RealTimeSTT 1.1.1__tar.gz → 1.2.0__tar.gz

realtimestt-1.2.0/LICENSE

@@ -0,0 +1,202 @@
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{realtimestt-1.1.1 → realtimestt-1.2.0}/MANIFEST.in

@@ -2,6 +2,7 @@ include requirements.txt
 include README.md
 include RELEASE_NOTES.md
 include LICENSE
+include NOTICE
 prune docs/development
 prune dev-log
 prune dev-logs

realtimestt-1.1.1/LICENSE → realtimestt-1.2.0/NOTICE

@@ -1,3 +1,23 @@
+RealtimeSTT
+Copyright 2023-2026 Kolja Beigel and the RealtimeSTT contributors
+
+This product is licensed under the Apache License, Version 2.0 (see LICENSE).
+
+Third-party components
+----------------------
+The Apache-2.0 license covers the RealtimeSTT source code only. Optional
+transcription engines, voice activity detectors, wake-word engines, model
+weights and hosted services used through RealtimeSTT are not part of this
+license grant. They remain under their own licenses and terms, which you must
+review and comply with yourself. See docs/licenses.md for an overview.
+
+Earlier MIT-licensed versions
+-----------------------------
+RealtimeSTT releases up to and including 1.1.2 were published under the MIT
+License, and those releases remain available under it. Portions of this
+software originate from those releases and from contributions made under the
+MIT License; for those portions the following notice is retained:
+
 MIT License
 
 Copyright (c) 2023 Kolja Beigel

{realtimestt-1.1.1/realtimestt.egg-info → realtimestt-1.2.0}/PKG-INFO

@@ -1,15 +1,16 @@
 Metadata-Version: 2.4
 Name: realtimestt
-Version: 1.1.1
+Version: 1.2.0
 Summary: A fast Voice Activity Detection and Transcription System
 Home-page: https://github.com/KoljaB/RealtimeSTT
 Author: Kolja Beigel
 Author-email: kolja.beigel@web.de
-License: MIT
+License: Apache-2.0
 Keywords: real-time,audio,transcription,speech-to-text,voice-activity-detection,VAD,real-time-transcription,ambient-noise-detection,microphone-input,faster_whisper,speech-recognition,voice-assistants,audio-processing,buffered-transcription,pyaudio,ambient-noise-level,voice-deactivity
 Requires-Python: >=3.11,<3.13
 Description-Content-Type: text/markdown
 License-File: LICENSE
+License-File: NOTICE
 Requires-Dist: PyAudio==0.2.14
 Requires-Dist: webrtcvad-wheels==2.0.14
 Requires-Dist: halo==0.0.31
@@ -20,6 +21,12 @@ Requires-Dist: scipy==1.17.1
 Requires-Dist: websockets==16.0
 Requires-Dist: websocket-client==1.9.0
 Requires-Dist: soundfile==0.13.1
+Provides-Extra: parakeet-onnx
+Requires-Dist: onnx-asr==0.12.0; extra == "parakeet-onnx"
+Requires-Dist: onnxruntime==1.30.0; extra == "parakeet-onnx"
+Provides-Extra: nemotron-ort
+Requires-Dist: onnxruntime==1.23.2; extra == "nemotron-ort"
+Requires-Dist: kaldi-native-fbank==1.22.4; extra == "nemotron-ort"
 Provides-Extra: minimal
 Provides-Extra: faster-whisper
 Requires-Dist: faster-whisper==1.2.1; extra == "faster-whisper"
@@ -134,6 +141,8 @@ Requires-Dist: uvicorn[standard]<1,>=0.30; extra == "all"
 Requires-Dist: silero-vad[onnx-cpu]>=6.2.1; python_version >= "3.8" and extra == "all"
 Requires-Dist: transformers; extra == "all"
 Requires-Dist: nemo_toolkit[asr]; extra == "all"
+Requires-Dist: onnx-asr==0.12.0; extra == "all"
+Requires-Dist: onnxruntime==1.30.0; extra == "all"
 Requires-Dist: torch==2.8.0; (python_version >= "3.10" and python_version < "3.12" and platform_system != "Windows") and extra == "all"
 Requires-Dist: torchaudio==2.8.0; (python_version >= "3.10" and python_version < "3.12" and platform_system != "Windows") and extra == "all"
 Requires-Dist: omnilingual-asr>=0.2.0; (python_version >= "3.10" and python_version < "3.12" and platform_system != "Windows") and extra == "all"
@@ -255,6 +264,10 @@ model after the builder finishes:
 
