Speech Processing Flow Graph
SpeechFlow is a command-line interface based tool for macOS, Windows and Linux, establishing a directed data flow graph of audio and text processing nodes. This way, it allows to perform various speech processing tasks in a very flexible and configurable way. The usual supported tasks are capturing audio, generate narrations of text (aka text-to-speech), generate transcriptions or subtitles for audio (aka speech-to-text), and generate translations for audio (aka speech-to-speech).
SpeechFlow comes with built-in graph nodes for various functionalities:
- file and audio device I/O for local connectivity,
- WebSocket and MQTT network I/O for remote connectivity,
- local Voice Activity Detection (VAD),
- local voice gender recognition,
- local audio LUFS-S/RMS metering,
- local audio Speex and RNNoise noise suppression,
- local audio compressor and expander dynamics processing,
- local audio gain adjustment,
- local audio pitch shifting and time stretching,
- local audio gap filler processing,
- remote-controlable audio muting,
- cloud-based speech-to-text conversion with Amazon Transcribe, OpenAI GPT-Transcribe, or Deepgram.
- cloud-based text-to-text translation (or spelling correction) with DeepL, Amazon Translate, Google Cloud Translate, or OpenAI GPT.
- local text-to-text translation (or spelling correction) with Ollama/Gemma or Transformers/OPUS.
- cloud-based text-to-speech conversion with ElevenLabs or Amazon Polly.
- local text-to-speech conversion with Kokoro.
- local FFmpeg-based speech-to-speech conversion,
- local WAV speech-to-speech decoding/encoding,
- local text-to-text formatting, regex-based modification, sentencing merging/splitting, subtitle generation, and formatting.
- local text or audio chunk filtering and tracing.
Additional, SpeechFlow graph nodes can be provided externally
by NPM packages named speechflow-node-xxx
which expose a class
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).
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
:
xio-device(device: env.SPEECHFLOW_DEVICE_MIC, mode: "r") |
a2a-meter(interval: 50, dashboard: "meter1") |
a2t-deepgram(language: "de", model: "nova-2", interim: true) |
x2x-trace(type: "text", dashboard: "text1") |
x2x-filter(name: "final", type: "text", var: "kind", op: "==", val: "final") |
t2t-sentence() |
x2x-trace(type: "text", dashboard: "text2") |
t2t-deepl(src: "de", dst: "en") |
x2x-trace(type: "text", dashboard: "text3") |
t2a-elevenlabs(voice: "Mark", optimize: "latency", speed: 1.05, language: "en") |
a2a-meter(interval: 50, dashboard: "meter2") |
xio-device(device: env.SPEECHFLOW_DEVICE_SPK, mode: "w")
Second, the corresponding SpeechFlow command was:
$ 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:
On the left you can see the volume meter of the microphone (xio-device
),
followed by the German result of the speech-to-text conversion
(a2t-deepgram
), followed by the still German results of the text-to-text
sentence splitting/aggregation (t2t-sentence
), followed by the English
results of the text-to-text translation (t2t-deepl
) and then finally on
the right you can see the volume meter of the text-to-speech conversion
(t2a-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.
$ npm install -g speechflow
$ speechflow
[-h|--help]
[-V|--version]
[-S|--status]
[-v|--verbose <level>]
[-a|--address <ip-address>]
[-p|--port <tcp-port>]
[-C|--cache <directory>]
[-e|--expression <expression>]
[-f|--file <file>]
[-c|--config <id>@<yaml-config-file>]
[<argument> [...]]
The SpeechFlow graph expression language is based on FlowLink, which itself has a language following the following BNF-style grammar:
# (sub-)graph expression: set or sequence of nodes, single node, or group
expr ::= parallel
| sequential
| node
| group
# set of nodes, connected in parallel
parallel ::= sequential ("," sequential)+
# sequence of nodes, connected in chain
sequential ::= node ("|" node)+
# single node with optional parameter(s) and optional links
node ::= id ("(" (param ("," param)*)? ")")? links?
