echogarden 1.1.0 → 1.2.0

package/dist/utilities/Utilities.js.map

@@ -1 +1 @@
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package/docs/API.md

@@ -290,21 +290,14 @@ Timeline object.
 
 ### `setGlobalOption(key, value)`
 
-Set a global option.
+Sets a global option.
 
-Supported keys:
+See the [options reference](Options.md) for more details about the available global options.
 
-* `'ffmpegPath'`: override FFmpeg executable path
-* `'soxPath'`: override SoX executable path
 
 ### `getGlobalOption(key)`
 
-Get a global option.
-
-Supported keys:
-
-* `'ffmpegPath'`: get FFmpeg executable path
-* `'soxPath'`: get SoX executable path
+Gets a global option.
 
 #### Returns:
 

package/docs/CLI.md

@@ -118,6 +118,73 @@ This would align the audio file `dutch-speech.mp3` with the translated transcrip
 echogarden align-translation dutch-speech.mp3 english-translation.txt result.srt result.json
 ```
 
+## Language detection
+
+**Task**: Given audio or textual input, try to identify which language it is spoken or written in.
+
+Try to identify the language of an audio file containing speech, and print the probabilities to the terminal:
+```bash
+echogarden detect-speech-language speech.mp3
+```
+
+Try to identify the language of a text file, and print the probabilities to the terminal:
+```bash
+echogarden detect-text-language story.txt
+```
+
+Try to identify the language of a text file, and store the detailed probabilities in a JSON file:
+```bash
+echogarden detect-text-language story.txt detection-results.json
+```
+
+## Voice activity detection
+
+**Task**: Given an audio file, try to classify which parts of the audio contain speech, and which don't.
+
+This would apply VAD and play the audio, synchronized with `speech` and `nonspeech` indicators, printed to the terminal.
+```bash
+echogarden detect-voice-activity speech.mp3
+```
+
+This would apply VAD and store the results in a timeline JSON file.
+```bash
+echogarden detect-voice-activity speech.mp3 timeline.json
+```
+
+## Speech denoising
+
+**Task**: Attempt to reduce the amount of background noise in a spoken recording.
+
+This would apply denoising and play the denoised audio:
+```bash
+echogarden denoise speech.mp3
+```
+
+This would apply denoising, and save the denoised audio to a file:
+```bash
+echogarden denoise speech.mp3 denoised-speech.mp3
+```
+
+## Source separation
+
+**Task**: Try to isolate a vocal track (or other type of track, depending on model used), from the audio.
+
+This would apply source separation and play the isolated audio:
+```bash
+echogarden isolate voice-with-music.mp3
+```
+
+This would apply source separation, and save both the isolated and background audio:
+```bash
+echogarden isolate voice-with-music.mp3 voice-isolated.mp3
+```
+
+Written files would be:
+```
+voice-isolated.mp3
+voice-isolated.background.mp3
+```
+
 ## Using output templates to split the output to multiple files
 
 Echogarden can split the output to multiple parts based on the segment boundaries detected. For example:
@@ -151,7 +218,7 @@ echogarden transcribe speech.mp3 --no-play
 
 ## File overwriting
 
-By default, the CLI doesn't overwrite existing files. If an output file `out.mp3` already exists, it will save it as `out_001.mp3`.
+By default, the CLI doesn't overwrite existing files. If an output file `out.mp3` already exists, it will save it as `out_00x.mp3`.
 
 To have existing files be overwritten, you can pass the `--overwrite` option.
 
@@ -161,9 +228,26 @@ Since there are many possible configuration options, it may be more convenient t
 
 When a file named `echogarden.config` is found at the current directory, it will be loaded automatically and its content would be used as default options. You can also specify a particular configuration file path with the option `--config=path/to/your-config-file.config`.
 
-The configuration file format is simple and has a dedicated section for each command (all `speak-` commands are grouped together under `speak`):
+The configuration file format is simple and has a dedicated section for each command (all `speak-` commands are grouped together under `speak`), `global` section for global API options, and `cli` for CLI-specific options. `#` is used as a comment character.
 
+Example:
 ```conf
+[global]
+
+# Custom remote packages base URL:
+packageBaseURL = https://hf-mirror.com/echogarden/echogarden-packages/resolve/main/
+
+# Log level:
+logLevel = info
+
+[cli]
+
+# Should play audio in the terminal:
+play = true
+
+# Overwrite existing files:
+overwrite = true
+
 [speak]
 