 # RealtimeSTT
 
+> **License:** RealtimeSTT's own code is Apache-2.0. Engines, models, VADs and
+> wake-word backends keep their own licenses, which you must check yourself:
+> [license notes](docs/licenses.md).
+
 RealtimeSTT is a Python speech-to-text library for applications that need
 voice activity detection, fast transcription, optional realtime text updates,
 wake words, and direct access to audio streams. It is designed for assistants,
@@ -265,6 +278,59 @@ The general-purpose default path uses `faster_whisper`. Other engines are
 available through install extras when their optional dependencies and models
 are present.
 
+## New in 1.2.0: fast pause transcripts for voice assistants
+
+A voice assistant has to decide, every time the user pauses, whether the turn
+is over and what was said. The sooner it has a reliable transcript of the
+words so far, the sooner it can judge the turn, start the language model, and
+answer. RealtimeSTT 1.2.0 makes that transcript (the **Preview**) much faster
+on the production server:
+
+- **Live-flush Preview** (`--preview-live-flush`, new `nemotron_ort` engine).
+  At a pause the streaming model decodes the audio it has not yet finalized on
+  a copy of its state, so the live stream continues undisturbed. The Preview
+  arrives in tens of milliseconds instead of waiting for a full re-decode. The
+  accurate full-model text follows as a separate `previewRecord` event, and an
+  optional second flush stage (`previewRefined`) adds a slightly longer read.
+- **Tail-anchored Preview** (`--preview-input-mode tail`). Instead of
+  re-decoding the whole turn at every pause, the final model decodes only the
+  newest audio and joins it to the text already recognized. The cost no longer
+  grows with the length of the turn. If the join cannot be proven, the server
+  decodes the full turn.
+- **Orukeet final model** (`stt-install-sherpa-models --model orukeet`): an
+  optional, hash-verified Parakeet TDT derivative for 25 European languages.
+
+Measured once on the Linux x86-64 production host, on a 12-second English
+utterance: live-flush Preview **48 ms**, tail-mode Preview **149 ms**, and a
+full-turn Preview **about 710 ms**, from request to event.
+
+**These fast paths are CPU only.** `nemotron_ort` runs on ONNX Runtime's CPU
+provider, and the tested profile is the CPU Nemotron/Orukeet pair on Linux
+x86-64. On GPU, `faster_whisper` remains the recommended general-purpose path
+and does not offer live flush.
+
+A voice-assistant server with all three enabled:
+
+```bash
+python -m pip install "RealtimeSTT[server,sherpa-onnx,nemotron-ort]"
+stt-install-sherpa-models --root ./models/sherpa-onnx --model nemotron
+stt-install-sherpa-models --root ./models/sherpa-onnx --model orukeet
+stt-server-production \
+  --engine sherpa_onnx_parakeet --model oruk/orukeet \
+  --realtime-engine nemotron_ort \
+  --realtime-model ./models/sherpa-onnx/sherpa-onnx-nemotron-3.5-asr-streaming-0.6b-560ms-int8-2026-06-11 \
+  --download-root ./models/sherpa-onnx --device cpu \
+  --preview-only-transcription \
+  --preview-input-mode tail --preview-live-flush
+```
+
+The client sends `{"type":"preview","previewRequestId":"pause-1","pauseTextSource":"live_flush"}`
+when it detects a pause. Protocol details, fallbacks and events are in the
+[production server guide](RealtimeSTT_server/PRODUCTION_SERVER.md#live-flush-preview---preview-live-flush).
+The `nemotron-ort` extra pins `onnxruntime==1.23.2`; do not combine it with the
+`parakeet-onnx` extra in one environment. Full change list:
+[release notes](RELEASE_NOTES.md).
+
 ## Recommended Engine Profiles
 
 - **CUDA / GPU:** Keep using the established `faster_whisper` CUDA setup. It
@@ -290,6 +356,20 @@ See the [production server guide](RealtimeSTT_server/PRODUCTION_SERVER.md) for
 the authenticated HTTP/WebSocket deployment recipe and exact pinned model
 directories.
 
