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1e1160906efe998845b43b02c5553a0986b1efed
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4111 Commits

Author SHA1 Message Date
Mark Backman
1e1160906e Update on_user_idle to on_user_turn_idle 2026-01-17 07:04:27 -05:00
Mark Backman
473d39791b Merge pull request #3482 from pipecat-ai/mb/user-idle-in-user-aggregator 
Add UserIdleController, deprecate UserIdleProcessor
 2026-01-16 18:47:10 -05:00
Aleix Conchillo Flaqué
4fb4c26f55 Merge pull request #3484 from amichyrpi/main 
Remove async_mode parameter from Mem0 storage
 2026-01-16 15:44:52 -08:00
Mark Backman
2e8e574ea5 Add UserIdleController, deprecate UserIdleProcessor 2026-01-16 18:44:19 -05:00
Amory Hen
a6e7c99d55 Remove async_mode parameter from Mem0 storage 2026-01-17 00:26:38 +01:00
Aleix Conchillo Flaqué
ac3fa7f91f BaseOuputTransport: minor cleanup 2026-01-16 15:15:49 -08:00
Aleix Conchillo Flaqué
6eadad53b2 BaseInputTransport: throttle UserSpeakingFrame 2026-01-16 15:15:49 -08:00
kompfner
b11150f31f Merge pull request #3480 from pipecat-ai/pk/fix-grok-realtime-smallwebrtc 
Fix an issue where Grok Realtime would error out when running with Sm…
 2026-01-16 15:46:27 -05:00
Paul Kompfner
836cf60611 Fix an issue where Grok Realtime would error out when running with SmallWebRTC transport. 
The underlying issue was related to the fact that we were sending audio to Grok before we had configured the Grok session with our default input sample rate (16000), so Grok was interpreting those initial audio chunks as having its default sample rate (24000). We didn't see this issue when using the Daily transport simply because in our test environments Daily took a smidge longer than a reflexive (localhost) pure WebRTC connection, so we would only send audio to Grok *after* we had configured the Grok session with the desired sample rate.
 2026-01-16 15:41:33 -05:00
Mark Backman
32c775311d Merge pull request #3471 from pipecat-ai/mb/fix-pydantic-2.12-docs 
Revert pydantic 2.12 extra type annotation
 2026-01-16 14:57:24 -05:00
Aleix Conchillo Flaqué
c2a0735975 MinWordsUserTurnStartStrategy: don't aggregate transcriptions 
If we aggregate transcriptions we will get incorrect interruptions. For example,
if we have a strategy with min_words=3 and we say "One" and pause, then "Two"
and pause and then "Three", this would trigger the start of the turn when it
shouldn't. We should only look at the incoming transcription text and don't
aggregate it with the previous.
 2026-01-16 11:16:06 -08:00
Aleix Conchillo Flaqué
c7ab87b0cc turns: move mute to user_mute 2026-01-16 11:07:20 -08:00
kompfner
19fb3eed9f Merge pull request #3466 from pipecat-ai/pk/fix-aws-nova-sonic-rtvi-bot-output 
Fix realtime (speech-to-speech) services' RTVI event compatibility
 2026-01-16 09:56:13 -05:00
Mark Backman
b292b32374 Merge pull request #3461 from glennpow/glenn/websocket-headers 
Allow WebsocketClientTransport to send custom headers
 2026-01-15 20:26:36 -05:00
Glenn Powell
37914cb062 Removed import and added changelog entry. 2026-01-15 16:47:15 -08:00
Mark Backman
ec40696854 Revert pydantic 2.12 extra type annotation 2026-01-15 19:16:15 -05:00
Mark Backman
64a1ad2649 Merge pull request #3470 from pipecat-ai/mb/fix-docs-0.0.99 
Docs fixes after 0.0.99
 2026-01-15 17:34:44 -05:00
Mark Backman
21aaa48e62 Fix pydantic issues impacting autodoc 2026-01-15 17:29:47 -05:00
Mark Backman
60216048a8 Docs fixes after 0.0.99 2026-01-15 16:40:42 -05:00
Mark Backman
f3c2e29fb4 Clean up CambTTSService 2026-01-15 15:59:17 -05:00
Paul Kompfner
5de80a60d4 Fix "bot-llm-text" not firing when using Grok Realtime 2026-01-15 15:30:00 -05:00
Paul Kompfner
5753762350 Fix "bot-llm-text" not firing when using OpenAI Realtime 2026-01-15 15:16:08 -05:00
Paul Kompfner
885b318b04 Fix "bot-llm-text" not firing when using Gemini Live 2026-01-15 15:03:45 -05:00