 # Engine for synthesis:
@@ -180,8 +264,9 @@ customLexiconPaths = ["lexicon1.json", "lexicon2.json"]
 # Engine for recognition:
 engine = whisper
 
-# Whisper model to use:
+# Whisper options:
 whisper.model = tiny
+whisper.temperature = 0.15
 ```
 
 ### JSON configuration file
@@ -216,74 +301,6 @@ Flattened property names are also accepted:
 }
 ```
 
-## Other operations
-
-### Language detection
-
-**Task**: Given audio or textual input, try to identify which language it is spoken or written in.
-
-Try to identify the language of an audio file containing speech, and print the probabilities to the terminal:
-```bash
-echogarden detect-speech-language speech.mp3
-```
-
-Try to identify the language of a text file, and print the probabilities to the terminal:
-```bash
-echogarden detect-text-language story.txt
-```
-
-Try to identify the language of a text file, and store the detailed probabilities in a JSON file:
-```bash
-echogarden detect-text-language story.txt detection-results.json
-```
-
-### Voice activity detection
-
-**Task**: Given an audio file, try to classify which parts of the audio contain speech, and which don't.
-
-This would apply VAD and play the audio, synchronized with `speech` and `nonspeech` indicators, printed to the terminal.
-```bash
-echogarden detect-voice-activity speech.mp3
-```
-
-This would apply VAD and store the results in a timeline JSON file.
-```bash
-echogarden detect-voice-activity speech.mp3 timeline.json
-```
-
-### Speech denoising
-
-**Task**: Attempt to reduce the amount of background noise in a spoken recording.
-
-This would apply denoising and play the denoised audio:
-```bash
-echogarden denoise speech.mp3
-```
-
-This would apply denoising, and save the denoised audio to a file:
-```bash
-echogarden denoise speech.mp3 denoised-speech.mp3
-```
-
-### Source separation
-
-**Task**: Try to isolate a vocal track (or other type of track, depending on model used), from the audio.
-
-This would apply source separation and play the isolated audio:
-```bash
-echogarden isolate voice-with-music.mp3
-```
-
-This would apply source separation, and save both the isolated and background audio:
-```bash
-echogarden isolate voice-with-music.mp3 voice-isolated.mp3
-```
-
-Written files would be:
-```
-voice-isolated.mp3
-voice-isolated.background.mp3
-```
 
 ## Information and lists
 
@@ -324,9 +341,3 @@ Uninstall one or more expansion packages
 
 Show a list of installed expansion packages
 
-## General CLI options
-
-* `--play`, `--no-play`: enable/disable audio playback. Defaults to play if there is no output file specified
-* `--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

package/docs/Contributing.md

@@ -6,15 +6,15 @@ However, there are many areas where contributions can be made.
 
 ## Report any issue or bug you encounter
 
-First, check the issue tracker, as well as the [task list](Tasklist.md) to see if the problem is already known to me. The task list allows me to efficiently document and organize a large quantity of small issues, enhancements or ideas that would otherwise flood an issue tracker with lots of unimportant entries, be ignored, or forgotten entirely.
+First, check the issue tracker, as well as the [task list](Tasklist.md) to see if the problem is already known to me. The task list allows me to efficiently document and organize a large number of small issues, enhancements or ideas that would otherwise flood an issue tracker with lots of unimportant entries, be ignored, or be forgotten entirely.
 
 If you find the issue you're encountering in the task list, you can still open an issue to discuss it. This allows me to know that someone cares about a particular issue, and I may give it higher priority.
 
 There might be some obvious errors that have gone unreported. Especially if:
-* You're using the macOS architecture: I don't have access to a macOS machine, so personally I did not and cannot perform testing on that platform.
+* You're using the macOS architecture: I don't have access to a macOS machine, so personally, I did not and cannot perform testing on that platform.
 * You're using cloud services: There may be changes in the service that will require updating the code. I don't often test they work correctly, since my trial periods in Google, Microsoft and Amazon have all expired, thus testing requires me to use paid requests.
 
-Since there is a large number of engines, options and behaviors, it is difficult for me to test everything at each release. It is likely there are edge cases that I didn't think about or noticed.
+Since there is a large number of engines, options and behaviors, it is difficult for me to test everything at each release. There are likely edge cases that I didn't think about or noticed.
 
 In any case, please let me know if you get any unexpected error message or surprising behavior that you care about, and I'll try to prioritize it, if possible.
 
@@ -30,7 +30,7 @@ If the problem is serious, you can report it, and we'll see what we can do.
 