+### Orukeet CPU performance: official runtime tuning
+
+**Official sherpa-ONNX 1.13.8 supports disabling CPU worker spinning through a
+provider configuration file.** On the tested Ubuntu 24.04 x86-64 / Python 3.12
+host, seven threads with spinning disabled beat our earlier custom build:
+**107.2 versus 113.3 ms** for five-second audio over 20 alternating-order rounds.
+The same Orukeet INT8 model and CPU affinity were used; transcripts matched.
+
+This configuration uses official PyPI wheels and requires no custom native build.
+Install and enable it explicitly; the standard extra and model installer do not
+automatically select a CPU performance profile. Seven threads is a measured
+setting for one host. See the
+[installation, configuration and benchmark details](docs/engines/sherpa-onnx.md#cpu-performance-and-installation).
+
 ### Support RealtimeSTT
 
 If RealtimeSTT saved you time, one GitHub star is a simple way to help make it more stable.
@@ -511,7 +591,17 @@ unit tests separate from opt-in real-model tests; see [docs/testing.md](docs/tes
 
 ## License
 
-MIT
+RealtimeSTT's own source code is licensed under the
+[Apache License 2.0](LICENSE); see also [NOTICE](NOTICE). Releases up to and
+including 1.1.2 were published under the MIT License.
+
+This license covers the RealtimeSTT code only. It does not relicense, and does
+not release you from checking and complying with, the licenses and terms of
+the transcription engines, VAD and wake-word engines, model weights or hosted
+services you use through this library. Some of them require attribution,
+share-alike, a commercial license or a provider agreement. Start with the
+[license notes](docs/licenses.md) and verify the upstream terms for the exact
+package, model and revision you ship.
 
 ## Author
 

{realtimestt-1.1.1 → realtimestt-1.2.0}/README.md

@@ -1,5 +1,9 @@
 # RealtimeSTT
 
+> **License:** RealtimeSTT's own code is Apache-2.0. Engines, models, VADs and
+> wake-word backends keep their own licenses, which you must check yourself:
+> [license notes](docs/licenses.md).
+
 RealtimeSTT is a Python speech-to-text library for applications that need
 voice activity detection, fast transcription, optional realtime text updates,
 wake words, and direct access to audio streams. It is designed for assistants,
@@ -10,6 +14,59 @@ The general-purpose default path uses `faster_whisper`. Other engines are
 available through install extras when their optional dependencies and models
 are present.
 