# single parameter: array, object, variable reference, template string,
# or string/number literal, or special value literal
param ::= array | object | variable | template | string | number | value
# set of links
links ::= link (_ link)*
link ::= "<" | "<<" | ">" | ">>" id
# group with sub-graph
group ::= "{" expr "}"
# identifier and variable
id ::= /[a-zA-Z_][a-zA-Z0-9_-]*/
variable ::= id
# array of values
array ::= "[" (param ("," param)*)? "]"
# object of key/valus
object ::= "{" (id ":" param ("," id ":" param)*)? "}"
# template string
template ::= "`" ("${" variable "}" / ("\\`"|.))* "`"
# string literal
string ::= /"(\\"|.)*"/
| /'(\\'|.)*'/
# number literal
number ::= /[+-]?/ number-value
number-value ::= "0b" /[01]+/
| "0o" /[0-7]+/
| "0x" /[0-9a-fA-F]+/
| /[0-9]*\.[0-9]+([eE][+-]?[0-9]+)?/
| /[0-9]+/
# special value literal
value ::= "true" | "false" | "null" | "NaN" | "undefined"
SpeechFlow makes available to FlowLink all SpeechFlow nodes as
node
, the CLI arguments under the array variable
named argv
, and all
environment variables under the object variable
named env
.
The following are examples of particular SpeechFlow processing graphs. They can also be found in the sample speechflow.yaml file.
-
Capturing: Capture audio from microphone device into WAV audio file:
xio-device(device: env.SPEECHFLOW_DEVICE_MIC, mode: "r") | a2a-wav(mode: "encode") | xio-file(path: "capture.wav", mode: "w", type: "audio")
-
Pass-Through: Pass-through audio from microphone device to speaker device and in parallel record it to WAV audio file:
xio-device(device: env.SPEECHFLOW_DEVICE_MIC, mode: "r") | { a2a-wav(mode: "encode") | xio-file(path: "capture.wav", mode: "w", type: "audio"), xio-device(device: env.SPEECHFLOW_DEVICE_SPK, mode: "w") }
-
Transcription: Generate text file with German transcription of MP3 audio file:
xio-file(path: argv.0, mode: "r", type: "audio") | a2a-ffmpeg(src: "mp3", dst: "pcm") | a2t-deepgram(language: "de", key: env.SPEECHFLOW_DEEPGRAM_KEY) | t2t-format(width: 80) | xio-file(path: argv.1, mode: "w", type: "text")
-
Subtitling: Generate text file with German subtitles of MP3 audio file:
xio-file(path: argv.0, mode: "r", type: "audio") | a2a-ffmpeg(src: "mp3", dst: "pcm") | a2t-deepgram(language: "de", key: env.SPEECHFLOW_DEEPGRAM_KEY) | t2t-subtitle(format: "vtt") | xio-file(path: argv.1, mode: "w", type: "text")
-
Speaking: Generate audio file with English voice for a text file:
xio-file(path: argv.0, mode: "r", type: "text") | t2a-kokoro(language: "en") | a2a-wav(mode: "encode") | xio-file(path: argv.1, mode: "w", type: "audio")
-
Ad-Hoc Translation: Ad-Hoc text translation from German to English via stdin/stdout:
xio-file(path: "-", mode: "r", type: "text") | t2t-deepl(src: "de", dst: "en") | xio-file(path: "-", mode: "w", type: "text")
-
Studio Translation: Real-time studio translation from German to English, including the capturing of all involved inputs and outputs:
xio-device(device: env.SPEECHFLOW_DEVICE_MIC, mode: "r") | { a2a-gender() | { a2a-meter(interval: 250) | a2a-wav(mode: "encode") | xio-file(path: "program-de.wav", mode: "w", type: "audio"), a2t-deepgram(language: "de", key: env.SPEECHFLOW_DEEPGRAM_KEY) | { t2t-sentence() | { t2t-format(width: 80) | xio-file(path: "program-de.txt", mode: "w", type: "text"), t2t-deepl(src: "de", dst: "en", key: env.SPEECHFLOW_DEEPL_KEY) | { x2x-trace(name: "text", type: "text") | { t2t-format(width: 80) | xio-file(path: "program-en.txt", mode: "w", type: "text"), t2t-subtitle(format: "srt") | xio-file(path: "program-en.srt", mode: "w", type: "text"), xio-mqtt(url: "mqtt://10.1.0.10:1883", username: env.SPEECHFLOW_MQTT_USER, password: env.SPEECHFLOW_MQTT_PASS, topicWrite: "stream/studio/sender"), { x2x-filter(name: "S2T-male", type: "text", var: "meta:gender", op: "==", val: "male") | t2a-elevenlabs(voice: "Mark", optimize: "latency", speed: 1.05, language: "en"), x2x-filter(name: "S2T-female", type: "text", var: "meta:gender", op: "==", val: "female") | t2a-elevenlabs(voice: "Brittney", optimize: "latency", speed: 1.05, language: "en") } | { a2a-wav(mode: "encode") | xio-file(path: "program-en.wav", mode: "w", type: "audio"), xio-device(device: env.SPEECHFLOW_DEVICE_SPK, mode: "w") } } } } } } }
First a short overview of the available processing nodes:
- Input/Output nodes: xio-file, xio-device, xio-websocket, xio-mqtt.
- Audio-to-Audio nodes: a2a-ffmpeg, a2a-wav, a2a-mute, a2a-meter, a2a-vad, a2a-gender, a2a-speex, a2a-rnnoise, a2a-compressor, a2a-expander, a2a-gain, a2a-pitch, a2a-filler.
- Audio-to-Text nodes: a2t-openai, a2t-amazon, a2t-deepgram.
- Text-to-Text nodes: t2t-deepl, t2t-amazon, t2t-openai, t2t-ollama, t2t-transformers, t2t-google, t2t-modify, t2t-subtitle, t2t-format, t2t-sentence.
- Text-to-Audio nodes: t2a-amazon, t2a-elevenlabs, t2a-kokoro.
- Any-to-Any nodes: x2x-filter, x2x-trace.
The following nodes are for external I/O, i.e, to read/write from external files, devices and network services.
-
Node: xio-file
Purpose: File and StdIO source/sink
Example:xio-file(path: "capture.pcm", mode: "w", type: "audio")
This node allows the reading/writing from/to files or from StdIO. It is intended to be used as source and sink nodes in batch processing, and as sing nodes in real-time processing.
Port Payload input text, audio output text, audio Parameter Position Default Requirement path 0 none none mode 1 "r" /^(?:r|w|rw)$/
type 2 "audio" /^(?:audio|text)$/
chunka 200 10 <= n <= 1000
chunkt 65536 1024 <= n <= 131072
-
Node: xio-device
Purpose: Microphone/speaker device source/sink
Example:xio-device(device: env.SPEECHFLOW_DEVICE_MIC, mode: "r")
This node allows the reading/writing from/to audio devices. It is intended to be used as source nodes for microphone devices and as sink nodes for speaker devices.
Port Payload input audio output audio Parameter Position Default Requirement device 0 none /^(.+?):(.+)$/
mode 1 "rw" /^(?:r|w|rw)$/
chunk 2 200 10 <= n <= 1000
-
Node: xio-websocket
Purpose: WebSocket source/sink
Example:xio-websocket(connect: "ws://127.0.0.1:12345", type: "text")
Notice: this node requires a peer WebSocket service!This node allows reading/writing from/to WebSocket network services. It is primarily intended to be used for sending out the text of subtitles, but can be also used for receiving the text to be processed.
Port Payload input text, audio output text, audio Parameter Position Default Requirement listen none none /^(?:|ws:\/\/(.+?):(\d+))$/
connect none none /^(?:|ws:\/\/(.+?):(\d+)(?:\/.*)?)$/
type none "audio" /^(?:audio|text)$/
-
Node: xio-mqtt
Purpose: MQTT sink
Example:xio-mqtt(url: "mqtt://127.0.0.1:1883", username: "foo", password: "bar", topic: "quux")
Notice: this node requires a peer MQTT broker!This node allows reading/writing from/to MQTT broker topics. It is primarily intended to be used for sending out the text of subtitles, but can be also used for receiving the text to be processed.
Port Payload input text output none Parameter Position Default Requirement url 0 none /^(?:|(?:ws|mqtt):\/\/(.+?):(\d+))$/
username 1 none /^.+$/
password 2 none /^.+$/
topic 3 none /^.+$/
The following nodes process audio chunks only.
-
Node: a2a-ffmpeg
Purpose: FFmpeg audio format conversion
Example:a2a-ffmpeg(src: "pcm", dst: "mp3")
This node allows converting between audio formats. It is primarily intended to support the reading/writing of external MP3 and Opus format files, although SpeechFlow internally uses PCM format only.
Port Payload input audio output audio Parameter Position Default Requirement src 0 "pcm" /^(?:pcm|wav|mp3|opus)$/
dst 1 "wav" /^(?:pcm|wav|mp3|opus)$/
-
Node: a2a-wav
Purpose: WAV audio format conversion
Example:a2a-wav(mode: "encode")
This node allows converting between PCM and WAV audio formats. It is primarily intended to support the reading/writing of external WAV format files, although SpeechFlow internally uses PCM format only.
