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Options reference

Here's a detailed reference for all the options accepted by the Echogarden CLI and API.

Related pages:

Text-to-speech

Applies to CLI operations: speak, speak-file, speak-url, speak-wikipedia, API method: synthesize

General:

  • engine: identifier of the synthesis engine to use, such as espeak, vits or google-translate (see the full engine list). Auto-selected if not set
  • language: language code (ISO 639-1), like en, fr, en-US, pt-BR. Auto-detected if not set
  • voice: name of the voice to use. Can be a search string. Auto-selected if not set
  • voiceGender: gender of the voice to use. Optional
  • speed: speech rate factor, relative to default. In the range 0.1..10.0. Defaults to 1.0
  • pitch: pitch factor, relative to default. In the range 0.1..10.0. Defaults to 1.0
  • pitchVariation: pitch variation factor. In the range 0.1..10.0. Defaults to 1.0
  • splitToSentences: split text to sentences before synthesis. Defaults to true
  • ssml: the input is SSML. Defaults to false
  • sentenceEndPause: pause duration (seconds) at end of sentence. Defaults to 0.75
  • segmentEndPause: pause duration (seconds) at end of segment. Defaults to 1.0
  • customLexiconPaths: a list of custom lexicon file paths. Optional
  • alignment: prefix to provide options for alignment. Options detailed in section for alignment
  • subtitles: prefix to provide options for subtitles. Options detailed in section for subtitles
  • languageDetection: prefix to provide options for text language detection. Options detailed in section for text language detection

Plain text processing:

  • plainText.paragraphBreaks: split to paragraphs based on single (single), or double (double) line breaks. Defaults to double
  • plainText.whitespace: determines how to process whitespace within paragraphs. Can be preserve (leave as is), removeLineBreaks (convert line breaks to spaces) or collapse (convert runs of whitespace characters, including line breaks, to a single space character). Defaults to collapse

Post-processing:

  • postProcessing.normalizeAudio: should normalize output audio. Defaults to true
  • postProcessing.targetPeak: target peak (decibels) for normalization. Defaults to -3
  • postProcessing.maxGainIncrease: max gain increase (decibels) when performing normalization. Defaults to 30
  • postProcessing.speed: target speed for time stretching. Defaults to 1.0
  • postProcessing.pitch: target pitch for pitch shifting. Defaults to 1.0
  • postProcessing.timePitchShiftingMethod: method for time and pitch shifting. Can be sonic or rubberband. Defaults to sonic
  • postProcessing.rubberband: prefix for RubberBand options (TODO: document options)

Output audio format:

  • outputAudioFormat.codec: Codec identifier (Note: API only. CLI uses file extensions instead), can be wav, mp3, opus, m4a, ogg, flac. Leaving as undefined would return a raw audio structure (see more information at the API documentation. Optional
  • outputAudioFormat.bitrate: Custom bitrate for encoding, applies only to mp3, opus, m4a, ogg. By default, bitrates are selected between 48Kbps and 64Kbps, to provide a good speech quality while minimizing file size. Optional

VITS:

  • vits.speakerId: speaker ID, for VITS models that support multiple speakers. Defaults to 0
  • vits.provider: ONNX execution provider to use. Can be cpu, dml (DirectML-based GPU acceleration - Windows only), or cuda (Linux only - requires CUDA Toolkit 12.x and cuDNN 9.x to be installed). Using GPU acceleration for VITS may or may not be faster than CPU, depending on your hardware. Defaults to cpu

eSpeak:

  • espeak.rate: speech rate, in eSpeak units. Overrides speed when set
  • espeak.pitch: pitch, in eSpeak units. Overrides pitch when set
  • espeak.pitchRange: pitch range, in eSpeak units. Overrides pitchVariation when set
  • espeak.useKlatt: use the Klatt synthesis method. Defaults to false

SAM:

  • sam.pitch: pitch value, between 0..255. Overrides pitch when set
  • sam.speed: speed value, between 0..255. Overrides speed when set
  • sam.mouth: mouth value, between 0..255 (defaults to 128)
  • sam.throat: throat value, between 0..255 (defaults to 128)

SAPI:

  • sapi.rate: SAPI speech rate, in its native units. An integer number between -10 and 10. Setting speed would apply time stretching instead. The two options can be used together

Microsoft Speech Platform:

  • msspeech.rate: same units and effects as the SAPI speech rate

Coqui Server:

  • coquiServer.serverUrl: server URL
  • coquiServer.speakerId: speaker ID (if applicable)

Google Cloud:

  • googleCloud.apiKey: API key (required)
  • googleCloud.pitchDeltaSemitones: pitch delta in semitones. Overrides pitch when set
  • googleCloud.customVoice.model: name of custom voice
  • googleCloud.customVoice.reportedUsage: reported usage of custom voice

Azure Cognitive Services:

  • microsoftAzure.subscriptionKey: subscription key (required)
  • microsoftAzure.serviceRegion: service region (required)
  • microsoftAzure.pitchDeltaHz: pitch delta in Hz. Overrides pitch when set

Amazon Polly:

  • amazonPolly.region: region (required)
  • amazonPolly.accessKeyId: access key ID (required)
  • amazonPolly.secretAccessKey: secret access key (required)
  • amazonPolly.pollyEngine: Amazon Polly engine kind, can be standard or neural. Defaults to neural
  • amazonPolly.lexiconNames: an array of lexicon names. Optional

OpenAI Cloud:

  • openAICloud.apiKey: API key (required)
  • openAICloud.organization: organization identifier. Optional
  • openAICloud.baseURL: override the default base URL for the API. Optional
  • openAICloud.model: model to use. Can be either tts-1 or tts-1-hd. Defaults to tts-1
  • openAICloud.timeout: request timeout. Optional
  • openAICloud.maxRetries: maximum retries on failure. Defaults to 10

Elevenlabs:

  • elevenLabs.apiKey: API key (required)
  • elevenLabs.stability: stability. Defaults to 0.5
  • elevenLabs.similarityBoost: similarity boost. Defaults to 0.5

Google Translate:

  • googleTranslate.tld: top level domain to connect to. Can change the dialect for a small number of voices. For example us gives American English for en, while com gives British English for en. Defaults to us

Microsoft Edge:

  • microsoftEdge.trustedClientToken: trusted client token (required). A special token required to use the service
  • microsoftEdge.pitchDeltaHz: pitch delta in Hz. Overrides pitch when set

Voice list request

Applies to CLI operation: list-voices, API method: requestVoiceList

General:

  • language: language code to filter by (optional)
  • voice: name or name pattern to filter by (optional)
  • voiceGender: gender to filter by (optional)

Also accepted are the following engine-specific options that may be required in order to retrieve the voice list:

  • googleCloud.apiKey
  • microsoftAzure.subscriptionKey, microsoftAzure.serviceRegion
  • amazonPolly.region, amazonPolly.accessKeyId, amazonPolly.secretAccessKey
  • elevenLabs.apiKey
  • microsoftEdge.trustedClientToken

Speech-to-text

Applies to CLI operation: transcribe, API method: recognize

General:

  • engine: identifier of the recognition engine to use, such as whisper or vosk (see the full engine list)
  • language: language code (ISO 639-1) for the audio, like en, fr, de. Auto-detected if not set
  • crop: crop to active parts using voice activity detection before starting recognition. Defaults to true
  • isolate: apply source separation to isolate voice before starting recognition. Defaults to false
  • alignment: prefix to provide options for alignment. Options detailed in section for alignment
  • languageDetection: prefix to provide options for language detection. Options detailed in section for speech language detection
  • subtitles: prefix to provide options for subtitles. Options detailed in section for subtitles
  • vad: prefix to provide options for voice activity detection when crop is set to true. Options detailed in section for voice activity detection
  • sourceSeparation: prefix to provide options for source separation when isolate is set to true. Options detailed in section for source separation