 ## Fork and make changes to the codebase
 
-See guide for [setting up a development environment](Development.md).
+See the guide for [setting up a development environment](Development.md).
 
 
 ## Notes about licensing

package/docs/Engines.md

@@ -80,7 +80,7 @@ These engines' goal is to match (or "align") a given spoken recording with a giv
 
 ## Speech-to-translated-transcript alignment
 
-These goal here is to match (or "align") a given spoken recording in one language, with a given translated transcript in a different language, as closely as possible.
+The goal here is to match (or "align") a given spoken recording in one language, with a given translated transcript in a different language, as closely as possible.
 
 * `whisper`: given a spoken recording in any of the [98 languages](https://platform.openai.com/docs/guides/speech-to-text/supported-languages) supported by Whisper, and an English translation of its transcript, the translated transcript is tokenized and then decoded, in order, using a guided approach, with any multilingual Whisper model, set to its `translate` task mode. In this way, the approximate mapping between the spoken audio and each word of the translation is estimated
 

package/docs/Options.md

@@ -142,11 +142,11 @@ Applies to CLI operation: `transcribe`, API method: `recognize`
 
 **Whisper**:
 * `whisper.model`: selects which Whisper 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` (**Note**: large models aren't yet supported by `onnxruntime-node` due to their size). Defaults to `tiny` or `tiny.en`
-* `whisper.temperature`: temperature setting for the text decoder. Impacts 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.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 symbols. Defaults to `0.2`
-* `whisper.autoPromptParts`: use previous part's recognized text as prompt for the next part. Disabling this may help to prevent repetition carrying over between parts, in some cases. Defaults to `true`
+* `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`
 * `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.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`
@@ -370,3 +370,26 @@ These are shared between text-to-speech, speech-to-text and alignment operations
 * `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](API.md) 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 alternative a 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
+* `--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](CLI.md) for more details.

package/docs/Releases.md

@@ -14,7 +14,7 @@ For releases after `1.0.0`, see the [GitHub releases page](https://github.com/ec
 * Integrate optional vocal isolation to speech recognition, alignment and translation operations using the new `--isolate` option, allowing for higher accuracy in difficult cases like achieving word-level lyrics alignment
 * Add the new `adaptive-gate` VAD engine using a custom bandlimited adaptive gate. Fast and robust. Works well for relatively clean tracks or tracks that have already been processed using vocal isolation
 * Add optional token-level repetition suppression to Whisper engine
-* Expose several new configuration options for the Whisper engine: a settings for maximum tokens per part, and a setting to enable/disable repetition suppression, set custom random seed, disable/enable decoding of timestamp tokens
+* Expose several new configuration options for the Whisper engine: settings for maximum tokens per part, and a setting to enable/disable repetition suppression, set custom random seed, disable/enable decoding of timestamp tokens
 * Expose more options for the Elevenlabs engine
 
 **Enhancements**:
@@ -25,13 +25,13 @@ For releases after `1.0.0`, see the [GitHub releases page](https://github.com/ec
 **Behavioral and breaking changes**:
 * Minimal required node version changed to `18.0.0`
 * All recognition, alignment, translation and language recognition operations first apply the new adaptive gate VAD (can be changed to any other VAD engine via the `vad.` option prefix) and remove any sections that are not identified as containing voice, before starting processing. This should improve results in most cases, and reduce processing time
-* To reduce Whisper hallucinations and repetition loops, these change were made:
+* To reduce Whisper hallucinations and repetition loops, these changes were made:
 	* Pre-cropping by default can significantly help with reducing hallucinations, giving the model less "empty space" to hallucinate on
 	* Enable new token-level repetition suppression (`suppressRepetition = true`) during decoding
 	* Disable decoding of timestamp tokens by default (`decodeTimestampTokens = false`), since more accurate timing is already extracted via cross-attention weight alignment. For unclear reasons, this can significantly reduce the occurrence of token repetition loops, and increases word timestamp accuracy. However, there are cases where this causes the model to end a part prematurely, especially in singing and less speech-like voice segments. In those cases the it can be enabled with `decodeTimestampTokens = true`
 * When `transcribe`, `align` or `translate-speech` operations are run with `--isolate` enabled, they will output the isolated part in `some-output-file.isolated.wav` and background part (isolated subtracted from original) in `some-output-file.background.wav` (any supported codec other than `wav` can be used - this is just an example)
 * VAD operations now return a timeline including only the active sections, labeled as `active`
-* When the specified language is not English, but an `.en` Whisper model was specified, a warning would be shown, and the model would be automatically switched to the corresponding multilingual model (omitting the `.en`), instead of producing an error
+* When the specified language is not English, but an `.en` Whisper model was specified, a warning would be shown, and the model will be automatically switched to the corresponding multilingual model (omitting the `.en`), instead of producing an error
 * Default speech language detection engine is now changed to `whisper`
 * Default voice activity detection engine is now changed to `silero`
 * CLI: duplicate file name outputs would now append the `_001` suffix pattern instead of ` (1)`. This change is meant to simplify sorting and typing the resulting file names and remove the space and parenthesis characters, to ensure compatibility with all operating systems
@@ -40,7 +40,7 @@ For releases after `1.0.0`, see the [GitHub releases page](https://github.com/ec
 * `dryMixGainDb` denoising option renamed to `dryMixGain`
 
 **Fixes**:
-* Integrated `whisper` engine now uses `tiktoken` to tokenize text, which produces near identical tokenization compared to the official Python implementation. This fixes issues with decoding Chinese characters, correctly encoding prompts, and several others languages with characters that span multiple tokens
+* Integrated `whisper` engine now uses `tiktoken` to tokenize text, which produces near identical tokenization compared to the official Python implementation. This fixes issues with decoding Chinese characters, correctly encoding prompts, and several other languages with characters that span multiple tokens
 * Fix voice language lists for multilingual voices in Elevenlabs TTS engine
 * Fix runtime error with `rnnoise` when used as VAD engine
 * Fix issue in CLI where supported output media formats weren't reported correctly
@@ -132,7 +132,7 @@ Many features, enhancements, and fixes were incrementally added over the span of
 * Expose subtitle configuration options to CLI
 
 **Enhancements**:
-* Improve speech language detection to work with arbitrary length audio. Split audio to overlapping parts and detect each part individually. Then average the results.
+* Improve speech language detection to work with arbitrary length audio. Split audio into overlapping parts and detect each part individually. Then average the results.
 * Add more heteronyms
 
 **Behavioral changes**:
@@ -172,7 +172,7 @@ Many features, enhancements, and fixes were incrementally added over the span of
 * Add support for custom lexicons. Change lexicon object structure to include language code and allow for multiple languages in a single lexicon
 * Add support for SSML inputs (currently only supported by Google, Microsoft and Amazon cloud engines). Ensure they are not split to segments or sentences
 * CLI: Add flag to enable or disable file overwriting in CLI
-* CLI: Add command to list engines
+* CLI: Add operation to list engines
 * Add arguments to customize paragraph parsing
 * Change whitespace option to include option for collapsing all whitespace
 
@@ -251,8 +251,8 @@ Many features, enhancements, and fixes were incrementally added over the span of
 **New features**:
 
 **Enhancements**:
-* Show current sentence and segment in synthesis log.
-* Add check for cancellation flag.
+* Show current sentence and segment in synthesis log
+* Add check for cancellation flag
 * Add support to additional VITS voices
 * Improve auto TTS engine selection
 * Many Enhancements

package/docs/Tasklist.md

@@ -4,7 +4,7 @@
 
 ### Alignment
 
-* In DTW-RA, recognition transcript including something like "Question 2.What does Juan", where "2.What" has a point in the middle, is breaking playback of the timeline
+* In DTW-RA, a recognized transcript including something like "Question 2.What does Juan", where "2.What" has a point in the middle, is breaking playback of the timeline
 * DTW-RA will not work correctly with Polish language texts, due to issues with the eSpeak engine pronouncing `|` characters, which are intended to be used as separators and ignored by all other eSpeak languages
 
 ### Synthesis
@@ -35,7 +35,7 @@
 ## Features and enhancements
 
 ### CLI
-* Show names of files written do disk. This is useful for cases where a file is auto-renamed to prevent overwriting existing data
+* Show names of files written to disk. This is useful for cases where a file is auto-renamed to prevent overwriting existing data
 * Restrict input media file extensions to ensure that invalid files are not passed to FFmpeg
 * Consider what to do with non-supported templates like `[hello]`
 * Show a message when a new version is available
@@ -67,7 +67,7 @@
 * Use the Wikipedia reader when the URL is detected to be from `wikipedia.org`
 
 ### CLI / `list-voices`
-* When given a configuration file, see if you can fall back to take options from `speak` options, for example, to take API keys that are required for the both the synthesis request and voice list request and
+* When given a configuration file, see if you can fall back to take options from `speak` options, for example, to take API keys that are required for both the synthesis request and voice list request and
 
 ### CLI / `list-packages`
 * Support filters
@@ -83,7 +83,6 @@
 * `speak-youtube-subtitles`: To speak the subtitles of a YouTube video
 
 ### API
-* Option to control logging verbosity
 * Accept full language names as language identifiers
 * Validate timelines to ensure timestamps are always increasing: no negative timestamps or timestamps over the duration of the audio. No sentences without words, etc. and correct if needed
 * See whether it's possible to detect and include / remove Emoji characters in timelines
@@ -91,7 +90,7 @@
 * Accept voice list caching options in `SynthesisOptions`
 
 ### Package manager
-* Better error message when package is not found remotely. Currently, it just gives a `404 not found` without any other information
+* Better error message when a package is not found remotely. Currently, it just gives a `404 not found` without any other information
 * Retry on network failure
 
 ### Speech language detection
@@ -100,8 +99,7 @@
 * Deploy and add the new n-gram based text language detection model
 
 ### Segmentation
-* Split long words
-* See if it's possible to reliably use eSpeak as a segmentation engine
+* See if it's possible or useful to reliably use eSpeak as a segmentation engine.
 
 ### Subtitles
 * Split long words if needed. This is especially important for Chinese and Japanese
@@ -115,8 +113,8 @@
 * Option to disable alignment (only for some engines). Alternative: use a low granularity DTW setting that is very fast to compute
 * Find places to add commas (",") to improve speech fluency. VITS voices don't normally add speech breaks if there is no punctuation
 * An isolated dash " - " can be converted to a " , " to ensure there's a break in the speech
-* Ensure abbreviations like "Ph.d" or names like are segmented and read correctly (does `cldr` treat it as a word? Maybe eSpeak doesn't recognize it as a word). "C#" and ".NET" as well
-* Find way to manually reset voice list cache
+* Ensure abbreviations like "Ph.d" or similar names are segmented and read correctly (does `cldr` treat it as a word? Maybe eSpeak doesn't recognize it as a word). "C#" and ".NET" as well
+* Find a way to manually reset voice list cache
 * When synthesized text isn't pre-split to sentences, apply sentence splits by using the existing method to convert the output of word timelines to sentence/segment timelines
 * Some `sapi` voices and `msspeech` languages output phones that are converted to Microsoft alphabet, not IPA symbols. Try to see if these can be translated to IPA
 * Decide whether asterisk `*` should be spoken when using `speak-url` or `speak-wikipedia`
@@ -155,11 +153,11 @@
 * Show alternatives when playing in the CLI. Clear current line and rewrite already printed text for alternatives during the speech recognition process
 
 ### Recognition / Whisper
-* Whisper's Chinese and Japanese output can be split to words in a more accurate way. Consider using a dedicated segmentation library to perform the segmentation in character sequences that have no spaces within them
+* Whisper's Chinese and Japanese output can be split into words in a more accurate way. Consider using a dedicated segmentation library to perform the segmentation in character sequences that have no spaces within them
 * Automatically disable using previous section recognized transcript as prompt for the next section when lots of repetition occurred in previous section
-* Cache last model (if enough memory available)
-* The segment output can be used to split to segments, otherwise it is possible to try to guess using pause lengths or voice activity detection
-* Bring back option to use eSpeak DTW based alignment on segments, as an alternative approach
+* Cache last model (if enough memory is available)
+* The segment output can be used to split into segments, otherwise it is possible to try to guess using pause lengths or voice activity detection
+* Bring back the option to use eSpeak DTW based alignment on segments, as an alternative approach
 
 ### Alignment
 
@@ -170,7 +168,7 @@
 ### Alignment / Whisper
 
 ### Source separation / MDX-NET
-* Since MDX-NET requires FFT with large window sizes, the FFT computation overhead currently acts as a bottleneck, especially when GPU is used for inference. Currently it uses a WASM port of KissFFT, running on a single thread, which is still relatively fast. To get higher performance, try to (optionally) use native, SIMD optimized FFT like FFTW3 via a NAPI addon, with multi-threading enabled
+* Since MDX-NET requires FFT with large window sizes, the FFT computation overhead currently acts as a bottleneck, especially when GPU is used for inference. Currently, it uses a WASM port of KissFFT, running on a single thread, which is still relatively fast. To get higher performance, try to (optionally) use native, SIMD optimized FFT like FFTW3 via a NAPI addon, with multi-threading enabled
 * Option to customize overlap
 * Add more models
 
@@ -225,7 +223,7 @@
 ### CLI
 * Auto-generate options file, with comments, based on default options of the API
 * Have the CLI launch a background worker (in a thread) to enable better parallelism
-* Play back result audio while synthesis or recognition is still processing in the background
+* Playback result audio while synthesis or recognition is still processing in the background
 * Auto-import and extract project Gutenberg texts (by URL or from a file)
 * `stdin` input support
 * `stdout` output support
@@ -243,10 +241,10 @@
 * Add capitalization and punctuation to recognized outputs if needed (Silero has a model for it for `en`, `de`, `ru`, `es`, but in `.pt` format only)
 
 ### Synthesis
-* Synthesize given subtitle file and try to preserve the existing timing of cues, or even align to existing speech
+* Synthesize the given subtitle file and try to preserve the existing timing of cues, or even align to existing speech
 
 ### Recognition
-* Low latency recognition mode. Make the partial transcription available as fast as possible
+* Low latency, streaming recognition mode. Make the partial transcription available as fast as possible
 * Live input / microphone recognition
 * Implement beam search for Whisper decoder
 * Implement beam search for Silero decoder
@@ -274,7 +272,7 @@
 ## Maybe?
 
 * Using a machine translation model to provide speech translation to languages other than English? How would the timing be determined?
-* Is it possible to get sentence boundaries without punctuation using NLP techniques like part of speech tagging?
+* Is it possible to get sentence boundaries without punctuation using NLP techniques like part-of-speech tagging?
 
 ## May or may not be good ideas
 
@@ -283,7 +281,7 @@
 
 ## Other ideas
 
-* Support alignment of EPUB 3 eBooks with corresponding audiobook
+* Support alignment of EPUB 3 eBooks with a corresponding audiobook
 * Voice cloning
 * Speech-to-speech voice conversion
 * Speech-to-speech translation

package/docs/Technical.md

@@ -61,9 +61,9 @@ I don't think that `stdin` and `stdout` are able to capture the type of complex,
 
 ## Why does the CLI use `--option=value` and not `--option value` syntax?
 
-I would have allowed `--option value` if I could. The reason `--option=value` syntax is used is because the options are defined using an auto-generated schema, which is different for each command. This means I don't know ahead of time which options require an extra value, like `--voice Bob`, and which don't, like `--play`, at the time the command line is parsed. This makes it difficult to intersperse switches and inputs/outputs like `echogarden speak "Hello" --play output.mp3 --voice Bob`.
+I would have allowed `--option value` if I could. The reason `--option=value` syntax is used is because the options are defined using an auto-generated schema, which is different for each operation. This means I don't know ahead of time which options require an extra value, like `--voice Bob`, and which don't, like `--play`, at the time the command line is parsed. This makes it difficult to intersperse switches and inputs/outputs like `echogarden speak "Hello" --play output.mp3 --voice Bob`.
 
-In order to support a syntax like `--option value` I will need to parse the command line separately for each command, but then it would be difficult to share common options between all of them.
+In order to support a syntax like `--option value` I will need to parse the command line separately for each operation, but then it would be difficult to share common options between all of them.
 
 Also, `option=value` is more similar to the syntax used in the configuration file, which makes it more consistent, and since the CLI accepts arbitrary numbers of free arguments, it helps prevent confusion on whether an argument is related to an option or a free one.
 

package/package.json

@@ -1,6 +1,6 @@
 {
 	"name": "echogarden",
-	"version": "1.1.0",
+	"version": "1.2.0",
 	"description": "An easy-to-use speech toolset. Includes tools for synthesis, recognition, alignment, speech translation, language detection, source separation and more.",
 	"author": "Rotem Dan",
 	"license": "GPL-3.0",
@@ -88,7 +88,7 @@
 		"moving-median": "^1.0.0",
 		"msgpack-lite": "^0.1.26",
 		"onnxruntime-node": "^1.17.3",
-		"openai": "^4.38.2",
+		"openai": "^4.38.5",
 		"sam-js": "^0.2.1",
 		"strip-ansi": "^7.1.0",
 		"tar": "^7.0.1",
@@ -123,7 +123,7 @@
 		"@types/recursive-readdir": "^2.2.4",
 		"@types/tar": "^6.1.13",
 		"@types/ws": "^8.5.10",
-		"ts-json-schema-generator": "^2.1.0",
+		"ts-json-schema-generator": "^2.1.1",
 		"typescript": "^5.4.5"
 	}
 }