+## New in 1.2.0: fast pause transcripts for voice assistants
+
+A voice assistant has to decide, every time the user pauses, whether the turn
+is over and what was said. The sooner it has a reliable transcript of the
+words so far, the sooner it can judge the turn, start the language model, and
+answer. RealtimeSTT 1.2.0 makes that transcript (the **Preview**) much faster
+on the production server:
+
+- **Live-flush Preview** (`--preview-live-flush`, new `nemotron_ort` engine).
+  At a pause the streaming model decodes the audio it has not yet finalized on
+  a copy of its state, so the live stream continues undisturbed. The Preview
+  arrives in tens of milliseconds instead of waiting for a full re-decode. The
+  accurate full-model text follows as a separate `previewRecord` event, and an
+  optional second flush stage (`previewRefined`) adds a slightly longer read.
+- **Tail-anchored Preview** (`--preview-input-mode tail`). Instead of
+  re-decoding the whole turn at every pause, the final model decodes only the
+  newest audio and joins it to the text already recognized. The cost no longer
+  grows with the length of the turn. If the join cannot be proven, the server
+  decodes the full turn.
+- **Orukeet final model** (`stt-install-sherpa-models --model orukeet`): an
+  optional, hash-verified Parakeet TDT derivative for 25 European languages.
+
+Measured once on the Linux x86-64 production host, on a 12-second English
+utterance: live-flush Preview **48 ms**, tail-mode Preview **149 ms**, and a
+full-turn Preview **about 710 ms**, from request to event.
+
+**These fast paths are CPU only.** `nemotron_ort` runs on ONNX Runtime's CPU
+provider, and the tested profile is the CPU Nemotron/Orukeet pair on Linux
+x86-64. On GPU, `faster_whisper` remains the recommended general-purpose path
+and does not offer live flush.
+
+A voice-assistant server with all three enabled:
+
+```bash
+python -m pip install "RealtimeSTT[server,sherpa-onnx,nemotron-ort]"
+stt-install-sherpa-models --root ./models/sherpa-onnx --model nemotron
+stt-install-sherpa-models --root ./models/sherpa-onnx --model orukeet
+stt-server-production \
+  --engine sherpa_onnx_parakeet --model oruk/orukeet \
+  --realtime-engine nemotron_ort \
+  --realtime-model ./models/sherpa-onnx/sherpa-onnx-nemotron-3.5-asr-streaming-0.6b-560ms-int8-2026-06-11 \
+  --download-root ./models/sherpa-onnx --device cpu \
+  --preview-only-transcription \
+  --preview-input-mode tail --preview-live-flush
+```
+
+The client sends `{"type":"preview","previewRequestId":"pause-1","pauseTextSource":"live_flush"}`
+when it detects a pause. Protocol details, fallbacks and events are in the
+[production server guide](RealtimeSTT_server/PRODUCTION_SERVER.md#live-flush-preview---preview-live-flush).
+The `nemotron-ort` extra pins `onnxruntime==1.23.2`; do not combine it with the
+`parakeet-onnx` extra in one environment. Full change list:
+[release notes](RELEASE_NOTES.md).
+
 ## Recommended Engine Profiles
 
 - **CUDA / GPU:** Keep using the established `faster_whisper` CUDA setup. It
@@ -35,6 +92,20 @@ See the [production server guide](RealtimeSTT_server/PRODUCTION_SERVER.md) for
 the authenticated HTTP/WebSocket deployment recipe and exact pinned model
 directories.
 
+### Orukeet CPU performance: official runtime tuning
+
+**Official sherpa-ONNX 1.13.8 supports disabling CPU worker spinning through a
+provider configuration file.** On the tested Ubuntu 24.04 x86-64 / Python 3.12
+host, seven threads with spinning disabled beat our earlier custom build:
+**107.2 versus 113.3 ms** for five-second audio over 20 alternating-order rounds.
+The same Orukeet INT8 model and CPU affinity were used; transcripts matched.
+
+This configuration uses official PyPI wheels and requires no custom native build.
+Install and enable it explicitly; the standard extra and model installer do not
+automatically select a CPU performance profile. Seven threads is a measured
+setting for one host. See the
+[installation, configuration and benchmark details](docs/engines/sherpa-onnx.md#cpu-performance-and-installation).
+
 ### Support RealtimeSTT
 
 If RealtimeSTT saved you time, one GitHub star is a simple way to help make it more stable.
@@ -256,7 +327,17 @@ unit tests separate from opt-in real-model tests; see [docs/testing.md](docs/tes
 
 ## License
 
-MIT
+RealtimeSTT's own source code is licensed under the
+[Apache License 2.0](LICENSE); see also [NOTICE](NOTICE). Releases up to and
+including 1.1.2 were published under the MIT License.
+
+This license covers the RealtimeSTT code only. It does not relicense, and does
+not release you from checking and complying with, the licenses and terms of
+the transcription engines, VAD and wake-word engines, model weights or hosted
+services you use through this library. Some of them require attribution,
+share-alike, a commercial license or a provider agreement. Start with the
+[license notes](docs/licenses.md) and verify the upstream terms for the exact
+package, model and revision you ship.
 