Whisper:

  • whisper.model: selects which Whisper model to use. Can be tiny, tiny.en, base, base.en, small, small.en, medium, medium.en or large-v3-turbo. Defaults to tiny or tiny.en
  • whisper.temperature: temperature setting for the text decoder. Impacts the amount of randomization for token selection. It is recommended to leave at 0.1 (close to no randomization - almost always chooses the top ranked token) or choose a relatively low value (0.25 or lower) for best results. Defaults to 0.1
  • whisper.prompt: initial text to give the Whisper model. Can be a vocabulary, or example text of some sort. Note that if the prompt is very similar to the transcript, the model may intentionally avoid producing the transcript tokens as it may assume that they have already been transcribed. Optional
  • whisper.topCandidateCount: the number of top candidate tokens to consider. Defaults to 5
  • whisper.punctuationThreshold: the minimal probability for a punctuation token, included in the top candidates, to be chosen unconditionally. A lower threshold encourages the model to output more punctuation characters. Defaults to 0.2
  • whisper.autoPromptParts: use previous part's recognized text as the prompt for the next part. Disabling this may help to prevent repetition carrying over between parts, in some cases. Defaults to true (Note: currently always disabled for large-v3-turbo model due to an apparent issue with corrupt output when prompted)
  • whisper.maxTokensPerPart: maximum number of tokens to decode for each audio part. Defaults to 250
  • whisper.suppressRepetition: attempt to suppress decoding of repeating token patterns. Defaults to true
  • whisper.repetitionThreshold: minimal repetition / compressibility score to cause a part not to be auto-prompted to the next part. Defaults to 2.4
  • whisper.decodeTimestampTokens: enable/disable decoding of timestamp tokens. Setting to false can reduce the occurrence of hallucinations and token repetition loops, possibly due to the overall reduction in the number of tokens decoded. This has no impact on the accuracy of timestamps, since they are derived independently using cross-attention weights. However, there are cases where this can cause the model to end a part prematurely, especially in singing and less speech-like voice segments, or when there are multiple speakers. Defaults to true
  • whisper.timestampAccuracy: timestamp accuracy. can be medium or high. medium uses a reduced subset of attention heads for alignment, high uses all attention heads and is thus more accurate at the word level, but slower for larger models. Defaults to high for the tiny and base models, and medium for the larger models
  • whisper.encoderProvider: identifier for the ONNX execution provider to use with the encoder model. Can be cpu, dml (DirectML-based GPU acceleration - Windows only), or cuda (Linux only - requires CUDA Toolkit 12.x and cuDNN 9.x to be installed). In general, GPU-based encoding should be significantly faster. Defaults to cpu, or dml if available
  • whisper.decoderProvider: identifier for the ONNX execution provider to use with the decoder model. Can be cpu, dml (DirectML-based GPU acceleration - Windows only), or cuda (Linux only - requires CUDA Toolkit 12.x and cuDNN 9.x to be installed). Using GPU acceleration for the decoder may be faster than CPU, especially for larger models, but that depends on your particular combination of CPU and GPU. Defaults to cpu, and on Windows, dml if available for larger models (small, medium, large)
  • whisper.seed: provide a custom random seed for token selection when temperature is greater than 0. Uses a constant seed by default to ensure reproducibility

Whisper.cpp:

  • whisperCpp.model: selects which whisper.cpp model to use. Can be tiny, tiny.en, base, base.en, small, small.en, medium, medium.en, large (same as large-v2), large-v1, large-v2, large-v3, large-v3-turbo. The following quantized models are also supported: tiny-q5_1, tiny.en-q5_1, tiny.en-q8_0,base-q5_1, base.en-q5_1, small-q5_1, small.en-q5_1, medium-q5_0, medium.en-q5_0, large-v2-q5_0, large-v3-q5_0, large-v3-turbo-q5_0. Defaults to base or base.en
  • whisperCpp.executablePath: a path to a custom whisper.cpp main executable (currently required for macOS)
  • whisperCpp.build: type of whisper.cpp build to use. Can be set to cpu, cublas-12.4.0 or custom. By default, builds are auto-selected and downloaded for Windows x64 (cpu, cublas-12.4.0) and Linux x64 (cpu). Using other builds requires providing a custom executablePath, which will automatically set this option to custom
  • whisperCpp.threadCount: number of threads to use, defaults to 4
  • whisperCpp.splitCount: number of splits of the audio data to process in parallel (called --processors in the whisper.cpp CLI). A value greater than 1 can increase memory use significantly, reduce timing accuracy, and slow down execution in some cases. Defaults to 1 (highly recommended)
  • whisperCpp.enableGPU: enable GPU processing. Setting to true will try to use a CUDA build, if available for your system. Defaults to true when a CUDA-enabled build is selected via whisperCpp.build, otherwise false. If a custom build is used, it will enable or disable GPU for that build
  • whisperCpp.topCandidateCount: the number of top candidate tokens to consider. Defaults to 5
  • whisperCpp.beamCount: the number of decoding paths to use during beam search. Defaults to 5
  • whisperCpp.temperature: set temperature. Defaults to 0.0
  • whisperCpp.temperatureIncrement: set temperature increment. Defaults to 0.2
  • whisperCpp.repetitionThreshold: minimal repetition / compressibility score to cause a decoded segment to be discarded. Defaults to 2.4
  • whisperCpp.prompt: initial text to give the Whisper model. Can be a vocabulary, or example text of some sort. Note that if the prompt is very similar to the transcript, the model may intentionally avoid producing the transcript tokens as it may assume that they have already been transcribed. Optional
  • whisperCpp.enableDTW: enable whisper.cpp's own experimental DTW-based token alignment to be used to derive timestamps. Defaults to false
  • whisperCpp.enableFlashAttention: enable flash attention. Can significantly increase performance for some configurations (Note: setting this to true will cause enableDTW to always be set to false since it's not compatible with flash attention). Defaults to false
  • whisperCpp.verbose: show all CLI messages during execution. Defaults to false

Vosk:

  • vosk.modelPath: path to the Vosk model to be used

Silero:

  • silero.modelPath: path to a Silero model. Note that latest en, de, fr and uk models are automatically installed when needed based on the selected language. This should only be used to manually specify a different model, otherwise specify language instead
  • silero.provider: ONNX execution provider to use. Can be cpu, dml (DirectML-based GPU acceleration - Windows only), or cuda (Linux only - requires CUDA Toolkit 12.x and cuDNN 9.x to be installed). Defaults to cpu, or dml if available

Google Cloud:

  • googleCloud.apiKey: Google Cloud API key (required)
  • googleCloud.alternativeLanguageCodes: An array of alternative language codes. Optional
  • googleCloud.profanityFilter: censor profanity. Defaults to false
  • googleCloud.autoPunctuation: add punctuation automatically. Defaults to true
  • googleCloud.useEnhancedModel: use enhanced model. Defaults to true

Azure Cognitive Services:

  • microsoftAzure.subscriptionKey: subscription key (required)
  • microsoftAzure.serviceRegion: service region (required)

Amazon Transcribe:

  • amazonTranscribe.region: region (required)
  • amazonTranscribe.accessKeyId: access key ID (required)
  • amazonTranscribe.secretAccessKey: secret access key (required)

OpenAI Cloud:

  • openAICloud.apiKey: API key (required)
  • openAICloud.model: model to use. When using the default provider (OpenAI), can only be whisper-1. For a custom provider, like Groq, see its documentation
  • openAICloud.organization: organization identifier. Optional
  • openAICloud.baseURL: override the default endpoint used by the API. For example, set https://api.groq.com/openai/v1 to use Groq's OpenAI-compatible API instead of the default one. Optional
  • openAICloud.temperature: temperature. Choosing 0 uses a dynamic temperature approach. Defaults to 0
  • openAICloud.prompt: initial prompt for the model. Optional
  • openAICloud.timeout: request timeout. Optional
  • openAICloud.maxRetries: maximum retries on failure. Defaults to 10
  • openAICloud.requestWordTimestamps: request word timestamps from the server. Defaults to true for the default OpenAI endpoint, and false if a custom one is set using baseURL

