Releases: speechmatics/speechmatics-python-sdk
rt/v1.0.0
Breaking Change
Add an extra optional attribute** to the ForceEndOfUtterance message to help the engine know the real-world time marker for when the forced end of utterance should be considered. It should only be used with transcribers that support this extended message.
{"message": "ForceEndOfUtterance", "timestamp": 1.23}
Using the client.audio_seconds_sent property will return the number of seconds of audio that have been sent to the transcriber, calculated using encoding format and sample rate of the session's audio (AudioFormat). This can then be used when calling the force end of utterance:
client.force_end_of_utterance(timestamp=client.audio_seconds_sent)
Transcribers that don't support the new timestamp parameter will return an error:
Unable to validate message against the API JSON schema: ["<ValidationError: '.: {'message': 'ForceEndOfUtterance', 'timestamp': 1.87} is not valid under any of the given schemas'>"]
It can be disabled by passing in the function timestamp=None as such:
client.force_end_of_utterance(timestamp=None)
New Contributors
- @giorgosHadji made their first contribution in #95
batch/v0.4.7
What's Changed
- add support for requesting parallel engines in a http batch job by @giorgosHadji in #95
- add user_id as input to header for http batch V2 API by @giorgosHadji in #97
Usage with:
from speechmatics.batch import AsyncClient
async def main():
async with AsyncClient(url='http://localhost:PORT/v2') as client:
result = await client.transcribe(audio_file_path, parallel_engines=2, user_id="MY_USER_ID")
print(result)
asyncio.run(main())
which will flow into the headers being sent as part of the POST request to the server.
These features are available for use for on Prem scenarios only, given the transcriber version supports them. If it doesn't support them, they will be ignored.
The need for user_id is specific for the speaker_id feature as its a requirement to be able to use it - https://docs.speechmatics.com/speech-to-text/features/speaker-identification
New Contributors
- @giorgosHadji made their first contribution in #95
batch/v0.4.6
What's Changed
- Add transcript filtering config and fix output config deserialization by @lgavincrl in #88
- Fix import TranscriptFilteringConfig in batch sdk by @dumitrugutu in #91
batch/v0.4.5
What's Changed
- Add OutputConfig option in JobConfig for batch sdk by @dumitrugutu in #80
voice/v0.2.8
What's Changed
- Update Scribe Preset Configuration by @LArmstrongDev in #77
- Fix for short utterances when using ForceEndOfUtterance by @sam-s10s in #78
voice/v0.2.7
What's Changed
- fix to use rt 0.5.3 by @sam-s10s in #69
- Fix for when diarization is not enabled by @sam-s10s in #71
- Reduce RT logging in Voice SDK by @sam-s10s in #72
- Adds comprehensive README documentation by @Zultran in #70
- Updated integration examples. by @sam-s10s in #73
- Fix to max delay mode and filter for final changes by @sam-s10s in #74
- Voice SDK url parameter handling by @sam-s10s in #76
New Contributors
Full Changelog: voice/v0.2.3...voice/v0.2.7
voice/v0.2.3
What's Changed
- support for RT client 0.5.3 @sam-s10s
rt/v0.5.3
voice/v0.2.2
What's Changed
- updated Pydantic models @sam-s10s
- improved presets @sam-s10s
- updated to Smart Turn 3.1 @sam-s10s
- improved README @LArmstrongDev
Breaking Changes
- use of
smart_turnfor the mode now deprecated (useadaptivewith smart turn options)
rt/v0.5.2
What's Changed
- Add support for custom ssl context in rt sdk by @dumitrugutu in #66
- Made the api key optional with StaticKeyAuth in rt sdk by @dumitrugutu in #66