- Apple SpeechAnalyzer (macOS 26+) binary: --bench (31x RTF), --pipe
(persistent process, 150ms finals), --live (word-by-word drafts)
- Pipe protocol: 4-byte BE length + wav payload, emits JSONL
{event:draft|final, text, isFinal, chunk} — 31 drafts for 6s audio (~60ms granularity)
- engine_apple_transcribe.py: ApplePipeTranscriber with
transcribe() + transcribe_with_draft_callback(), VAD + draft
queue, new flags --apple-stream (on), --apple-stream-interval,
--apple-pipe (on). Fixes PIL/transformers import crash by lazy import.
- main_v3.py: engine selector {whisper,apple}, passthrough translate
when no -es/-fr/-ar, freeflow flags same as v2
- Freeflow polish: deterministic punctuation commands (comma,
question mark, new paragraph, at sign), filler stripping,
<keep> protection, skip-clean heuristic, freeflow/qwen/legacy
prompt styles. Much better final readability vs raw Apple/Whisper.
- main_v2.py, engine_llm.py, engine_distribute.py: integrate freeflow
- bench: Apple 2.12% WER vs Whisper Small 3.74% (Inscribe), CPU
0mW ANE (measured via powermetrics), 196M EN cryptex per locale.
- Verified: 31 word-by-word drafts, 2 finals, exit 0, bench regression ok.
Freeflow still much better for final polish — Apple wins on speed
and raw accuracy, freeflow wins on readable paragraph output.
Co-authored-by: internal-model
6.8 KiB
Python Whisper Live Transcription & Translation
A real-time, low-latency audio transcription and translation tool utilizing OpenAI's Whisper (via mlx-whisper for Apple Silicon optimization), Silero VAD for speech detection, and Helsinki-NLP's Opus-MT models for translation.
Features
- Live Transcription: Real-time speech-to-text with automatic language detection.
- On-the-fly Translation: Bridge translations to English and then to target languages (Spanish, Arabic, French, etc.).
- Voice Activity Detection (VAD): Intelligent audio buffering using Silero VAD to process only actual speech.
- Apple Silicon Optimized: Uses MLX for high performance on Mac (MPS).
- Background Ingestion: Optional background thread to send JSON payloads to a remote server.
- Configurable: Command-line parameters to select languages, devices, and ingestion.
Installation
-
Clone the repository:
git clone <repository-url> cd pythonwhisper -
Install dependencies: Ensure you have Python 3.9+ and the required libraries:
pip install mlx-whisper numpy sounddevice torch requests transformers silero-vadNote: On Apple Silicon, ensure
mlxandtorchwith MPS support are correctly installed.
Usage
Run the script using python3 transcribe.py with optional flags.
LLM Prompt Testing
You can now send the line-correction or paragraph-refinement prompts directly without starting live transcription. Every LLM request is appended as JSON Lines to logs/llm_requests.jsonl by default.
- Test the line-correction prompt:
python3 main_v2.py --post-correct-model qwen3.5:0.8b --llm-test-line "so um we should probably ship it tomorrow" --llm-test-prev1 "We finished the staging deploy." --llm-test-prev2 "QA signed off this morning." - Test the paragraph-refinement prompt:
python3 main_v2.py --post-correct-model qwen3.5:0.8b --llm-test-context "We reviewed the launch checklist." --llm-test-segments "we confirmed monitoring we confirmed rollback and then talked about the release window" - Change where requests are logged:
python3 main_v2.py --llm-request-log-path tmp/my_llm_requests.jsonl --llm-test-line "example text" - Replay the most recent logged request:
python3 main_v2.py --llm-test-from-log -1 - Replay a logged request using the original model from the log entry:
python3 main_v2.py --llm-test-from-log 12 --llm-test-use-logged-model
Each log entry includes the timestamp, provider, model, mode, temperature, and full messages payload that was sent to the LLM.
Freeflow-style Local Polish
This implementation includes a Python port of the portable parts of Freeflow's polish pipeline:
- spoken punctuation commands such as
comma,question mark,new paragraph,at sign, andhashtag - filler/noise stripping before any LLM call
- protected
<keep>...</keep>symbols so the LLM does not reinterpret dictated symbols - a clean-transcript skip heuristic for low-latency local runs
- Freeflow-inspired English/minimal/Qwen line-polish prompts
Run deterministic cleanup only:
python3 main_v2.py --post-correct
Run deterministic cleanup plus a local Ollama line-polish model when needed:
python3 main_v2.py --post-correct-llm --post-correct-model qwen3.5:0.8b --post-correct-prompt-style qwen
Force every line through the LLM, even if the deterministic output already looks clean:
python3 main_v2.py --post-correct-llm --no-post-correct-skip-clean
Common Commands
- List available audio devices:
python3 transcribe.py -l - Caption system audio (speakers) using a loopback device:
python3 transcribe.py --loopback -es - Transcribe and translate to Spanish (screen only):
python3 transcribe.py -es - Enable Spanish, Arabic, and French with English bridging, and send to server:
python3 transcribe.py -es -ar -fr -en -i - Use a specific input device (e.g., index 3) and translate to Spanish:
python3 transcribe.py -d 3 -es
Arguments
-es: Enable Spanish translation.-en: Enable English (shows original if detected, or bridged if not).-ar: Enable Arabic translation.-fr: Enable French translation.-i,--ingest: Enable data transmission to the remote server.-l,--list-devices: Show available audio devices and exit.-d,--device [ID]: Input device index (bypasses selection prompt).--loopback: Automatically select a loopback device (e.g., BlackHole, Stereo Mix) to caption system audio.-q,--quantize: Use 4-bit quantized Whisper model for faster transcription (Strategy 3).-s,--stream: Enable real-time streaming transcription/draft mode (Strategy 1).-c,--context: Enable prompt caching/rolling context to help the model maintain sentence continuity across chunks.--lang [CODE]: Hardcode the source language (e.g.,en,es) to bypass automatic language detection for faster processing.--silence [MS]: Set the minimum silence duration in milliseconds to end a chunk. Defaults to 1000ms. Increase to force longer sentences before translation.--max-buffer [SEC]: Maximum buffer duration in seconds before forcing a flush (default: 20).--channels [N]: Number of input channels (default: 1).--pick-channel [0|1]: If stereo, select channel 0 (Left) or 1 (Right) to focus transcription.--filter-lang: If used with--lang, discards segments that do not match the target language.
Technical Note: Universal Translation Models
While the current implementation uses specialized, per-language models from the Helsinki-NLP Opus-MT project (e.g., opus-mt-en-es, opus-mt-en-ar), there is an alternative approach: Universal Models.
Universal Model Alternative (e.g., Meta's NLLB-200)
The current per-language model approach is highly accurate and memory-efficient if you only need 1 or 2 target languages. However, if you require support for many languages simultaneously, loading multiple specialized models can consume significant RAM/VRAM.
We have the option to switch the translation engine to a single, universal model such as NLLB-200 (No Language Left Behind):
- Model ID:
facebook/nllb-200-distilled-600M - Benefits:
- Supports over 200 languages in a single model.
- Simplified code: no need to load/manage multiple model objects.
- More efficient for complex multilingual environments.
- Trade-off: Slightly higher memory footprint for the single model compared to a single specialized model, but more efficient than 3+ specialized models.
If you wish to switch to a universal model, the transcribe.py logic can be updated to use a single M2M100 or NLLB pipeline instead of the current MarianMT loop.