**<br/>
   Purpose: **ElevenLabs Text-to-Speech conversion**<br/>
   Example: `elevenlabs(language: "en")`<br/>
   Notice: this node requires an ElevenLabs API key!
 
-  > This node perform Text-to-Speech (T2S) conversion, i.e., it converts
-  > the input text stream into an output audio stream. It is intended to
-  > generate speech.
+  > This node uses ElevenLabs to perform Text-to-Speech (T2S)
+  > conversion, i.e., it converts the input text stream into an output
+  > audio stream. It is intended to generate speech.
 
   | Port    | Payload     |
   | ------- | ----------- |
@@ -647,9 +913,9 @@ The following nodes convert text chunks to audio chunks.
   Example: `kokoro(language: "en")`<br/>
   Notice: this currently support English language only!
 
-  > This node perform Text-to-Speech (T2S) conversion, i.e., it converts
-  > the input text stream into an output audio stream. It is intended to
-  > generate speech.
+  > This node uses Kokoro to perform Text-to-Speech (T2S) conversion,
+  > i.e., it converts the input text stream into an output audio stream.
+  > It is intended to generate speech.
 
   | Port    | Payload     |
   | ------- | ----------- |

package/etc/claude.md

@@ -0,0 +1,70 @@
+
+# CLAUDE.md
+
+This file provides guidance to Claude Code (claude.ai/code) when working
+with code in this repository.
+
+## Project Overview
+
+SpeechFlow is a command-line interface tool for establishing directed
+data flow graphs of audio and text processing nodes. It enables flexible
+speech processing tasks including capturing audio, text-to-speech,
+speech-to-text, and speech-to-speech translation.
+
+## Architecture
+
+SpeechFlow uses a modular node-based architecture:
+
+- **Core Engine**: TypeScript-based CLI tool that orchestrates processing flows
+- **Processing Nodes**: Modular components for different speech processing tasks (see `src/speechflow-node-*.ts`)
+- **Flow Expression Language**: Based on FlowLink for defining processing graphs
+- **Web Interfaces**: Two Vue.js applications for dashboard and subtitle display
+- **REST/WebSocket API**: External control interface for nodes
+
+### Key Components
+
+- **Main CLI**:
+  `src/speechflow.ts` - Entry point and CLI parsing
+- **Nodes**:
+  - Input/Output: `file`, `device`, `websocket`, `mqtt`  
+  - Audio-to-Audio: `ffmpeg`, `wav`, `mute`, `meter`, `vad`, `gender`
+  - Audio-to-Text: `deepgram`
+  - Text-to-Text: `deepl`, `openai`, `ollama`, `transformers`, `subtitle`, `format`, `sentence`
+  - Text-to-Audio: `elevenlabs`, `kokoro`
+  - Any-to-Any: `filter`, `trace`
+
+## Development Commands
+
+The project uses STX (Simple Task eXecutor) for build automation. Main commands:
+
+### Core Project
+
+```bash
+npm start lint          # Static code analysis (TypeScript, ESLint, Biome, Oxlint)
+npm start build         # Compile TypeScript to JavaScript in dst/
+npm start dev           # Multi-pane development dashboard with linting, building, and server
+npm start server        # Run the main speechflow program
+npm start clean         # Remove generated files
+```
+
+## Project Structure
+
+- `src/` - Main TypeScript source files
+- `dst/` - Compiled JavaScript output
+- `etc/` - Configuration files (TypeScript, ESLint, Biome, etc.)
+- `package.d/` - NPM package patches
+
+## Development Notes
+
+- Node.js 22+ required
+- Uses object-mode streaming with timestamps for audio/text processing
+- External services integration: Deepgram, ElevenLabs, DeepL, OpenAI, Ollama
+- Supports local processing: FFmpeg, WAV, Voice Activity Detection, Gender Detection
+- REST/WebSocket API on port 8484 for external control
+
+## Configuration
+
+Main configuration in `etc/speechflow.yaml` with example
+processing graphs. Environment variables used for API keys (e.g.,
+`SPEECHFLOW_DEEPGRAM_KEY`, `SPEECHFLOW_ELEVENLABS_KEY`).
+