Speech-to-transcript alignment

Applies to CLI operation: align, API method: align

General:

  • engine: alignment algorithm to use, can be dtw, dtw-ra or whisper. Defaults to dtw
  • language: language code for the audio and transcript (ISO 639-1), like en, fr, en-US, pt-BR. Auto-detected from transcript if not set
  • crop: crop to active parts using voice activity detection before starting. Defaults to true
  • isolate: apply source separation to isolate voice before starting alignment. Defaults to false
  • customLexiconPaths: an array of custom lexicon file paths. Optional
  • subtitles: prefix to provide options for subtitles. Options detailed in section for subtitles
  • vad: prefix to provide options for voice activity detection when crop is set to true. Options detailed in section for voice activity detection
  • sourceSeparation: prefix to provide options for source separation when isolate is set to true. Options detailed in section for source separation

Plain text processing:

  • plainText.paragraphBreaks: split transcript to paragraphs based on single (single), or double (double) line breaks. Defaults to double
  • plainText.whitespace: determines how to process whitespace within transcript paragraphs. Can be preserve (leave as is), removeLineBreaks (convert line breaks to spaces) or collapse (convert runs of whitespace characters, including line breaks, to a single space character). Defaults to collapse

DTW:

  • dtw.granularity: adjusts the MFCC frame width and hop size based on the profile selected. Can be set to either xx-low (400ms width, 160ms hop), x-low (200ms width, 80ms hop), low (100ms width, 40ms hop), medium (50ms width, 20ms hop), high (25ms width, 10ms hop), x-high (20ms width, 5ms hop). For multi-pass processing, multiple granularities can be provided, like dtw.granularity=['xx-low','medium']. Auto-selected by default.
  • dtw.windowDuration: sets the maximum duration of the Sakoe-Chiba window when performing DTW alignment. The value can be specified in seconds, like 240, or as an integer percentage (formatted like 15%), relative to the total duration of the source audio. The estimated memory requirement is shown in the log before alignment starts. Recommended to be set to at least 10% - 20% of total audio duration. For multi-pass processing, multiple durations can be provided (which can mix absolute and relative values), like dtw.windowDuration=['15%',20]. Auto-selected by default

DTW-RA:

  • recognition: prefix to provide recognition options when using dtw-ra method, for example: setting recognition.engine = whisper and recognition.whisper.model = base.en
  • dtw.phoneAlignmentMethod: algorithm to use when aligning phones: can either be set to dtw or interpolation. Defaults to dtw

Whisper:

Applies to the whisper engine only. To provide Whisper options for dtw-ra, use recognition.whisper instead.

  • whisper.model: Whisper model to use. Defaults to tiny or tiny.en
  • whisper.endTokenThreshold: minimal probability to accept an end token for a recognized part. The probability is measured via the softmax between the end token's logit and the second highest logit. You can try to adjust this threshold in cases the model is ending a part with too few, or many tokens decoded. Defaults to 0.9. On the last audio part, it is always effectively set to Infinity, to ensure the remaining transcript tokens are decoded in full
  • whisper.maxTokensPerPart: maximum number of tokens to decode per part. Should help avoid edge cases where the model never reaches an end token for the part, which otherwise may cause the model to decode too many tokens and eventually crash. Defaults to 250
  • whisper.timestampAccuracy: timestamp accuracy. can be medium or high. medium uses a reduced subset of attention heads for alignment, high uses all attention heads and is thus more accurate at the word level, but slower for larger models. Defaults to high for the tiny and base models, and medium for the larger models
  • whisper.encoderProvider: encoder ONNX provider. See details in recognition section above
  • whisper.decoderProvider: decoder ONNX provider. See details in recognition section above

Speech-to-text translation

Applies to CLI operation: translate-speech, API method: translateSpeech

General:

  • engine: only whisper supported
  • sourceLanguage: the source language code for the input speech. Auto-detected if not set
  • targetLanguage: the target language code for the output speech. Only en (English) supported by the whisper engine. Optional
  • crop: crop to active parts using voice activity detection before starting. Defaults to true
  • isolate: apply source separation to isolate voice before starting speech translation. Defaults to false
  • languageDetection: prefix to provide options for language detection. Options detailed in section for speech language detection
  • vad: prefix to provide options for voice activity detection when crop is set to true. Options detailed in section for voice activity detection
  • sourceSeparation: prefix to provide source separation options when isolate is to true

Whisper:

  • whisper: prefix to provide options for the Whisper model. Same options as detailed in the recognition section above

Whisper.cpp:

  • whisper.cpp: prefix to provide options for the Whisper.cpp model. Same options as detailed in the recognition section above

OpenAI Cloud:

  • openAICloud: prefix to provide options for OpenAI cloud. Same options as detailed in the recognition section above

Text-to-text translation

Applies to CLI operation: translate-text, API method: translateText

General:

  • engine: only google-translate supported
  • sourceLanguage: the source language code for the input text. Auto-detected if not set
  • targetLanguage: the target language code for the output text. Required
  • languageDetection: language detection options. Optional
  • plainText: plain text processing options. Optional

Google Translate:

  • googleTranslate.tld: top-level domain to request from. Defaults to com
  • googleTranslate.maxCharactersPerPart: maximum number of characters in each part requested from the server. Defaults to 2000

Speech-to-translated-transcript alignment

Applies to CLI operation: align-translation, API method: alignTranslation

General:

  • engine: alignment algorithm to use, can only be whisper. Defaults to whisper
  • sourceLanguage: language code for the source audio (ISO 639-1), like en, fr, zh, etc. Auto-detected from audio if not set
  • targetLanguage: language code for the translated transcript. Can only be en for now. Defaults to en
  • crop: crop to active parts using voice activity detection before starting. Defaults to true
  • isolate: apply source separation to isolate voice before starting alignment. Defaults to false
  • subtitles: prefix to provide options for subtitles. Options detailed in section for subtitles
  • vad: prefix to provide options for voice activity detection when crop is set to true. Options detailed in section for voice activity detection
  • sourceSeparation: prefix to provide options for source separation when isolate is set to true. Options detailed in section for source separation

Whisper:

  • whisper.model: Whisper model to use. Only multilingual models can be used. Defaults to tiny
  • whisper.endTokenThreshold: see details in the alignment section above
  • whisper.encoderProvider: encoder ONNX execution provider. See details in recognition section above
  • whisper.decoderProvider: decoder ONNX execution provider. See details in recognition section above

Speech-to-transcript-and-translation alignment

Applies to CLI operation: align-transcript-and-translation, API method: alignTranscriptAndTranslation

General:

  • engine: can only be two-stage. Defaults to two-stage
  • sourceLanguage: language code for the source audio (ISO 639-1), like en, fr, zh, etc. Auto-detected from audio if not set
  • targetLanguage: language code for the translated transcript. Can only be en for now. Defaults to en
  • crop: crop to active parts using voice activity detection before starting. Defaults to true
  • isolate: apply source separation to isolate voice before starting alignment. Defaults to false
  • alignment: prefix to provide options for alignment. Options detailed in section for alignment
  • timelineAlignment: prefix to provide options for timeline alignment. Options detailed in section for timeline alignment
  • vad: prefix to provide options for voice activity detection when crop is set to true. Options detailed in section for voice activity detection
  • sourceSeparation: prefix to provide options for source separation when isolate is set to true. Options detailed in section for source separation
  • subtitles: prefix to provide options for subtitles. Options detailed in section for subtitles

Timeline-to-translated-text alignment

Applies to CLI operation: align-timeline-translation, API method: alignTimelineTranslation

General:

  • engine: alignment engine to use. Can only be e5. Defaults to e5
  • sourceLanguage: language code for the source timeline. Auto-detected from timeline if not set
  • targetLanguage: language code for the translated transcript. Auto-detected if not set
  • audio: spoken audio to play when previewing the result in the CLI (not required or used by the alignment itself). Optional
  • languageDetection: prefix to provide options for language detection. Options detailed in section for text language detection
  • subtitles: prefix to provide options for subtitles. Options detailed in section for subtitles

E5:

  • e5.model: E5 model to use. Defaults to e5-small-fp16 (support for additional models will be added in the future)

Language detection

Speech language detection

Applies to CLI operation: detect-speech-langauge, API method: detectSpeechLangauge

General:

  • engine: whisper or silero. Defaults to whisper
  • defaultLanguage: language to fallback to when confidence for top candidate of is low. Defaults to en
  • fallbackThresholdProbability: confidence threshold to cause fallback. Defaults to 0.05
  • crop: crop to active parts using voice activity detection before starting. Defaults to true (recommended, otherwise inactive sections may skew the probabilities towards various random languages)
  • vad: prefix to provide options for voice activity detection when crop is set to true. Options detailed in section for voice activity detection

Whisper:

  • whisper.model: Whisper model to use. See model list in the recognition section
  • whisper.temperature: impacts the distribution of candidate languages when applying the softmax function to compute language probabilities over the model output. Higher temperature causes the distribution to be more uniform, while lower temperature causes it to be more strongly weighted towards the best scoring candidates. Defaults to 1.0
  • whisper.encoderProvider: encoder ONNX execution provider. See details in recognition section above
  • whisper.decoderProvider: decoder ONNX execution provider. See details in recognition section above

Silero:

  • silero.provider: ONNX execution provider to use. Can be cpu, dml (DirectML-based GPU acceleration - Windows only), or cuda (Linux only - requires CUDA Toolkit 12.x and cuDNN 9.x to be installed). Using GPU may be faster, but the initialization overhead is larger. Note: dml provider seems to be unstable at the moment for this model. Defaults to cpu

Text language detection

Applies to CLI operation: detect-text-langauge, API method: detectTextLangauge

General:

  • engine: tinyld or fasttext. Defaults to tinyld
  • defaultLanguage: language to fallback to when confidence for top candidate is low. Defaults to en
  • fallbackThresholdProbability: confidence threshold to cause fallback. Defaults to 0.05

Voice activity detection

Applies to CLI operation: detect-voice-activity, API method: detectVoiceActivity

General:

  • engine: VAD engine to use. Can be webrtc, silero, rnnoise, or adaptive-gate. Defaults to silero
  • activityThreshold: minimum predicted probability for determining a frame as having speech activity. Defaults to 0.5

WebRTC:

  • webrtc.frameDuration: WebRTC frame duration (ms). Can be 10, 20 or 30. Defaults to 30
  • webrtc.mode: WebRTC mode (aggressiveness). Can be 0, 1, 2 or 3. Defaults to 1

Silero:

  • silero.frameDuration: Silero frame duration (ms). Can be 30, 60 or 90. Defaults to 90
  • silero.provider: ONNX provider to use. Can be cpu, dml (DirectML-based GPU acceleration - Windows only), or cuda (Linux only - requires CUDA Toolkit 12.x and cuDNN 9.x to be installed). Using GPU is likely to be slower than CPU due to inference being independently executed on each audio frame. Defaults to cpu (recommended)

Speech denoising

Applies to CLI operation: denoise, API method: denoise

General:

  • engine: rnnoise or nsnet2. Defaults to rnnoise

Post-processing:

  • postProcessing.normalizeAudio: should normalize output audio. Defaults to false
  • postProcessing.targetPeak: target peak (decibels) for normalization. Defaults to -3
  • postProcessing.maxGainIncrease: max gain increase (decibels) when performing normalization. Defaults to 30
  • postProcessing.dryMixGain: gain (decibels) of dry (original) signal to mix back to the denoised (wet) signal. Defaults to -100

