So the rendered changelog has the (PR [...]) line aligned as a list
continuation under its bullet. Verified with both short and wrapped
entries via `towncrier build --draft`.
Introduce SonioxTTSService, a WebSocket TTS provider that streams text and
receives audio over a persistent connection, multiplexing up to 5 concurrent
streams per socket via Soniox's `stream_id`. Also updates the README service
table and the Soniox voice example to use the new TTS end-to-end.
Replaces the hardcoded camera publishing send settings in
DailyTransport with a new DailyParams.camera_out_send_settings dict that
applications can pass through verbatim to the Daily client. This makes
the encoding/codec/bitrate configuration user-controllable instead of
being driven solely by the generic TransportParams fields.
As a consequence, TransportParams.video_out_bitrate is deprecated for
the Daily transport (now configured via camera_out_send_settings) and
its default is changed to None.
Adds a dedicated screen video track alongside the existing camera track
so applications can publish to Daily's built-in "screenVideo" destination
via video_out_destinations. The track is created at join time and wired
into the client settings (inputs and publishing) when "screenVideo" is
configured; write_video_frame routes frames to the appropriate track
based on the frame's transport_destination.
Bound methods are created fresh on each attribute access, so
'self._missing_function_call_handler is self._missing_function_call_handler'
is always False. Using 'is' meant the placeholder branch never fired and
both warnings logged when a function was missing at queue time.
Switch to == so equality compares the underlying function and instance.
Strengthen the missing-at-queue-time test to assert the second warning
does not fire.
Address review feedback: a function may be unregistered between when
run_function_calls queues it and when _run_function_call executes it.
Restore the live lookup, falling back to the missing-function handler
when the entry is gone, so the call still terminates with a normal
tool result. Factor the missing-handler item construction into a
helper since it's now built in two places.
Runner-created Daily rooms previously had no expiration when callers
posted partial `dailyRoomProperties` (e.g. `{"start_video_off": true}`).
The model-default `exp=None` and `eject_at_room_exp=False` meant Daily's
cron never cleaned them up, so rooms accumulated indefinitely.
Encode the policy in the runner: define `PIPECAT_CLOUD_ROOM_EXP_HOURS=4.0`,
inject `exp` and `eject_at_room_exp=True` into user-supplied properties via
`setdefault` (so explicit caller values still win), and pass
`room_exp_duration` to all four `configure()` call sites.
* VIVA SDK TT v3 support
* Format fix.
* Renamed the API naming, removed '3' from the name.
* Implementation of User turn start strategy using Krisp VIVA Interruption Prediction in scope of TT v3 support.
* TT demo tool
* Some improvements for demo scripts, audio recordin, etc.
* Enhance demo scripts with VAD selection and audio embedding features. Updated HTML report to include annotated audio players and improved response time metrics in summary formatting. Added README for setup and usage instructions.
* Refactor interrupt prediction demo to compare multiple interruption strategies (Krisp IP vs VAD). Updated README with usage instructions and output details. Enhanced audio processing with new helper functions for generating beeps and mixing audio.
* Refactor demo scripts to improve latency metrics by introducing total_delay property in TurnEvent. Update formatting in reports and visualizations to reflect accurate speech end times, including VAD wait times. Enhance HTML report with detailed latency information and adjust audio processing to account for VAD stop seconds.
* Add audio resampling functionality and update demo scripts for improved audio processing
- Introduced `resample_audio` function to handle audio resampling with linear interpolation.
- Updated `demo_audio_recorder.py` to utilize the new resampling feature, ensuring audio is saved at the requested sample rate.
- Modified `demo_interrupt_prediction.py` and `demo_turn_taking.py` to resample audio to 16 kHz for compatibility with Silero VAD.
- Adjusted imports in demo scripts to include the new resampling function.
- Enhanced error handling for sample rate discrepancies in audio recording.
* Enhance demo_interrupt_prediction.py with VAD type selection and improved processing logic
- Added support for selecting between "silero" and "krisp" VAD engines in the demo script.
- Introduced a new create_vad function to configure VAD analyzers based on the selected type.
- Updated audio processing logic to handle VAD type-specific resampling and state management.
- Modified the KrispVivaIPUserTurnStartStrategy to utilize a separate vad_flag for per-frame VAD input, improving interruption detection accuracy.
* Refactor audio processing scripts for improved readability and consistency
- Updated type hinting in `resample_audio` function to use `tuple` instead of `Tuple`.
- Simplified print statements in `demo_audio_recorder.py`, `demo_formatting.py`, and `demo_interrupt_prediction.py` for better readability.
- Adjusted argument formatting in `demo_audio_recorder.py` and `demo_formatting.py` for consistency.
- Cleaned up list comprehensions in `demo_formatting.py`, `demo_html_report.py`, and `demo_interrupt_prediction.py` for clarity.
- Enhanced error handling in `__init__.py` for the KrispVivaIPUserTurnStartStrategy import.
* Refactor VAD handling in KrispVivaIPUserTurnStartStrategy and update tests for clarity
- Simplified the argument formatting in the _handle_vad_started method for improved readability.
- Updated test assertions to reflect changes in VAD processing logic, ensuring that the vad_flag is correctly set to False during continuous state processing.
- Enhanced test cases to verify that the process method is called appropriately under different conditions.
* more format fixes.
* removed demo scripts.
* reverted wrongly removed file.
* Corrected the IP integration logic.
* style fix.
* Refactor audio processing and state management in KrispVivaIPUserTurnStartStrategy
- Removed the unused _vad_flag attribute to streamline state tracking.
- Updated the reset method to clear the audio buffer instead of resetting the vad_flag.
- Adjusted the process_frame method to use _speech_active for VAD input, enhancing clarity in the logic.
- Modified tests to reflect changes in state management and ensure proper functionality of the reset method and audio buffer handling.
* FIxed formatting
---------
Co-authored-by: Aram Poghosyan <apoghosyan@krisp.ai>
Pipecat 1.0.8 hard-required protobuf 6.x via the base `protobuf>=6.31.1,<7`
pin, blocking users whose dependency graph already constrains protobuf to
the 5.x line. The original bump (PR #4136) was only needed because
`nvidia-riva-client>=2.25.1` ships gencode compiled with protoc 6.31.1.
Changes:
- Widen base pin to `protobuf>=5.29.6,<7`.
- Regenerate `frames_pb2.py` with `grpcio-tools~=1.67.1` (protoc 5.x). Per
Google's cross-version runtime guarantee, 5.x gencode runs on both 5.x
and 6.x runtimes, so this single artifact serves all users.
- Loosen the dev pin `grpcio-tools` to `>=1.67.1,<2` so contributors can
install `pipecat[dev,nvidia]` without resolver conflict. Comment in
`frames.proto` documents the 1.67.x requirement for regeneration.
- Add an explicit `protobuf>=6.31.1,<7` to the `nvidia` extra. This
compensates for nvidia-riva-client's missing `protobuf` install
requirement (upstream packaging gap, see
https://github.com/nvidia-riva/python-clients/issues/172). When that
issue is resolved, the explicit protobuf entry in the `nvidia` extra
can be removed.
Verified: pipecat imports cleanly on both protobuf 5.29.6 and 6.33.6;
`tests/test_protobuf_serializer.py` passes; `import riva.client` succeeds
when `pipecat[nvidia]` is installed.
Nova Sonic sessions have an AWS-imposed ~8-minute time limit. This adds
transparent session continuation that rotates sessions in the background
before the limit is reached, preserving conversation context with no
user-perceptible interruption.
Implementation follows the AWS reference architecture:
- Monitor loop detects when session age exceeds threshold
- On assistant AUDIO contentStart: start buffering user audio, create next
session (sessionStart + promptStart + system instruction)
- Track SPECULATIVE/FINAL text counts as completion signal
- On completion signal: send conversation history + audioInputStart +
buffered audio to next session, then promote immediately
- Close old session in background (non-blocking)
- Dead session detection: recreate next session if idle >30s
Key design decisions:
- Session continuation enabled by default (fundamental for long conversations)
- Conversation history tracked in real-time via _sc_conversation_history
(independent of pipeline context aggregator which updates asynchronously)
- Completion signal check in _handle_content_end_event (after history update)
to ensure latest text is included in handoff
- Rolling audio buffer (default 3s) captures user audio during transition
- transition_threshold_seconds capped at 420s (7min) for safety margin
- Unified event methods (_send_text_event, _send_client_event, etc.) accept
optional stream/prompt_name params, eliminating duplicate SC methods
Also adds:
- SessionContinuationParams config (enabled, threshold, buffer, timeout)