 ## Author
 

{realtimestt-1.1.1 → realtimestt-1.2.0}/RELEASE_NOTES.md

@@ -1,5 +1,86 @@
 # Release Notes
 
+## 1.2.0 - 2026-11-11
+
+This minor release relicenses RealtimeSTT under Apache-2.0, adds faster
+Preview paths to the production server, adds the optional Orukeet final model
+and a `nemotron_ort` streaming engine, and includes community fixes merged
+since 1.1.2.
+
+### License
+
+- RealtimeSTT is now licensed under the Apache License 2.0 (#344). Releases up
+  to and including 1.1.2 were published under the MIT License.
+
+### Added
+
+- Production server: opt-in tail-anchored Preview decoding with
+  `--preview-input-mode tail` and `--preview-tail-overlap-seconds`. The
+  Preview decodes only the recent speech tail and splices it onto the live
+  text; when the splice cannot be proven it decodes the full turn.
+  Capabilities report the mode as `preview.decodeMode`.
+- Production server: `--preview-live-flush`. A Preview request with
+  `pauseTextSource = "live_flush"` is answered from a snapshot flush of the
+  live streaming lane; the full-model decode follows as a `previewRecord`
+  event, and an optional second flush stage publishes `previewRefined`.
+  Capabilities report `preview.liveFlush`.
+- `nemotron_ort` streaming engine (ONNX Runtime Nemotron with snapshot flush)
+  and the `nemotron-ort` extra. It pins `onnxruntime==1.23.2`; do not install
+  it in the same environment as the `parakeet-onnx` extra.
+- Optional verified Orukeet final transcription model
+  (`stt-install-sherpa-models --model orukeet`) (#328).
+- Spinner status messages are configurable on `AudioToTextRecorder` (#317,
+  hardened in #342).
+
+### Fixed
+
+- Tail merge keeps Devanagari and other Brahmic vowel marks and accepts
+  mark-only corrections backed by exact neighbouring words (#338).
+- Words re-decoded in a Preview tail keep the anchor's casing.
+- The final period is added when a transcript ends in a combining vowel sign
+  (#329).
+- `feed_audio` no longer crashes on a chunk too short to resample (#330).
+- The server starts with the client's language when the client sends an empty
+  language (#332).
+- The client can turn off `silero_deactivity_detection` (#335).
+- GPU installs require a compatible OpenWakeWord (`>=0.6.0`) (#327, #336).
+
+### Documentation and tests
+
+- Requiring a wake word for each recording is documented (#339).
+- OpenWakeWord backend aliases are covered by tests (#337).
+- Server priority and cancellation fences run in unit CI.
+
+## 1.1.2 - 2026-08-30
+
+This patch release reduces advisory early-RMS Preview latency while preserving
+the authoritative transcription retry and full-turn accuracy contract. See
+[the release-readiness checklist](docs/release-1.1.2-readiness.md) for the
+required build, deployment, attestation, and publication evidence.
+
+### Added
+
+- Production capabilities now advertise the exact `preview.earlyRms` contract:
+  request mode, 25 ms first-attempt decoder flush, one-attempt limit, and no
+  advisory empty-result retry.
+
+### Changed
+
+- Signed source/wheel/sdist parity now canonicalizes CRLF/LF line endings for
+  shipped HTML and command files; the release privacy scan inspects those text
+  assets too.
+- The exact advisory `previewMode = "early_rms"` appends 25 ms of zero PCM to
+  its first and only decode and publishes an empty result immediately.
+- Authoritative, missing, unknown, and malformed Preview modes retain the
+  existing conditional 500 ms empty-result retry unchanged.
+
+### Release boundaries
+
+- Miep decides when an early browser RMS pause is advisory; RealtimeSTT does
+  not change endpoint detection, transcript authority, or response text.
+- LLM admission-proxy cancellation is a separate deployment component and is
+  not part of this Python distribution.
+
 ## 1.1.1 - 2026-08-28
 