NSNet2:

  • nsnet2.model: can be baseline-16khz or baseline-48khz. Defaults to baseline-48khz
  • nsnet2.provider: ONNX execution provider (Note: dml provider seems to fail with these models). Defaults to cpu
  • maxAttenuation: maximum amount of attenuation, in decibels, applied to an FFT bin when filtering the audio frames. Defaults to 30

Source separation

Applies to CLI operation: isolate, API method: isolate

General:

  • engine: can only be mdx-net

MDX-NET:

  • mdxNet.model: model to use. Currently available models are UVR_MDXNET_1_9703, UVR_MDXNET_2_9682, UVR_MDXNET_3_9662, UVR_MDXNET_KARA, and higher quality models UVR_MDXNET_Main and Kim_Vocal_2. Defaults to UVR_MDXNET_1_9703
  • mdxNet.provider: ONNX execution provider to use. Can be cpu, dml (DirectML-based GPU acceleration - Windows only), or cuda (Linux only - requires CUDA Toolkit 12.x and cuDNN 9.x to be installed). Defaults to dml if available (Windows) or cpu (other platforms)

Common options

Subtitles

These are shared between text-to-speech, speech-to-text and alignment operations, usually prefixed with subtitles..

  • mode: subtitle generation mode. Can be segment (ensures each segment starts at a new cue), sentence (ensures each sentence starts at a new cue), word (one word per cue, no punctuation included), phone (one phone per cue), word+phone (include both word and phone cues, with overlapping time ranges), line (each text line is made a separate cue). Defaults to sentence
  • maxLineCount: maximum number of lines per cue. Defaults to 2
  • maxLineWidth: maximum characters in a line. Defaults to 42
  • minWordsInLine: minimum number of remaining words to break to a new line. Defaults to 4
  • separatePhrases: try to separate phrases or sentences in new lines or cues, if possible. Defaults to true
  • maxAddedDuration: maximum extra time (in seconds) that may be added after a cue's end time. This gives the reader additional time to read the cue, and also ensures that very short duration cues aren't shown in a flash. Defaults to 3.0

Note: options maxLineCount, maxLineWidth, minWordsInLine, separatePhrases, are only effective when using the segment and sentence modes, and are ignored in all other modes. maxAddedDuration doesn't apply to modes word, phone and word+phone (they always use the exact start and end timestamps).

Global options

On the CLI, global options can be used with any operation. To set global options via the API, use the setGlobalOption(key, value) method (see the API reference for more details).

  • ffmpegPath: sets a custom path for the FFmpeg executable
  • soxPath: sets a custom path for the SoX executable
  • packageBaseURL: sets a custom base URL for the remote package repository used to download missing packages. Default is https://huggingface.co/echogarden/echogarden-packages/resolve/main/. If huggingface.co isn't accessible in your location, you can set to use a mirror by changing huggingface.co to an alternative domain like hf-mirror.com
  • logLevel: adjusts the quantity of log messages shown during processing. Possible values: silent, output, error, warning, info, trace. Defaults to info

CLI options

These options are for the CLI only.

  • --play, --no-play: enable/disable audio playback. Defaults to play if there is no output file specified
  • --player: audio player to use. Can be audio-io (uses the audio-io package to directly output to native OS audio buffers) or sox (requires sox to be available on path on macOS, auto-downloaded on other platforms). Defaults to audio-io
  • --overwrite, --no-overwrite: overwrite/keep existing files. Doesn't overwrite by default
  • --debug, --no-debug: show/hide the full details of JavaScript errors, if they occur. Disabled by default
  • --config=...: path to configuration file to use. Defaults to echogarden.config or echogarden.config.json, if found at the current directory

Using a configuration file

The CLI supports loading options from a default or custom configuration file in various formats. See the CLI Guide for more details.