Port Payload input audio output audio Parameter Position Default Requirement mode 0 "encode" /^(?:encode|decode)$/
-
Node: a2a-mute
Purpose: volume muting node
Example:a2a-mute()
Notice: this node has to be externally controlled via REST/WebSockets!This node allows muting the audio stream by either silencing or even unplugging. It has to be externally controlled via REST/WebSocket (see below).
Port Payload input audio output audio Parameter Position Default Requirement -
Node: a2a-meter
Purpose: Loudness metering node
Example:a2a-meter(250)
This node allows measuring the loudness of the audio stream. The results are emitted to both the logfile of SpeechFlow and the WebSockets API (see below). It can optionally send the meter information to the dashboard.
Port Payload input audio output audio Parameter Position Default Requirement interval 0 250 none mode 1 "filter" /^(?:filter|sink)$/
dashboard none none -
Node: a2a-vad
Purpose: Voice Audio Detection (VAD) node
Example:a2a-vad()
This node perform Voice Audio Detection (VAD), i.e., it detects voice in the audio stream and if not detected either silences or unplugs the audio stream.
Port Payload input audio output audio Parameter Position Default Requirement mode none "unplugged" /^(?:silenced|unplugged)$/
posSpeechThreshold none 0.50 none negSpeechThreshold none 0.35 none minSpeechFrames none 2 none redemptionFrames none 12 none preSpeechPadFrames none 1 none postSpeechTail none 1500 none -
Node: a2a-gender
Purpose: Gender Detection node
Example:a2a-gender()
This node performs gender detection on the audio stream. It annotates the audio chunks with
gender=male
orgender=female
meta information. Use this meta information with the "filter" node.Port Payload input audio output audio Parameter Position Default Requirement window 0 500 none treshold 1 0.50 none hysteresis 2 0.25 none -
Node: a2a-speex
Purpose: Speex Noise Suppression node
Example:a2a-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 -
Node: a2a-rnnoise
Purpose: RNNoise Noise Suppression node
Example:a2a-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: a2a-compressor
Purpose: audio compressor node
Example:a2a-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: a2a-expander
Purpose: audio expander node
Example:a2a-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: a2a-gain
Purpose: audio gain adjustment node
Example:a2a-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: a2a-pitch
Purpose: audio pitch shifting and time stretching
Example:a2a-pitch(pitch: 1.2, semitones: 3)
This node performs real-time pitch shifting and time stretching on audio streams using the SoundTouch algorithm. It can adjust pitch without changing tempo, change tempo without affecting pitch, or modify both independently.
Port Payload input audio output audio Parameter Position Default Requirement rate none 1.0 0.25 <= n <= 4.0
tempo none 1.0 0.25 <= n <= 4.0
pitch none 1.0 0.25 <= n <= 4.0
semitones none 0.0 -24 <= n <= 24
-
Node: a2a-filler
Purpose: audio filler node
Example:a2a-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
The following nodes convert audio to text chunks.
-
Node: a2t-openai
Purpose: OpenAI/GPT Speech-to-Text conversion
Example:a2t-openai(language: "de")
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)$/
interim none false none -
Node: a2t-amazon
Purpose: Amazon Transcribe Speech-to-Text conversion
Example:a2t-amazon(language: "de")
Notice: this node requires an API key!This node uses Amazon Trancribe to perform Speech-to-Text (S2T) conversion, i.e., it recognizes speech in the input audio stream and outputs a corresponding text stream.
Port Payload input audio output text Parameter Position Default Requirement key none env.SPEECHFLOW_AMAZON_KEY none secKey none env.SPEECHFLOW_AMAZON_KEY_SEC none region none "eu-central-1" none language none "en" `/^(?:en interim none false none -
Node: a2t-deepgram
Purpose: Deepgram Speech-to-Text conversion
Example:a2t-deepgram(language: "de")
Notice: this node requires an API key!This node performs Speech-to-Text (S2T) conversion, i.e., it recognizes speech in the input audio stream and outputs a corresponding text stream.
Port Payload input audio output text Parameter Position Default Requirement key none env.SPEECHFLOW_DEEPGRAM_KEY none keyAdm none env.SPEECHFLOW_DEEPGRAM_KEY_ADM none model 0 "nova-3" none version 1 "latest" none language 2 "multi" none
The following nodes process text chunks only.
-
Node: t2t-deepl
Purpose: DeepL Text-to-Text translation
Example:t2t-deepl(src: "de", dst: "en")
Notice: this node requires an API key!This node performs translation between English and German languages.
Port Payload input text output text Parameter Position Default Requirement key none env.SPEECHFLOW_DEEPL_KEY none src 0 "de" /^(?:de|en)$/
dst 1 "en" /^(?:de|en)$/
-
Node: t2t-amazon
Purpose: AWS Translate Text-to-Text translation
Example:t2t-amazon(src: "de", dst: "en")
Notice: this node requires an API key!This node performs translation between English and German languages.
Port Payload input text output text Parameter Position Default Requirement key none env.SPEECHFLOW_AMAZON_KEY none secKey none env.SPEECHFLOW_AMAZON_KEY_SEC none region none "eu-central-1" none src 0 "de" /^(?:de|en)$/
dst 1 "en" /^(?:de|en)$/
-
Node: t2t-openai
Purpose: OpenAI/GPT Text-to-Text translation and spelling correction
Example:t2t-openai(src: "de", dst: "en")
Notice: this node requires an OpenAI API key!This node performs translation between English and German languages in the text stream or (if the source and destination language is the same) spellchecking of English or German languages in the text stream. It is based on the remote OpenAI cloud AI service and uses the GPT-4o-mini LLM.
Port Payload input text output text Parameter Position Default Requirement api none "https://api.openai.com" /^https?:\/\/.+?:\d+$/
src 0 "de" /^(?:de|en)$/
dst 1 "en" /^(?:de|en)$/
key none env.SPEECHFLOW_OPENAI_KEY none model none "gpt-5-mini" none -
Node: t2t-ollama
Purpose: Ollama/Gemma Text-to-Text translation and spelling correction
Example:t2t-ollama(src: "de", dst: "en")
Notice: this node requires Ollama to be installed!This node performs translation between English and German languages in the text stream or (if the source and destination language is the same) spellchecking of English or German languages in the text stream. It is based on the local Ollama AI service and uses the Google Gemma 3 LLM.
Port Payload input text output text Parameter Position Default Requirement api none "http://127.0.0.1:11434" /^https?:\/\/.+?:\d+$/
model none "gemma3:4b-it-q4_K_M" none src 0 "de" /^(?:de|en)$/
dst 1 "en" /^(?:de|en)$/
-
Node: t2t-transformers
Purpose: Transformers Text-to-Text translation
Example:t2t-transformers(src: "de", dst: "en")
This node performs translation between English and German languages in the text stream. It is based on local OPUS or SmolLM3 LLMs.
Port Payload input text output text Parameter Position Default Requirement model none "OPUS" /^(?:OPUS|SmolLM3)$/
src 0 "de" /^(?:de|en)$/
dst 1 "en" /^(?:de|en)$/
-
Node: t2t-google
Purpose: Google Cloud Translate Text-to-Text translation
Example:t2t-google(src: "de", dst: "en")
Notice: this node requires a Google Cloud API key and project ID!This node performs translation between multiple languages in the text stream using Google Cloud Translate API. It supports German, English, French, and Italian languages.
Port Payload input text output text Parameter Position Default Requirement key none env.SPEECHFLOW_GOOGLE_KEY none src 0 "de" /^(?:de|en|fr|it)$/
dst 1 "en" /^(?:de|en|fr|it)$/
-
Node: t2t-modify
Purpose: regex-based text modification
Example:t2t-modify(match: "\\b(hello)\\b", replace: "hi $1")
This node allows regex-based modification of text chunks using pattern matching and replacement with support for $n backreferences. It is primarily intended for text preprocessing, cleanup, or transformation tasks.
Port Payload input text output text Parameter Position Default Requirement match 0 "" required replace 1 "" required -
Node: t2t-sentence
Purpose: sentence splitting/merging
Example:t2t-sentence()
This node allows you to ensure that a text stream is split or merged into complete sentences. It is primarily intended to be used after the "a2t-deepgram" node and before "t2t-deepl" or "t2a-elevenlabs" nodes in order to improve overall quality.
Port Payload input text output text Parameter Position Default Requirement -
Node: t2t-subtitle
Purpose: SRT/VTT Subtitle Generation
Example:t2t-subtitle(format: "srt")
This node generates subtitles from the text stream (and its embedded timestamps) in the formats SRT (SubRip) or VTT (WebVTT).
Port Payload input text output text Parameter Position Default Requirement format none "srt" /^(?:srt|vtt)$/ words none false none -
Node: t2t-format
Purpose: text paragraph formatting
Example:t2t-format(width: 80)
This node formats the text stream into lines no longer than a certain width. It is primarily intended for use before writing text chunks to files.
Port Payload input text output text Parameter Position Default Requirement width 0 80 none
The following nodes convert text chunks to audio chunks.
-
Node: t2a-amazon
Purpose: Amazon Polly Text-to-Speech conversion
Example:t2a-amazon(language: "en", voice: "Danielle)
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 language 1 "en" /^(?:de|en)$/
-
Node: t2a-elevenlabs
Purpose: ElevenLabs Text-to-Speech conversion
Example:t2a-elevenlabs(language: "en")
Notice: this node requires an ElevenLabs API key!This node uses ElevenLabs to perform Text-to-Speech (T2S) conversion, i.e., it converts the input text stream into an output audio stream. It is intended to generate speech.
Port Payload input text output audio Parameter Position Default Requirement key none env.SPEECHFLOW_ELEVENLABS_KEY none voice 0 "Brian" /^(?:Brittney|Cassidy|Leonie|Mark|Brian)$/
language 1 "de" /^(?:de|en)$/
speed 2 1.00 n >= 0
7 && n <= 1.2`stability 3 0.5 n >= 0.0 && n <= 1.0
similarity 4 0.75 n >= 0.0 && n <= 1.0
optimize 5 "latency" /^(?:latency|quality)$/
-
Node: t2a-kokoro
Purpose: Kokoro Text-to-Speech conversion
Example:t2a-kokoro(language: "en")
Notice: this currently support English language only!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 input text output audio Parameter Position Default Requirement voice 0 "Aoede" /^(?:Aoede|Heart|Puck|Fenrir)$/
language 1 "en" /^en$/
speed 2 1.25 1.0...1.30
The following nodes process any type of chunk, i.e., both audio and text chunks.
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Node: x2x-filter
Purpose: meta information based filter
Example:x2x-filter(type: "audio", var: "meta:gender", op: "==", val: "male")
This node allows you to filter nodes based on certain criteria. It is primarily intended to be used in conjunction with the "a2a-gender" node and in front of the
elevenlabs
orkokoro
nodes in order to translate with a corresponding voice.Port Payload input text, audio output text, audio Parameter Position Default Requirement type 0 "audio" /^(?:audio|text)$/
name 1 "filter" /^.+$/
var 2 "" /^(?:meta:.+|payload:(?:length|text)|time:(?:start|end))$/
op 3 "==" /^(?:<|<=|==|!=|~~|!~|>=|>)$/
val 4 "" /^.*$/
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Node: x2x-trace
Purpose: data flow tracing
Example:x2x-trace(type: "audio")
This node allows you to trace the audio and text chunk flow through the SpeechFlow graph. It just passes through its chunks (in mode "filter") or acts as a sink for the chunks (in mode "sink"), but always sends information about the chunks to the log. For type "text", the information can be also send to the dashboard.
Port Payload input text, audio output text, audio Parameter Position Default Requirement type 0 "audio" /^(?:audio|text)$/
name 1 "trace" none mode 2 "filter" /^(?:filter|sink)$/
dashboard none none
SpeechFlow has an externally exposed REST/WebSockets API which can be used to control the nodes and to receive information from nodes. For controlling a node you have three possibilities (illustrated by controlling the mode of the "a2a-mute" node):
# use HTTP/REST/GET:
$ curl http://127.0.0.1:8484/api/COMMAND/a2a-mute/mode/silenced
# use HTTP/REST/POST:
$ curl -H "Content-type: application/json" \
--data '{ "request": "COMMAND", "node": "a2a-mute", "args": [ "mode", "silenced" ] }' \
http://127.0.0.1:8484/api
# use WebSockets:
$ wscat -c ws://127.0.0.1:8484/api \
> { "request": "COMMAND", "node": "a2a-mute", "args": [ "mode", "silenced" ] }
For receiving emitted information from nodes, you have to use the WebSockets API (illustrated by the emitted information of the "a2a-meter" node):
# use WebSockets:
$ wscat -c ws://127.0.0.1:8484/api \
< { "response": "NOTIFY", "node": "a2a-meter", "args": [ "meter", "LUFS-S", -35.75127410888672 ] }
SpeechFlow, as a technical cut-through, was initially created in March 2024 for use in the msg Filmstudio context. It was later refined into a more complete toolkit in April 2025 and this way the first time could be used in production. It was fully refactored in July 2025 in order to support timestamps in the streams processing.
Copyright © 2024-2025 Dr. Ralf S. Engelschall
Licensed under GPL 3.0