Paul Kompfner
7a22d58cf4 Fix "bot-llm-text" not firing when using AWS Nova Sonic 2026-01-15 14:56:50 -05:00
Neil Ruaro
f60eeaa212 reverted uv.lock, updated readthedocs.yaml, copyright year updates 2026-01-16 02:50:18 +08:00
Neil Ruaro
80604ba7b6 remove _update_settings method 2026-01-16 02:00:48 +08:00
Glenn Powell
0e3532c529 Allow WebsocketClientTransport to send custom headers 2026-01-15 09:31:48 -08:00
Neil Ruaro
9942fcfeb2 updated per PR reviews 2026-01-16 01:20:17 +08:00
Neil Ruaro
003c24ca6e Make model parameter explicit in docstring example 2026-01-16 01:18:37 +08:00
Neil Ruaro
ed120d014d Add model-specific sample rates, transport example, and fix audio buffer alignment 2026-01-16 01:18:37 +08:00
Neil Ruaro
e76a3d04f0 Update Camb TTS to 48kHz sample rate 2026-01-16 01:18:37 +08:00
Neil Ruaro
641d17007f Clean up Camb TTS service and tests 2026-01-16 01:18:37 +08:00
Neil Ruaro
9293b5f24a Migrate Camb TTS service from raw HTTP to official SDK 
- Replace aiohttp with camb SDK (AsyncCambAI client)
- Add support for passing existing SDK client instance
- Simplify API: no longer requires aiohttp_session parameter
- Update example to use simplified initialization
- Rewrite tests to mock SDK client instead of HTTP servers
 2026-01-16 01:18:37 +08:00
Neil Ruaro
c1f3cbd1d4 Yield TTSAudioRawFrame directly instead of calling private method 2026-01-16 01:18:37 +08:00
Neil Ruaro
78fa2ab65e Update default voice ID, fix MARS naming, and clean up example 2026-01-16 01:18:37 +08:00
Neil Ruaro
56da2caeed Update Camb.ai TTS inference options 2026-01-16 01:18:37 +08:00
Neil Ruaro
a541d65255 Update MARS model names to mars-flash, mars-pro, mars-instruct 
Rename model identifiers from mars-8-* to the new naming convention:
- mars-8-flash -> mars-flash (default)
- mars-8 -> removed
- mars-8-instruct -> mars-instruct
- Added mars-pro
 2026-01-16 01:18:37 +08:00
Neil Ruaro
7ae0d651d6 added cambai tts integration 2026-01-16 01:18:36 +08:00
kompfner
24082b84f2 Merge pull request #3453 from pipecat-ai/pk/consistency-pass-on-user-started-stopped-speaking-frames 
Do a consistency pass on how we're sending `UserStartedSpeakingFrame`…
 2026-01-15 09:24:14 -05:00
Aleix Conchillo Flaqué
9e705ce768 UserTurnController: reset user turn start strategies when turn triggered 2026-01-14 18:20:29 -08:00
Mark Backman
f3993f1775 fix to make on_user_turn_stop_timeout work with ExternalUserTurnStrategies 2026-01-14 20:10:56 -05:00
Paul Kompfner
e107902b14 Do a consistency pass on how we're sending UserStartedSpeakingFrames and UserStoppedSpeakingFrames. The codebase is now consistent in broadcasting both types of frames up and downstream. 2026-01-14 18:47:15 -05:00
Ashot
c4ae4025f3 Adjustments of Async TTS for multicontext websocket support 2026-01-14 16:33:30 +04:00
Ashot
15067c678d adapt Async TTS to updated AudioContextTTSService 2026-01-14 15:45:27 +04:00
Ashot
5ae592f38e Improve Async TTS interruption handling by using AudioContextTTSService class and add changelog fragments 2026-01-14 15:45:27 +04:00
Ashot
9cdbc56be3 Fix TTFB metric and add multi-context WebSocket support for Async TTS 2026-01-14 15:45:27 +04:00
Aleix Conchillo Flaqué
0d6bdbee10 LLMContextAggregatorPair: make strategy logs less verbose 2026-01-13 15:11:22 -08:00
Aleix Conchillo Flaqué
248dac3a9d Merge pull request #3420 from pipecat-ai/pk/fix-gemini-3-parallel-function-calls 
Fix parallel function calling with Gemini 3.
 2026-01-13 14:40:33 -08:00
Paul Kompfner
be49a54856 Fast-exit in the fix for parallel function calling with Gemini 3, if we can determine up-front that there's no work to do 2026-01-13 17:32:20 -05:00
Aleix Conchillo Flaqué
bd9ee0d646 Merge pull request #3434 from pipecat-ai/aleix/context-appregator-pair-tuple 
context aggregator pair tuple
 2026-01-13 14:12:51 -08:00