package/etc/speechflow.yaml

@@ -68,8 +68,10 @@ studio-transcription: |
                 subtitle(format: "vtt") |
                     file(path: argv.2, mode: "w", type: "text"),
                 subtitle(format: "srt") |
-                    file(path: argv.3, mode: "w", type: "text")
-                elevenlabs(voice: "Mark", optimize: "quality", speed: 1.05, language: "en")
+                    file(path: argv.3, mode: "w", type: "text"),
+                elevenlabs(voice: "Mark", optimize: "quality", speed: 1.05, language: "en") |
+                    wav(mode: "encode") |
+                        file(path: argv.4, mode: "w", type: "audio")
             }
         }
     }
@@ -79,31 +81,30 @@ studio-transcription: |
 studio-translation: |
     device(device: "coreaudio:Elgato Wave:3", mode: "r") | {
         gender() | {
-            meter(interval: 250) |
+            meter(interval: 250, dashboard: "meter1") |
                 wav(mode: "encode") |
                     file(path: "program-de.wav", mode: "w", type: "audio"),
-            deepgram(language: "de", key: env.SPEECHFLOW_DEEPGRAM_KEY) | {
+            deepgram(language: "de", key: env.SPEECHFLOW_DEEPGRAM_KEY, interim: true) | {
+                trace(name: "trace1", type: "text", dashboard: "text1")
                 subtitle(format: "vtt", words: true) |
                     file(path: "program-de.vtt", mode: "w", type: "text"),
                 sentence() | {
-                    format(width: 80) |
-                        file(path: "program-de.txt", mode: "w", type: "text"),
+                    trace(name: "trace2", type: "text", notify: true, dashboard: "text2") |
+                        format(width: 80) |
+                            file(path: "program-de.txt", mode: "w", type: "text"),
                     deepl(src: "de", dst: "en", key: env.SPEECHFLOW_DEEPL_KEY) | {
-                        trace(name: "text", type: "text") | {
+                        trace(name: "trace3", type: "text", dashboard: "text3") | {
                             format(width: 80) |
                                 file(path: "program-en.txt", mode: "w", type: "text"),
                             subtitle(format: "vtt", words: false) |
                                 file(path: "program-en.vtt", mode: "w", type: "text"),
-                            mqtt(url: "mqtt://10.1.0.10:1883",
-                                username: env.SPEECHFLOW_MQTT_USER,
-                                password: env.SPEECHFLOW_MQTT_PASS,
-                                topicWrite: "stream/studio/sender"),
                             {
                                 filter(name: "S2T-male", type: "text", var: "meta:gender", op: "==", val: "male") |
                                     elevenlabs(voice: "Mark", optimize: "latency", speed: 1.05, language: "en"),
                                 filter(name: "S2T-female", type: "text", var: "meta:gender", op: "==", val: "female") |
                                     elevenlabs(voice: "Brittney", optimize: "latency", speed: 1.05, language: "en")
                             } | {
+                                meter(interval: 250, dashboard: "meter2"),
                                 wav(mode: "encode") |
                                     file(path: "program-en.wav", mode: "w", type: "audio"),
                                 device(device: "coreaudio:USBAudio2.0", mode: "w")
@@ -115,3 +116,20 @@ studio-translation: |
         }
     }
 
+#   Test-drive for development
+test: |
+    device(device: "coreaudio:Elgato Wave:3", mode: "r") |
+        meter(interval: 50, dashboard: "meter1") |
+            deepgram(language: "de", model: "nova-2", key: env.SPEECHFLOW_DEEPGRAM_KEY, interim: true) |
+                trace(type: "text", dashboard: "text1") | {
+                    subtitle(mode: "render", addr: "127.0.0.1", port: 8585),
+                    filter(name: "final", type: "text", var: "kind", op: "==", val: "final") |
+                        sentence() |
+                            trace(type: "text", dashboard: "text2") |
+                                deepl(src: "de", dst: "en", key: env.SPEECHFLOW_DEEPL_KEY) |
+                                    trace(type: "text", dashboard: "text3") |
+                                        elevenlabs(voice: "Mark", optimize: "latency", speed: 1.05, language: "en") |
+                                            meter(interval: 50, dashboard: "meter2") |
+                                                device(device: "coreaudio:USBAudio2.0", mode: "w")
+                }
+

package/etc/stx.conf

@@ -17,6 +17,13 @@ upd
     (cd speechflow-ui-db && npx -y upd) && \
     (cd speechflow-ui-st && npx -y upd)
 
+#   [top-level] provide statistics about code base
+cloc
+    cloc etc \
+        speechflow-cli/etc   speechflow-cli/src   \
+        speechflow-ui-db/etc speechflow-ui-db/src \
+        speechflow-ui-st/etc speechflow-ui-st/src
+
 #   [top-level] lint components for development
 lint
     npm --prefix speechflow-cli   start lint && \

package/package.json

@@ -1,10 +1,11 @@
 {
     "name":             "speechflow",
-    "version":          "1.4.4",
-    "x-stdver":         "1.4.4-GA",
-    "x-release":        "2025-08-07",
+    "version":          "1.5.0",
+    "x-stdver":         "1.5.0-GA",
+    "x-release":        "2025-08-31",
     "homepage":         "https://github.com/rse/speechflow",
     "description":      "Speech Processing Flow Graph",
+    "keywords":         [ "speech", "audio", "flow", "graph" ],
     "license":          "GPL-3.0-only",
     "author": {
         "name":         "Dr. Ralf S. Engelschall",
@@ -16,17 +17,17 @@
         "url":          "git+https://github.com/rse/speechflow.git"
     },
     "dependencies": {
-        "@rse/stx":     "1.0.7"
+        "@rse/stx":     "1.0.9"
     },
     "devDependencies": {
         "nodemon":      "3.1.10",
         "watch":        "1.0.2",
-        "concurrently": "9.2.0",
+        "concurrently": "9.2.1",
         "wait-on":      "8.0.4",
         "cross-env":    "10.0.0",
         "shx":          "0.4.0"
     },
-    "engines" : {
+    "engines": {
         "npm":          ">=10.0.0",
         "node":         ">=22.0.0"
     },

package/speechflow-cli/dst/speechflow-node-a2a-compressor-wt.d.ts

@@ -0,0 +1 @@
+export {};