 This patch release fixes stale Preview ASR work delaying newer full-turn

{realtimestt-1.1.1 → realtimestt-1.2.0}/RealtimeSTT/audio_recorder.py

@@ -233,6 +233,10 @@ class AudioToTextRecorder:
                  on_merged_realtime_transcription_update=None,
                  on_realtime_transcription_merge_update=None,
                  ultrafast_realtime_max_tail_words: int = 5,
+                 spinner_listening_text: str = "speak now",
+                 spinner_wakeword_text: str = "say {wake_words}",
+                 spinner_transcribing_text: str = "transcribing",
+                 spinner_recording_text: str = "recording",
                  ):
         """
         Initializes an audio recorder and  transcription
@@ -292,6 +296,17 @@ class AudioToTextRecorder:
             feed_audio() method.
         - spinner (bool, default=True): Show spinner animation with current
             state.
+        - spinner_listening_text (str, default="speak now"): Spinner text
+            shown while listening for voice activity. Set to "" to show no
+            message.
+        - spinner_wakeword_text (str, default="say {wake_words}"): Spinner
+            text shown while waiting for wake words. ``{wake_words}`` is
+            replaced with the configured wake words. Set to "" to show no
+            message.
+        - spinner_transcribing_text (str, default="transcribing"): Spinner
+            text shown while transcribing. Set to "" to show no message.
+        - spinner_recording_text (str, default="recording"): Spinner text
+            shown while recording. Set to "" to show no message.
         - level (int, default=logging.WARNING): Logging level.
         - batch_size (int, default=16): Batch size for the main transcription
 
@@ -729,6 +744,10 @@ class AudioToTextRecorder:
             ultrafast_realtime_max_tail_words=(
                 ultrafast_realtime_max_tail_words
             ),
+            spinner_listening_text=spinner_listening_text,
+            spinner_wakeword_text=spinner_wakeword_text,
+            spinner_transcribing_text=spinner_transcribing_text,
+            spinner_recording_text=spinner_recording_text,
         )
 
         initialize_recorder(

{realtimestt-1.1.1 → realtimestt-1.2.0}/RealtimeSTT/audio_recorder_client.py

@@ -515,7 +515,7 @@ class AudioToTextRecorderClient:
         if self.debug_mode:
             args.append('--debug')  # flag, no need for True/False
             
-        if self.language:
+        if self.language is not None:
             args += ['--language', self.language]
         if self.silero_sensitivity is not None:
             args += ['--silero_sensitivity', str(self.silero_sensitivity)]
@@ -536,8 +536,9 @@ class AudioToTextRecorderClient:
                 '--deactivity_silence_confirmation_duration',
                 str(self.deactivity_silence_confirmation_duration),
             ]
-        if self.silero_deactivity_detection:
-            args.append('--silero_deactivity_detection')  # flag, no need for True/False
+        # stt-server defaults this to True, so False has to be sent explicitly too
+        args.append('--silero_deactivity_detection' if self.silero_deactivity_detection
+                     else '--no-silero_deactivity_detection')
         if self.beam_size is not None:
             args += ['--beam_size', str(self.beam_size)]
         if self.beam_size_realtime is not None:

{realtimestt-1.1.1 → realtimestt-1.2.0}/RealtimeSTT/core/initialization.py

@@ -287,6 +287,10 @@ def _assign_initial_attributes(recorder, init_args, normalize_wakeword_backend):
     recorder.warmup_vad = init_args["warmup_vad"]
     recorder.listen_start = 0
     recorder.spinner = init_args["spinner"]
+    recorder.spinner_listening_text = init_args["spinner_listening_text"]
+    recorder.spinner_wakeword_text = init_args["spinner_wakeword_text"]
+    recorder.spinner_transcribing_text = init_args["spinner_transcribing_text"]
+    recorder.spinner_recording_text = init_args["spinner_recording_text"]
     recorder.halo = None
     recorder.state = "inactive"
     recorder.wakeword_detected = False

{realtimestt-1.1.1 → realtimestt-1.2.0}/RealtimeSTT/core/manual_audio_input.py

@@ -28,6 +28,11 @@ def _numpy_audio_to_pcm16_bytes(chunk, original_sample_rate):
 
     if original_sample_rate != TARGET_SAMPLE_RATE:
         num_samples = int(len(chunk) * TARGET_SAMPLE_RATE / original_sample_rate)
+        if num_samples == 0:
+            # Empty, or shorter than one 16 kHz sample: scipy cannot resample to
+            # zero samples (ZeroDivisionError). Like the fraction int() already
+            # drops from every chunk, a sub-sample chunk yields no output.
+            return b""
         chunk = resample(chunk, num_samples)
 
     if original_is_float: