Compare commits

...

577 Commits

Author SHA1 Message Date
Aleix Conchillo Flaqué
c4a72802f0 Merge pull request #2074 from pipecat-ai/aleix/pipecat-0.0.72
update CHANGELOG for 0.0.72
2025-06-26 12:10:14 -07:00
Aleix Conchillo Flaqué
917394803c update CHANGELOG for 0.0.72 2025-06-26 11:42:52 -07:00
Mark Backman
01040ddcdd Merge pull request #2071 from pipecat-ai/mb/services-docstrings-update
Add/update docstrings to LLM services
2025-06-26 14:42:32 -04:00
Aleix Conchillo Flaqué
7947497f7e Merge pull request #2073 from a6kme/patch-1
Start HeartBeat when all processors have processed StartFrame
2025-06-26 11:34:46 -07:00
Aleix Conchillo Flaqué
539ca5856f Merge pull request #2072 from pipecat-ai/aleix/utils-watchdog-cleanup
utils(asyncio): simplify watchdog helpers
2025-06-26 11:29:21 -07:00
Abhishek
89c801f82c Start HeartBeat when all processors have processed StartFrame
Some of the processors like STTService and TTSService don't push StartFrame ahead in the pipeline, unless they have connected with their service providers. This delays StartFrame in downstream processors. 

If we receive HeartBeat frame before StartFrame, we will get AttributeError `'Processor' object has no attribute '_FrameProcessor__input_queue'`. 

Idea is to start HeartBeats after StartFrame has been processed by all the Processors in the pipeline.
2025-06-26 23:28:37 +05:30
Aleix Conchillo Flaqué
3de4f22d34 utils(asyncio): simplify watchdog helpers 2025-06-26 09:40:42 -07:00
Mark Backman
0e4d2be98c Update AzureRealtimeBetaLLMService docstrings 2025-06-26 12:12:00 -04:00
Mark Backman
d8ce108ccd Update OpenAIRealtimeBetaLLMService docstrings 2025-06-26 12:06:47 -04:00
Mark Backman
d123cd4b2b Update GeminiMultimodalLiveLLMService docstrings 2025-06-26 11:47:30 -04:00
Aleix Conchillo Flaqué
4d34aa7cd6 Merge pull request #2069 from pipecat-ai/aleix/utils-asyncio-package
move things to new utils.asyncio package
2025-06-26 08:26:47 -07:00
Aleix Conchillo Flaqué
b860e94582 move things to new utils.asyncio package 2025-06-26 08:24:25 -07:00
Aleix Conchillo Flaqué
9d653e3788 Merge pull request #2068 from pipecat-ai/aleix/task-manager-dont-warn-reset-watchdog
TaskManager: don't warn on reset_watchdog()
2025-06-26 08:23:51 -07:00
Mark Backman
9e518cf2ba Update AWSNovaSonicLLMService docstrings 2025-06-26 11:21:18 -04:00
Mark Backman
2856372ad6 Update TogetherLLMService docstrings 2025-06-26 11:01:35 -04:00
Mark Backman
efbf574613 Update SambaNovaLLMService docstrings 2025-06-26 11:00:40 -04:00
Mark Backman
c018eb2f0e Update QwenLLMService docstrings 2025-06-26 10:57:42 -04:00
Mark Backman
d7bfe54b7c Update PerplexityLLMService docstrings 2025-06-26 10:56:48 -04:00
Mark Backman
137282b7a9 Update OpenRouterLLMService docstrings 2025-06-26 10:53:42 -04:00
Mark Backman
769f8c8f34 Update OpenPipeLLMService docstrings 2025-06-26 10:53:05 -04:00
Mark Backman
8b8a37ae7c Update OLLamaLLMService docstrings 2025-06-26 10:48:19 -04:00
Mark Backman
56e2b006f5 Update NimLLMService docstrings 2025-06-26 10:47:26 -04:00
Mark Backman
79cca05e43 Update GroqLLMService docstrings 2025-06-26 10:46:07 -04:00
Mark Backman
166c8e8e82 Update GrokLLMService docstrings 2025-06-26 10:39:46 -04:00
Mark Backman
9b64d2c325 Update GoogleLLMService docstrings 2025-06-26 10:37:22 -04:00
Mark Backman
03e3e9fae9 Update FireworksLLMService docstrings 2025-06-26 10:28:35 -04:00
Mark Backman
65234ae41a Update DeepSeekLLMService docstrings 2025-06-26 10:27:36 -04:00
Mark Backman
3828df8cf9 Update CerebrasLLMService docstrings 2025-06-26 10:26:42 -04:00
Mark Backman
9cbe85bf99 Update AzureLLMService docstrings 2025-06-26 10:25:17 -04:00
Mark Backman
7bf805b829 Update AWSBedrock docstrings 2025-06-26 10:23:40 -04:00
Mark Backman
990ee436e1 Add Anthropic docstrings 2025-06-26 07:42:22 -04:00
Mark Backman
1cd42066a6 Merge pull request #2067 from pipecat-ai/mb/update-docstrings-for-ref-docs
Update base service class docstrings for better docs auto-generation
2025-06-26 07:07:59 -04:00
Filipi da Silva Fuchter
ba43558049 Merge pull request #2066 from pipecat-ai/filipi/sentry_freeze_test
Enabling watchdog and sentry into the freeze-test
2025-06-26 08:01:51 -03:00
Mark Backman
951c8d34da Add special case handling for STT, TTS, LLM 2025-06-26 00:15:09 -04:00
Mark Backman
ac61139243 Add OpenAI LLM docstrings 2025-06-26 00:06:57 -04:00
Mark Backman
5b8f1fe3e3 Add Cartesia TTS docstrings 2025-06-25 23:50:55 -04:00
Mark Backman
0aa197e4a4 Add docstrings to DeepgramSTTService 2025-06-25 23:36:04 -04:00
Mark Backman
f04e058c96 Programmatically set the copyright date in docs 2025-06-25 23:29:37 -04:00
Mark Backman
6ef2ae12b7 Mock mcp imports 2025-06-25 23:29:37 -04:00
Mark Backman
fe6bbdaefe Skip dataclass attributes to remove duplicate entries 2025-06-25 23:29:37 -04:00
Mark Backman
cc66fddca9 Modify docs auto-gen rules to remove duplicate parameters listing 2025-06-25 23:29:37 -04:00
Mark Backman
04b70ddf13 Add MCPClient docstrings 2025-06-25 22:38:11 -04:00
Mark Backman
bb3bb8d9c6 Improve WebsocketService docstrings 2025-06-25 22:38:11 -04:00
Mark Backman
f80f62c7d1 Add VisionService docstrings 2025-06-25 22:38:11 -04:00
Mark Backman
2007ae4317 Add ImageGenService docstrings 2025-06-25 22:38:11 -04:00
Mark Backman
a1e5a1eff4 Add AIService docstrings 2025-06-25 22:38:11 -04:00
Mark Backman
691999b402 Add AIServices docstring 2025-06-25 22:38:11 -04:00
Mark Backman
33f3a4cea1 Add TTSService docstrings 2025-06-25 22:38:11 -04:00
Mark Backman
ab1d2dbe6a Add STTService docstrings 2025-06-25 22:27:07 -04:00
Mark Backman
f622b281d0 Make call_start_function a private function in llm_service 2025-06-25 22:23:13 -04:00
Mark Backman
fb12bf9b4c Update LLMService docstrings 2025-06-25 22:23:13 -04:00
Aleix Conchillo Flaqué
27af50087e TaskManager: don't warn on reset_watchdog() 2025-06-25 17:29:45 -07:00
Filipi Fuchter
03502bed52 Enabling watchdog and sentry into the freeze-test 2025-06-25 20:53:30 -03:00
Aleix Conchillo Flaqué
27c7e2d150 Merge pull request #2063 from pipecat-ai/aleix/watchdog-timers-remove-start-watchdog
no need to call start_watchdog() only reset_watchdog()
2025-06-25 16:47:44 -07:00
Aleix Conchillo Flaqué
e81d387971 TaskManager: rely on add_done_callback() 2025-06-25 16:44:20 -07:00
Aleix Conchillo Flaqué
ef1ade3a71 allow enabling watchdog timers per frame processor or task 2025-06-25 16:36:19 -07:00
Aleix Conchillo Flaqué
4f032f5b96 update keepalive times depending on watchdog timers 2025-06-25 15:55:16 -07:00
Aleix Conchillo Flaqué
72cb967780 update CHANGELOG with watchdog timers updates 2025-06-25 15:55:16 -07:00
Aleix Conchillo Flaqué
357934a644 watchdog timers are disabled by default use enable_watchdog_timers 2025-06-25 15:55:16 -07:00
Aleix Conchillo Flaqué
327973657f TaskManager: remove wathcdog timer when main task is done 2025-06-25 11:26:21 -07:00
Aleix Conchillo Flaqué
d2730e6741 GooglSTTService: cleanup request queues 2025-06-25 11:12:32 -07:00
Aleix Conchillo Flaqué
eb5ecab104 no need to call start_watchdog() only reset_watchdog() 2025-06-25 11:12:32 -07:00
Mark Backman
202055a9b8 Merge pull request #2065 from pipecat-ai/mb/fix-configdict-openai-realtime
fix: add missing ConfigDict import in openai_realtime_beta/events
2025-06-25 11:40:35 -04:00
Mark Backman
7034a9e3fd fix: add missing ConfigDict import in openai_realtime_beta/events 2025-06-25 11:32:29 -04:00
Filipi da Silva Fuchter
8f7ed12262 Merge pull request #2061 from pipecat-ai/not_force_bot_speaking
Not forcing the bot resume speaking in case we receive no transcription.
2025-06-24 20:57:46 -03:00
Aleix Conchillo Flaqué
96b5320ef9 Merge pull request #2055 from pipecat-ai/aleix/fix-sentry-async
SentryMetrics: send metrics to sentry asynchronously
2025-06-24 16:32:01 -07:00
Filipi Fuchter
d5cd742237 Not forcing the bot resume speaking in case we receive no transcription. 2025-06-24 20:12:49 -03:00
Aleix Conchillo Flaqué
1f1da8942d SentryMetrics: send metrics to sentry asynchronously 2025-06-24 15:56:08 -07:00
Mark Backman
7953e1e9d9 Merge pull request #2054 from pipecat-ai/mb/telnyx-catch-hangup-error
fix: Telnyx, catch error when user has hung up the call first
2025-06-24 18:04:19 -04:00
Mark Backman
d6f7ecc0a3 fix: Telnyx, catch error when user has hung up the call first 2025-06-24 17:28:00 -04:00
Mark Backman
3eed316049 Merge pull request #2020 from snova-jorgep/snova-jorgep/sambanova-integration
Add Sambanova LLM and STT integration
2025-06-24 17:04:24 -04:00
Jorge Piedrahita Ortiz
851cf079c3 Merge branch 'main' into snova-jorgep/sambanova-integration 2025-06-24 16:00:28 -05:00
jhpiedrahitao
dfb0da32a9 fmt 2025-06-24 15:59:40 -05:00
Aleix Conchillo Flaqué
f450da57e5 Merge pull request #2056 from pipecat-ai/khk/fix-22d
Update google libraries used in google audio-in examples
2025-06-24 13:47:59 -07:00
Aleix Conchillo Flaqué
2ec6b6c995 Merge pull request #2060 from pipecat-ai/aleix/watchdog-timeout-secs
FrameProcessor: use watchdog_timeout_secs
2025-06-24 13:36:39 -07:00
Aleix Conchillo Flaqué
53b769a8ec FrameProcessor: use watchdog_timeout_secs 2025-06-24 13:33:47 -07:00
Filipi da Silva Fuchter
4f9adc173a Merge pull request #2004 from pipecat-ai/filipi/pipeline_freeze
Pipeline freeze improvements
2025-06-24 17:20:38 -03:00
Filipi Fuchter
dc4a58877e Fixing merge conflict. 2025-06-24 17:12:40 -03:00
Filipi Fuchter
a6243a6fe7 Merge branch 'main' into filipi/pipeline_freeze
# Conflicts:
#	CHANGELOG.md
#	src/pipecat/pipeline/task.py
#	src/pipecat/processors/frame_processor.py
#	src/pipecat/transports/base_input.py
2025-06-24 17:11:21 -03:00
Aleix Conchillo Flaqué
cf5f1b541a Merge pull request #2049 from pipecat-ai/aleix/introduce-watchdog-timers
introduce watchdog timers
2025-06-24 13:00:57 -07:00
Filipi Fuchter
70e6c48233 Mentioning the fixes in the changelog. 2025-06-24 16:56:46 -03:00
Filipi Fuchter
51f7d14d0a Merge branch 'main' into filipi/pipeline_freeze 2025-06-24 16:44:07 -03:00
Filipi Fuchter
4853d5d1fc Handling the case where user stopped speaking but no new aggregation received. 2025-06-24 16:42:10 -03:00
Aleix Conchillo Flaqué
076a8938f0 add start_watchdog/reset_watchdog to tasks 2025-06-24 11:56:20 -07:00
Aleix Conchillo Flaqué
5a3457ba33 introduce task watchdog timers 2025-06-24 11:56:20 -07:00
Aleix Conchillo Flaqué
2fc224384d Merge pull request #2059 from pipecat-ai/aleix/heartbeatframe-control-frames
HeartbeatFrames are now control frames
2025-06-24 11:55:18 -07:00
Aleix Conchillo Flaqué
a4e6ea5a3f HeartbeatFrames are now control frames 2025-06-24 11:27:39 -07:00
Vanessa Pyne
d3c211f293 Merge pull request #2058 from pipecat-ai/vp-mcp-sse-up
follow up to #1887 - proper MCP SSE support
2025-06-24 13:06:01 -05:00
vipyne
20047c369e mcp: update examples to use SseServerParameter 2025-06-24 12:58:39 -05:00
vipyne
dd1ff237a8 lint mcp_service 2025-06-24 12:58:33 -05:00
Vanessa Pyne
39d80d0b0e Merge pull request #1887 from ezun-kim/feat/mcp-sse-params
Fix SSE server connection handling for MCP client
2025-06-24 12:58:05 -05:00
Kwindla Hultman Kramer
7a48316534 update google libraries used in google audio-in examples 2025-06-24 09:52:04 -07:00
Filipi da Silva Fuchter
031a93ac46 Merge pull request #2053 from pipecat-ai/sentry_dsn_environment_variable
Creating an environment variable for sentry dsn.
2025-06-24 12:10:20 -03:00
Mark Backman
ea6cc1aa95 Merge pull request #2052 from pipecat-ai/mb/11labs-keepalive
Send context_id when available in ElevenLabsTTSService keepalive message
2025-06-24 11:07:07 -04:00
Filipi Fuchter
365260ec44 Creating an environment variable for sentry dsn. 2025-06-24 11:57:14 -03:00
Mark Backman
2eb244c80a Send context_id when available in ElevenLabsTTSService keepalive message 2025-06-24 10:52:49 -04:00
Mark Backman
aee3011d61 Merge pull request #2037 from pipecat-ai/mb/11labs-close-context
Fix: Correctly close the context for ElevenLabsTTSService
2025-06-24 07:44:22 -04:00
Aleix Conchillo Flaqué
40496e7b0f Merge pull request #2034 from pipecat-ai/khk/pause-frames
small fix for processor pause/resume frames
2025-06-23 17:08:41 -07:00
Kwindla Hultman Kramer
6b24f89fa7 small fix for processor pause/resume frames 2025-06-23 16:44:32 -07:00
Filipi Fuchter
2097800042 Allowing to clear the turn analyser 2025-06-23 18:50:37 -03:00
Filipi Fuchter
6739318e68 Forcing user stopped speaking due to timeout to receive audio frame! 2025-06-23 18:50:02 -03:00
Filipi Fuchter
d0bd563d42 Logging the BaseException inside the cancel_task. 2025-06-23 18:48:44 -03:00
Filipi Fuchter
74280829fc Fixed an issue with the FastAPIWebsocketClient to disconnect in case the websocket is already closed. 2025-06-23 18:48:03 -03:00
Filipi Fuchter
3fde8880f2 Fixed a couple of places inside the FrameProcessor where we should not raise the exceptions. 2025-06-23 18:47:54 -03:00
Filipi Fuchter
98d39e0d38 Logging the last 10 frames received in case idle timeout is detected. 2025-06-23 18:47:17 -03:00
Filipi Fuchter
c9cebb5ffe Created an example for testing the bot and try to create freezing conditions. 2025-06-23 18:46:58 -03:00
Mark Backman
f52ac6e99c Merge pull request #1998 from pipecat-ai/mb/fix-38-smart-turn-fal 2025-06-23 17:15:29 -04:00
Mark Backman
787a6b1c6a Merge pull request #2038 from pipecat-ai/mb/openai-realtime-model-update
Update OpenAIRealtimeBetaLLMService model to gpt-4o-realtime-preview-…
2025-06-23 16:30:31 -04:00
Mark Backman
d00a91074e Update OpenAIRealtimeBetaLLMService model to gpt-4o-realtime-preview-2025-06-03 2025-06-23 16:26:42 -04:00
Mark Backman
4e11497a38 Merge pull request #2048 from thibaudbrg/patch-1
Fix missing video_in_enabled in vision bot.py for Moondream template
2025-06-23 16:11:50 -04:00
Tibo
0443d5202a Fix missing video_in_enabled in vision bot.py for Moondream template
The parameter video_in_enabled=True was missing in DailyParams, which prevented image capture 
from working. Without this parameter, UserImageRequestFrame would be sent but no actual image data would be captured from participants.

This fix enables the "Let me take a look" functionality to work as 
intended by allowing the transport to capture video frames for vision processing with Moondream.
2025-06-23 21:17:41 +02:00
Mark Backman
633c25cb13 Merge pull request #2039 from pipecat-ai/mb/remove-lang-validation
OpenAIRealtimeBetaLLMService accepts language for all InputAudioTrans…
2025-06-23 14:41:09 -04:00
jhpiedrahitao
d07f45132f update changelog 2025-06-23 12:54:00 -05:00
jhpiedrahitao
a51280afa6 add 13 and 14 type foundational examples for sambanova iontegration 2025-06-23 12:53:32 -05:00
Jorge Piedrahita Ortiz
be14eb2460 Merge branch 'pipecat-ai:main' into snova-jorgep/sambanova-integration 2025-06-23 12:23:00 -05:00
jhpiedrahitao
e26dbffcbe update sambanova init imports 2025-06-23 12:22:08 -05:00
Mark Backman
59992fd24a Merge pull request #2044 from pipecat-ai/mb/daily-rest-docstring
Add missing arg docstring in DailyRESTHelper
2025-06-23 11:24:44 -04:00
Mark Backman
455362ccaf Merge pull request #2022 from pipecat-ai/mb/turn-tracking-end-cancel-frame
TurnTrackingObserver ends turn upon seeing EndFrame, CancelFrame
2025-06-23 11:24:27 -04:00
Mark Backman
16c0e2460b TurnTrackingObserver ends turn upon seeing EndFrame, CancelFrame 2025-06-23 11:08:51 -04:00
Mark Backman
92246f7125 Add missing arg docstring in DailyRESTHelper 2025-06-22 13:44:59 -04:00
Mark Backman
7737335ec9 OpenAIRealtimeBetaLLMService accepts language for all InputAudioTranscription models 2025-06-21 10:08:46 -04:00
Mark Backman
5cc9b7e0d1 Fix: Correctly close the context for ElevenLabsTTSService 2025-06-20 15:47:03 -04:00
Mark Backman
8c6a441064 Merge pull request #2035 from smokyabdulrahman/feat/aws-polly-lexicon-names-support
Support AWS Polly Lexicon Names parameter
2025-06-20 10:03:27 -04:00
Alrahma
fddc058ce2 add CHANGELOG entry 2025-06-20 14:15:24 +01:00
Alrahma
89750086c5 Support AWS Polly Lexicon Names parameter
Documentation reference
[AWS Managing
Lexicons](https://docs.aws.amazon.com/polly/latest/dg/managing-lexicons.html)
2025-06-20 09:47:46 +01:00
Aleix Conchillo Flaqué
e69406c7e2 Merge pull request #2032 from pipecat-ai/aleix/aws-nova-sonic-function-calls
AWSNovaSonicLLMService: fix function calling
2025-06-19 14:42:47 -07:00
Aleix Conchillo Flaqué
878ae42d84 AWSNovaSonicLLMService: fix function calling 2025-06-19 14:26:34 -07:00
Aleix Conchillo Flaqué
d34ebfc126 Merge pull request #2027 from pipecat-ai/aleix/task-on-idle-timeout-repeated
PipelineTask: fix repeated on_idle_timeout
2025-06-19 14:13:10 -07:00
Aleix Conchillo Flaqué
028f7b2d65 PipelineTask: fix repeated on_idle_timeout 2025-06-19 09:14:10 -07:00
Mark Backman
0aa3ec50f2 Merge pull request #2023 from pipecat-ai/mb/allow-interruptions-true
allow_interruptions=True
2025-06-19 10:24:53 -04:00
Mark Backman
9146def21b Update examples to use default allow_interruptions, fixes to align examples 2025-06-19 10:07:32 -04:00
Aleix Conchillo Flaqué
ebb23a5a8c Merge pull request #2024 from pipecat-ai/aleix/audio-buffer-processor-sync-issues
AudioBufferProcessor: treat all streams as intermittent
2025-06-18 18:26:38 -07:00
Aleix Conchillo Flaqué
b118082984 AudioBufferProcessor: treat all streams as intermittent
This fixes an issue with STTMuteFilter that prevents user audio to be pushed
downstream.
2025-06-18 18:23:31 -07:00
Mark Backman
b5c0ac5f25 allow_interruptions=True 2025-06-18 20:33:40 -04:00
Filipi da Silva Fuchter
dc78e874af Merge pull request #2021 from pipecat-ai/gladia_stt_improvements_changelog
Adding the GladiaSTTService improvements in the changelog.
2025-06-18 18:25:36 -03:00
Filipi Fuchter
c30bde0a2b Adding the GladiaSTTService improvements in the changelog. 2025-06-18 16:19:58 -03:00
Filipi da Silva Fuchter
171597fbe9 Merge pull request #1952 from jqueguiner/feat/gladia-auto-reconnect
feat: Enhance GladiaSTTService with reconnection and audio buffer management features
2025-06-18 16:14:58 -03:00
jhpiedrahitao
fae2d272d5 fmt 2025-06-18 10:53:06 -05:00
jhpiedrahitao
03a067d3e6 add sambanova llm and stt 2025-06-18 10:50:42 -05:00
Mark Backman
f5d028f3b3 Merge pull request #2017 from pipecat-ai/mb/fix-11labs-voice-settings
fix: ElevenLabsTTSService voice settings not being sent
2025-06-18 09:56:46 -04:00
Mark Backman
e5b7dbba90 fix: ElevenLabsTTSService voice settings not being sent 2025-06-18 09:49:17 -04:00
Filipi da Silva Fuchter
7ffba1e0b3 Merge pull request #1950 from pipecat-ai/filipi/tavus_custom_tracks
Sending audio to Tavus using custom tracks
2025-06-18 07:57:19 -03:00
Filipi Fuchter
72cdbf0b78 Mentioning the Tavus improvements in the changelog. 2025-06-18 07:46:04 -03:00
Filipi Fuchter
8b4a86f629 Ignoring the audio level when creating the custom tracks. 2025-06-18 07:45:54 -03:00
Filipi Fuchter
fa15e64fc9 Test script that mimics the behavior expected to be supported by Tavus. 2025-06-18 07:45:38 -03:00
Filipi Fuchter
564f064c71 Refactoring TavusVideoService to send audio using WebRTC audio tracks instead of app-messages. 2025-06-18 07:44:51 -03:00
Filipi Fuchter
4062c7afa0 Refactoring TavusTransport to send audio using WebRTC audio tracks instead of app-messages. 2025-06-18 07:44:38 -03:00
Jean-Louis Queguiner
8071c4ba1c Merge branch 'main' into feat/gladia-auto-reconnect 2025-06-18 08:57:21 +02:00
jqueguiner
3d0ffbc832 🐛 (stt.py): handle websocket connection closure gracefully and log warnings
♻️ (stt.py): refactor reconnection logic into a separate method for clarity
 (stt.py): implement exponential backoff for reconnection attempts to improve reliability
2025-06-18 08:52:43 +02:00
Filipi da Silva Fuchter
1cac94bf97 Merge pull request #1925 from pipecat-ai/filipi/websocket_transport_example_twilio
Websocket client web app to test Twilio.
2025-06-17 16:24:18 -03:00
Mark Backman
c94c51d44f Fix: 38-smart-turn-fal 2025-06-17 15:10:52 -04:00
Mark Backman
96958933af Merge pull request #2016 from pipecat-ai/aleix/example-params-allow-async-objects
examples: create transport params async
2025-06-17 15:08:37 -04:00
Filipi Fuchter
2300c2632e Refactoring how we are organizing the twilio chatbot examples and improving the readmes 2025-06-17 16:08:35 -03:00
Filipi Fuchter
cbd0529674 Merge branch 'main' into filipi/websocket_transport_example_twilio 2025-06-17 15:54:31 -03:00
Filipi da Silva Fuchter
5614e35ac4 Merge pull request #2015 from pipecat-ai/bumping_pipecat_required_versions
Bumping pipecat-ai-krisp required version
2025-06-17 15:42:20 -03:00
Aleix Conchillo Flaqué
c11172caba examples: create transport params async 2025-06-17 11:37:42 -07:00
Filipi Fuchter
11b6e409bb Bumping pipecat-ai-krisp required version 2025-06-17 15:22:31 -03:00
Aleix Conchillo Flaqué
3dca95aa3c Merge pull request #2014 from pipecat-ai/aleix/daily-python-0.19.3
update daily-python to 0.19.3
2025-06-17 10:10:23 -07:00
Aleix Conchillo Flaqué
7ddc706434 update daily-python to 0.19.3 2025-06-17 09:30:28 -07:00
Aleix Conchillo Flaqué
20eebb08e9 update CHANGELOG with AWSTranscribeSTTService Polish support 2025-06-16 10:34:56 -07:00
Aleix Conchillo Flaqué
4abf41b85a Merge pull request #2011 from wuodar/wuodar/polish-lang-aws-transcribe
Support polish language in Amazon Transcribe
2025-06-16 10:33:55 -07:00
Aleix Conchillo Flaqué
e426f7ee7c Merge pull request #2012 from pipecat-ai/aleix/frame-pause-resume-frames
FrameProcessor: handle new FrameProcessorPauseFrame/FrameProcessorResumeFrame
2025-06-16 10:32:38 -07:00
Aleix Conchillo Flaqué
14dc6a7984 FrameProcessor: handle new FrameProcessorPauseFrame/FrameProcessorResumeFrame 2025-06-16 10:31:33 -07:00
Mark Backman
e0a24a3f07 Merge pull request #2006 from pipecat-ai/mb/expose-function-calls-in-progress
Expose has_function_calls_in_progress property
2025-06-16 12:49:07 -04:00
Mark Backman
d1bee22d73 Expose has_function_calls_in_progress property 2025-06-16 12:45:16 -04:00
Jon Taylor
d73f7908f2 Merge pull request #2008 from pipecat-ai/khk/groq-audio
fix groq wav file header parsing
2025-06-16 14:09:09 +01:00
Aleix Conchillo Flaqué
a4ea0d2b82 dev-requirements: update pyright 1.1.400 and ruff 0.11.13 2025-06-15 21:05:03 -07:00
Kacper Włodarczyk
e2c15169b8 feat: support polish language in Amazon Transcribe 2025-06-15 21:44:06 +02:00
Kwindla Hultman Kramer
fe16ed3c73 added changelog entry 2025-06-15 10:49:40 -07:00
Filipi Fuchter
80ce097f90 Using relative URL for the websocket. 2025-06-15 10:49:25 -03:00
Filipi Fuchter
eceaf8a46b Making the path to the web client relative 2025-06-14 21:07:15 -03:00
Kwindla Hultman Kramer
1e3fa4a9c7 fix groq wav file header parsing 2025-06-14 17:41:44 -04:00
Filipi da Silva Fuchter
dc640a7591 Merge pull request #2001 from pipecat-ai/filipi/google_stt_reconnection_issue
Fixed an issue with `GoogleSTTService` where it was constantly reconnecting
2025-06-13 08:29:18 -03:00
Filipi Fuchter
1f072d182c Merge branch 'main' into filipi/google_stt_reconnection_issue
# Conflicts:
#	CHANGELOG.md
2025-06-13 08:26:00 -03:00
Mark Backman
1d64e04ed5 Merge pull request #2002 from pipecat-ai/mb/google-fix-ttfb
Fix: GoogleLLMService TTFB
2025-06-12 12:10:01 -04:00
Mark Backman
22f4f0b79e Update 14e example name 2025-06-12 11:45:59 -04:00
Mark Backman
69c63293fb fix: GoogleLLMService TTFB value 2025-06-12 11:43:27 -04:00
Filipi Fuchter
c1db13ceeb Fixed an issue with GoogleSTTService where it was constantly reconnecting before starting to receive audio from the user. 2025-06-12 12:07:33 -03:00
Filipi Fuchter
70eadee0aa Bumping the @pipecat-ai/websocket-transport dependency. 2025-06-11 18:30:16 -03:00
Aleix Conchillo Flaqué
228afe01ed Merge pull request #1993 from pipecat-ai/aleix/pipecat-0.0.71
update CHANGELOG for 0.0.71
2025-06-10 14:42:09 -07:00
Aleix Conchillo Flaqué
61a5154e49 update CHANGELOG for 0.0.71 2025-06-10 14:34:30 -07:00
Sunah Suh
d3df75aaa0 Add additional_span_attributes param to PipelineTask for extra otel… (#1992) 2025-06-10 17:32:24 -04:00
Aleix Conchillo Flaqué
c59180dd6e udpate CHANGELOG 2025-06-10 14:23:02 -07:00
Mark Backman
e4c2310632 Merge pull request #1990 from pipecat-ai/mb/more-cartesia-stt
Add Cartesia STT docs link to README, fix set_model error
2025-06-10 17:19:11 -04:00
Aleix Conchillo Flaqué
e1735a2da1 Merge pull request #1991 from pipecat-ai/aleix/pipecat-0.0.70
update CHANGELOG for 0.0.70
2025-06-10 14:08:52 -07:00
Aleix Conchillo Flaqué
c101c9c8e1 update CHANGELOG for 0.0.70 2025-06-10 13:37:28 -07:00
Aleix Conchillo Flaqué
96dc162de5 Merge pull request #1988 from pipecat-ai/aleix/update-examples-22b
examples(22b): remove unnecessary parallel pipeline branch
2025-06-10 12:58:37 -07:00
Mark Backman
257dbe3104 Fix model param error 2025-06-10 15:14:47 -04:00
Mark Backman
cd98657e3c Add Cartesia STT docs link to README 2025-06-10 15:09:13 -04:00
Aleix Conchillo Flaqué
03eb22fe0a examples(22b): remove unnecessary parallel pipeline branch 2025-06-10 09:05:58 -07:00
Filipi Fuchter
0073a868d4 Websocket client web app to test Twilio. 2025-06-10 11:34:02 -03:00
Mark Backman
0bb61d72ab Merge pull request #1984 from pipecat-ai/mb/cartesia-stt-cleanup
CartesiaSTTService cleanup
2025-06-10 10:30:18 -04:00
Mark Backman
f758508a82 Merge pull request #1978 from pipecat-ai/mb/rime-languages
Add languages to RimeHttpTTSService, extend lang support to German an…
2025-06-10 10:27:15 -04:00
Mark Backman
69d0218d7e Add languages to RimeHttpTTSService, extend lang support to German and French 2025-06-10 10:20:41 -04:00
Aleix Conchillo Flaqué
eb5e5ab1df update CHANGELOG 2025-06-09 20:22:39 -07:00
Aleix Conchillo Flaqué
093697906c Merge pull request #1954 from WebinarGeek/wg/gladia-informal-translations
Gladia informal translations
2025-06-09 20:21:40 -07:00
Aleix Conchillo Flaqué
efe96b7ed1 Merge pull request #1986 from pipecat-ai/aleix/daily-python0.19.2
pyproject: update daily-python to 0.19.2
2025-06-09 19:46:14 -07:00
Aleix Conchillo Flaqué
7ecdd41ab9 pyproject: update daily-python to 0.19.2 2025-06-09 17:29:07 -07:00
Mark Backman
aec70d61e9 CartesiaSTTService cleanup 2025-06-09 15:20:57 -04:00
Mark Backman
2efac13344 Merge pull request #1983 from pipecat-ai/mb/exotel-resampling
Resample audio in ExotelFrameSerializer
2025-06-09 14:41:08 -04:00
Mark Backman
15aeb11c36 Resample audio in ExotelFrameSerializer 2025-06-09 14:02:25 -04:00
Mark Backman
e705f4d984 Merge pull request #1972 from Vaibhav159/vl_add_exotel_serializer
adding exotel serializer
2025-06-09 13:54:26 -04:00
Shrey Gupta
96fa62fdfe [Add] Support for Cartesia AI STT (#1982) 2025-06-09 14:51:01 -03:00
Mark Backman
845c70797a Merge pull request #1975 from pipecat-ai/mb/11labs-flush-context-reset
fix: ElevenLabsTTSService reset context when flushing audio
2025-06-09 13:21:25 -04:00
kompfner
16048956c3 Merge pull request #1956 from pipecat-ai/pk/make-add-observer-sync
Make `PipelineTask.add_observer()` synchronous. This allows callers t…
2025-06-09 13:19:34 -04:00
Mark Backman
cf2f4b5902 fix: ElevenLabsTTSService reset context when flushing audio 2025-06-09 13:17:55 -04:00
marcus-daily
db46f33f34 Update to Android transports 0.3.7 2025-06-09 17:09:59 +01:00
Aleix Conchillo Flaqué
25d1515daf Merge pull request #1979 from pipecat-ai/aleix/buffer-tts-before-playback
buffer audio from TTS service before pushing frames
2025-06-09 08:43:55 -07:00
Paul Kompfner
a3469cd59f Add CHANGELOG entry describing PipelineTask.add_observer() being made synchronous 2025-06-09 11:37:30 -04:00
Paul Kompfner
513ce26200 Add unit test exercising synchronous usage of PipelineTask.add_observer() right after initializing the PipelineTask (before anything else is done with it) 2025-06-09 11:30:24 -04:00
Paul Kompfner
1cd96f94ff Make PipelineTask.add_observer() synchronous. This allows callers to call it before run()ning the PipelineTask first. Without this change, if they tried to do that, they would get an error because the TaskManager's event loop hadn't been set yet. 2025-06-09 11:30:24 -04:00
Aleix Conchillo Flaqué
901dd041f0 buffer audio from TTS service before pushing frames 2025-06-09 07:29:09 -07:00
Vaibhav159
a2ee94651e removing resampling 2025-06-07 12:53:55 +05:30
Aleix Conchillo Flaqué
abdce063f1 Merge pull request #1973 from pipecat-ai/aleix/assemblyai-yield-none
AssemblyAISTTService: yield None instead of Frame()
2025-06-06 15:12:16 -07:00
Aleix Conchillo Flaqué
a33ce5e4bf AssemblyAISTTService: yield None instead of Frame() 2025-06-06 14:41:01 -07:00
Filipi da Silva Fuchter
c9575eaef9 Merge pull request #1911 from pipecat-ai/filipi/websocket_transport_example
Adding support to WebsocketTransport
2025-06-06 17:25:07 -03:00
Filipi Fuchter
1e74476a71 Refactoring to use the observer inside the pipelinetask, and moving to start the bot inside on_client_ready. 2025-06-06 17:22:50 -03:00
Filipi Fuchter
82935884c4 Mentioning the new websocket example in the changelog. 2025-06-06 17:17:11 -03:00
Filipi Fuchter
d774a23768 Improving the readme to mention that can choose which server websocket to use. 2025-06-06 17:12:05 -03:00
Filipi Fuchter
e9f041e170 Removing the old websocket-server example 2025-06-06 17:09:01 -03:00
Filipi Fuchter
1f51b6e4f1 A Pipecat example demonstrating how to use WebsocketTransport 2025-06-06 17:08:43 -03:00
Filipi Fuchter
028650249c Adding support in ProtobufFrameSerializer to deserialize MessageFrame. 2025-06-06 17:07:39 -03:00
Vaibhav159
534197239f updating changelog 2025-06-07 00:24:54 +05:30
Vaibhav159
d2f4bb574c adding exotel serializer 2025-06-07 00:22:41 +05:30
jqueguiner
25ff8ef37b (config.py): add new configuration options for lip-sync optimization, context adaptation, and additional context to enhance translation accuracy
♻️ (stt.py): increase default max buffer size from 5MB to 20MB to accommodate larger audio data
♻️ (stt.py): simplify audio sending logic by removing chunking and sending the entire buffered audio at once for improved performance
2025-06-05 16:51:29 -07:00
Aleix Conchillo Flaqué
07fb1a2c39 Merge pull request #1967 from counterleft/unused-http-client-session
Remove instantiation of unused http client session from certain examples
2025-06-05 12:59:01 -07:00
Aleix Conchillo Flaqué
581b800c43 Merge pull request #1961 from ken-kuro/patch-1
fix(piper-tts): typo
2025-06-05 12:57:58 -07:00
Aleix Conchillo Flaqué
30ca39287f Merge pull request #1962 from ken-kuro/patch-2
fix(fastapi_websocket): typo
2025-06-05 12:57:22 -07:00
Aleix Conchillo Flaqué
01fa9698de Merge pull request #1960 from pipecat-ai/aleix/disable-uvloop
disable uvloop by default and just let the user set it
2025-06-05 12:12:47 -07:00
Brian Mathiyakom
10bd969636 Remove instantiation of unused http client session
These examples don't make any HTTP requests with `session` so there
doesn't seem be a need to create one in the first place. Probably a
copy-paste from a previous example.
2025-06-05 11:45:13 -07:00
Kendrick Ha
f7761f2b61 fix(fastapi_websocket): typo 2025-06-05 13:55:28 +07:00
Kendrick Ha
49ff38a21f fix(piper-tts): typo 2025-06-05 13:50:56 +07:00
Aleix Conchillo Flaqué
8d161306c7 disable uvloop by default and just let the user set it 2025-06-04 21:25:06 -07:00
Vanessa Pyne
027a82dff1 Merge pull request #1958 from pipecat-ai/vp-livekit-fix
fix: transports/services/livekit.py typo
2025-06-04 12:27:47 -05:00
vipyne
cb409d58e0 fix: transports/services/livekit.py typo 2025-06-04 11:14:21 -05:00
Dan Berg
094e2f8151 Fix formatting 2025-06-03 17:21:51 +02:00
Dan Berg
71d121aeb9 Update CHANGELOG.md explaining informal on Gladia TranslationConfig 2025-06-03 17:15:29 +02:00
Dan Berg
b1a88af43c Add informal to Gladia TranslationConfig 2025-06-03 17:10:52 +02:00
Filipi da Silva Fuchter
f73eb4ebd9 Merge pull request #1949 from pipecat-ai/filipi/transport_destination_issue
Fixed transport destination issue
2025-06-03 08:41:34 -03:00
Filipi Fuchter
31ca9be299 Fixing missing await to self.reset. 2025-06-03 08:37:47 -03:00
jqueguiner
02cc6f3d56 Enhance GladiaSTTService with reconnection and audio buffer management features
- Added parameters for maximum reconnection attempts, reconnection delay, and maximum audio buffer size.
- Implemented automatic reconnection logic with exponential backoff.
- Introduced audio buffer management to handle audio data efficiently, including trimming excess data.
- Updated connection handling to ensure proper cleanup and management of WebSocket connections.
- Enhanced audio sending logic to support buffered audio transmission after reconnections.
2025-06-03 03:16:57 -07:00
Filipi Fuchter
1642c082d1 Describing the fix in the changelog. 2025-06-02 22:28:31 -03:00
Filipi Fuchter
892d213442 Fixing issue to keep the transport_destination. 2025-06-02 22:16:10 -03:00
Filipi Fuchter
fc24267e09 Waiting for the LLM response to reset. 2025-06-02 22:15:53 -03:00
Aleix Conchillo Flaqué
9b71bdc608 Merge pull request #1947 from pipecat-ai/aleix/pipecat-0.0.69
update CHANGELOG for 0.0.69
2025-06-02 12:51:51 -07:00
Aleix Conchillo Flaqué
310be89895 update CHANGELOG for 0.0.69 2025-06-02 12:07:50 -07:00
Aleix Conchillo Flaqué
71fbd57e12 Merge pull request #1938 from pipecat-ai/aleix/custom-interruption-strategies
allow custom interruption strategies
2025-06-02 12:05:50 -07:00
Aleix Conchillo Flaqué
ab4b48c823 examples(04a): fix daily_runner import 2025-06-02 12:01:26 -07:00
Aleix Conchillo Flaqué
532767cfa1 LLMUserContextAggregator: reset strategies when reseting the aggregator 2025-06-02 12:01:26 -07:00
Aleix Conchillo Flaqué
5512de3221 allow custom interruption strategies 2025-06-02 12:01:26 -07:00
Mark Backman
13546d5e8f Merge pull request #1946 from pipecat-ai/mb/fix-11labs-context
fix: Use AudioContextWordTTSService context methods in ElevenLabsTTSS…
2025-06-02 14:55:49 -04:00
Mark Backman
c6f1aa8086 fix: Use AudioContextWordTTSService context methods in ElevenLabsTTSService 2025-06-02 14:49:05 -04:00
Mark Backman
5606c47cb7 Merge pull request #1945 from pipecat-ai/mb/gemini-2.0
Reverting Gemini Live model back to gemini-2.0-flash-live-001
2025-06-02 14:25:30 -04:00
Filipi da Silva Fuchter
7f7cd96211 Merge pull request #1944 from pipecat-ai/fixing_tavus_transport
Adding the direction when pushing the frame.
2025-06-02 15:21:58 -03:00
Filipi Fuchter
b828bfd890 Adding the direction when pushing the BotStartedSpeaking and BotStoppedSpeaking frames. 2025-06-02 15:05:56 -03:00
Mark Backman
31d084eb78 Reverting Gemini Live model back to gemini-2.0-flash-live-001 2025-06-02 13:29:05 -04:00
Mark Backman
ab18b280e9 Merge pull request #1943 from pipecat-ai/mb/add-transcription-19-openai
Add a TranscriptProcessor to 19-openai-realtime-beta.py
2025-06-02 13:01:52 -04:00
Mark Backman
24e89c4081 Merge pull request #1936 from pipecat-ai/mb/fix-01-quickstart
Add daily to the foundational examples requirements.txt
2025-06-02 12:55:36 -04:00
Mark Backman
e129390f56 Add a TranscriptProcessor to 19-openai-realtime-beta.py 2025-06-02 11:38:49 -04:00
Mark Backman
4d7c87bb4c Merge pull request #1941 from pipecat-ai/vp-observer-up
chore: move observers arg in p2p-webrtc example; add deprecated to in…
2025-06-02 11:17:48 -04:00
Mark Backman
dac3f82a75 Merge pull request #1934 from counterleft/use-disconnect-on-small-webrtc-connection-example
Fix type checker error with missing function call in the small WebRTC transport example
2025-06-02 11:17:30 -04:00
Mark Backman
fd860921f1 Add daily to the foundational examples requirements.txt 2025-06-02 11:10:49 -04:00
vipyne
0482ccd48b chore: move observers arg in p2p-webrtc example; add deprecated to in line comments 2025-06-02 09:41:09 -05:00
Dominic Stewart
b8b1990617 Fix example env file (#1939)
* Fixed typo in the example env files
2025-06-02 15:12:43 +09:00
Dominic Stewart
70951b1198 Add simplified pstn examples (#1822)
* Add simplified pstn examples
* Add daily_twilio_sip_dial_out example
2025-06-02 14:50:21 +09:00
Mark Backman
6d24514ace Merge pull request #1937 from pipecat-ai/mb/fix-message-for-logging
fix: correctly display non-roman characters
2025-06-01 12:49:48 -04:00
Aleix Conchillo Flaqué
49915ceb84 Merge pull request #1683 from pipecat-ai/aleix/run-function-calls-sequentially
run function calls sequentially or in parallel
2025-06-01 09:47:35 -07:00
Mark Backman
925b13e337 fix: correctly display non-roman characters 2025-06-01 12:29:26 -04:00
Aleix Conchillo Flaqué
ef3143d558 LLMService: don't run function calls if none are given 2025-05-31 14:01:56 -07:00
Mark Backman
ed84637b55 Add additional function call for testing to 14e, 14r, 19, 19a, 26b 2025-05-31 13:57:52 -07:00
Aleix Conchillo Flaqué
897a944478 examples(14,14a): add restaurant recommendation function call 2025-05-31 13:57:52 -07:00
Aleix Conchillo Flaqué
d86343c38d examples: update to use on_function_calls_started event 2025-05-31 13:57:52 -07:00
Aleix Conchillo Flaqué
297afdd126 LLMService: add new FunctionCallsStartedFrame 2025-05-31 13:57:52 -07:00
Aleix Conchillo Flaqué
f0cbdc4e68 LLMService: add on_function_calls_started event 2025-05-31 13:57:52 -07:00
Aleix Conchillo Flaqué
40b52cadde LLMService: s/FunctionCallLLM/FunctionCallFromLLM/ s/FunctionCallRunner/FunctionCallRunnerItem/ 2025-05-31 13:57:52 -07:00
Aleix Conchillo Flaqué
04bf85ddfe LLMService: allow executing tasks sequentially and in parallel 2025-05-31 13:57:52 -07:00
Aleix Conchillo Flaqué
4809684a13 LLMService: cancel function calls on interruptions by default 2025-05-31 13:57:50 -07:00
Aleix Conchillo Flaqué
1eb50ad88f LLMService: pass LLM function calls all at once 2025-05-31 13:57:36 -07:00
Aleix Conchillo Flaqué
52569bcdb2 LLMService: don't allow running functions concurrently for now 2025-05-31 13:57:36 -07:00
Aleix Conchillo Flaqué
a50a407415 LLMService: run function calls sequentially 2025-05-31 13:57:35 -07:00
Aleix Conchillo Flaqué
9f223442c2 Merge pull request #1935 from pipecat-ai/aleix/script-evals
improve evals
2025-05-30 19:14:59 -07:00
Mark Backman
c647114bb9 Merge pull request #1927 from pipecat-ai/mb/gemini-tracing 2025-05-30 22:01:44 -04:00
Mark Backman
43719ec737 Update CHANGELOG 2025-05-30 21:36:39 -04:00
Mark Backman
8602557985 Refactor Gemini tracing to more closely match OpenAI Realtime, add TTFB metrics 2025-05-30 21:36:00 -04:00
Mark Backman
dd1f7d0875 Add tracking to OpenAI Realtime 2025-05-30 21:36:00 -04:00
Mark Backman
ec39e794d3 _handle_transcription 2025-05-30 21:36:00 -04:00
Mark Backman
7b1a937d4c Add tracing for Gemini Live 2025-05-30 21:36:00 -04:00
Mark Backman
0fd38d8115 Merge pull request #1931 from pipecat-ai/mb/num-words
Add support for interruption strategies
2025-05-30 21:14:02 -04:00
Mark Backman
7a4efc6212 Code review feedback 2025-05-30 21:09:15 -04:00
Brian Mathiyakom
2eb2c5a413 Use disconnect() because close() doesn't exist
SmallWebRTCConnection doesn't have a `close()`. There's a `_close()` but I assume that's private due to its naming. The closest function that uses `_close()` is `disconnect()`. I assume then, that the intended resource freeing function call should be to `disconnect()`.
2025-05-30 17:14:53 -07:00
Aleix Conchillo Flaqué
2fcfb0aa9f evals: don't use Deepgram's smart formatting 2025-05-30 16:55:55 -07:00
Aleix Conchillo Flaqué
f1df079512 evals: allow running a single eval 2025-05-30 16:55:55 -07:00
Aleix Conchillo Flaqué
d77bedbafb evals: move scripts/release to script/evals and add README 2025-05-30 15:04:05 -07:00
Mark Backman
b34c593c54 Add changelog entry 2025-05-30 16:48:42 -04:00
Mark Backman
62efbc3342 Add foundational example 42 2025-05-30 16:48:42 -04:00
Mark Backman
2d609a0bde Update LLmUserContextAggregator to conditionally push_aggregation 2025-05-30 16:48:42 -04:00
Mark Backman
6bc4b4a17f Update BaseInputTransport to not push StartInterruptionFrame when InterruptionConfig is set and bot is speaking 2025-05-30 16:48:42 -04:00
Mark Backman
b489e52080 Add InterruptionConfig 2025-05-30 16:15:20 -04:00
Aleix Conchillo Flaqué
a8aaeec52b Merge pull request #1926 from pipecat-ai/aleix/pause-base-input-transport
handle StopFrame in base input transport and stop pushing frames
2025-05-30 08:27:20 -07:00
Mark Backman
ad7eec181e Merge pull request #1923 from philipp-eisen/philipp/fix-ttfb_ms-implementation
Fix implementation of ttfb_ms metric
2025-05-30 11:03:42 -04:00
Aleix Conchillo Flaqué
b33897ffb9 SmallWebRTCTransport: don't initialize if a second StartFrame is received 2025-05-30 07:51:51 -07:00
Aleix Conchillo Flaqué
1c3d3f2f4b WebsocketClientTransport: don't initialize if a second StartFrame is received 2025-05-30 07:51:51 -07:00
Aleix Conchillo Flaqué
9a5a1edb6b WebsocketServerTransport: don't initialize if a second StartFrame is received 2025-05-30 07:51:51 -07:00
Aleix Conchillo Flaqué
f2eb869b02 FastAPIWebsocketTransport: don't initialize if a second StartFrame is received 2025-05-30 07:51:51 -07:00
Aleix Conchillo Flaqué
0c7e3cfcb2 LiveKitTransport: don't initialize if a second StartFrame is received 2025-05-30 07:51:51 -07:00
Aleix Conchillo Flaqué
24e19db29e TavusTransport: don't initialize if a second StartFrame is received 2025-05-30 07:51:51 -07:00
Aleix Conchillo Flaqué
bc6d7b7bbd examples(phone-chatbot): don't show dangling tasks in first pipeline 2025-05-30 07:51:51 -07:00
Aleix Conchillo Flaqué
cad271068e BaseInputTransport: handle StopFrame and pause pushing frames 2025-05-30 07:51:49 -07:00
Philipp Eisen
3425293115 Ensure correct formatting 2025-05-30 15:41:45 +02:00
Mark Backman
20dbfec3a9 Merge pull request #1876 from m-ods/m-ods/assemblyai-universal-streaming
Update AssemblyAI Streaming STT
2025-05-30 08:55:43 -04:00
marcus-daily
170057a75a Updating dependency version 2025-05-30 12:38:32 +01:00
marcus-daily
b86b761e0b Fixing Yaml syntax 2025-05-30 12:38:32 +01:00
marcus-daily
da0d2f0266 Small WebRTC transport demo app for Android 2025-05-30 12:38:32 +01:00
Martin Schweiger
321ea27c34 changelog entry 2025-05-30 17:15:58 +08:00
Philipp Eisen
b712e6b9aa Switch ttfb metric to use seconds instead 2025-05-30 11:13:26 +02:00
Martin Schweiger
b3652e6527 set vad_force_turn_endpoint to True by default 2025-05-30 17:07:54 +08:00
Philipp Eisen
bc97f397ef Switch to using float instead of int, but keep ms 2025-05-30 10:27:20 +02:00
Martin Schweiger
e5da3f6e68 add tracing 2025-05-30 10:55:19 +08:00
Martin Schweiger
8400539acf set formatted_finals true by default 2025-05-30 10:40:01 +08:00
Martin Schweiger
b5eac8dfed add message to TranscriptionFrame result 2025-05-30 10:39:07 +08:00
Martin Schweiger
ba312b5591 Merge branch 'main' into m-ods/assemblyai-universal-streaming 2025-05-30 10:29:49 +08:00
Martin Schweiger
f23572b318 Merge branch 'main' into m-ods/assemblyai-universal-streaming 2025-05-30 10:11:02 +08:00
Martin Schweiger
db838634e7 fix: double finals bug 2025-05-30 10:00:31 +08:00
Martin Schweiger
7f2e848a5c use FrameProcessor class methods 2025-05-30 09:40:46 +08:00
Martin Schweiger
096e854d50 remove .dict() 2025-05-30 09:31:20 +08:00
Martin Schweiger
3ffe8b3155 remove parse_obj 2025-05-30 09:29:51 +08:00
Mark Backman
a471f49b61 Merge pull request #1889 from pipecat-ai/mb/add-dtmf-aggregator
Add DTMFAggregator
2025-05-29 17:13:28 -04:00
Mark Backman
4d2a02f318 Refactor to create task on StartFrame, also cleanup 2025-05-29 17:10:54 -04:00
Mark Backman
0bec7db03b Emit a BotInterruptionFrame when the first keypress of a sequence is received 2025-05-29 17:07:18 -04:00
Mark Backman
74827f983f Add tests, improve frame timing 2025-05-29 17:07:18 -04:00
Mark Backman
0ed46f457e Add DTMFAggregator 2025-05-29 17:07:17 -04:00
Aleix Conchillo Flaqué
36b731be73 Merge pull request #1915 from pipecat-ai/aleix/uvloop-event-loop
use uvloop as the new event loop on Linux and macOS
2025-05-29 14:06:44 -07:00
Aleix Conchillo Flaqué
62fbdd4e81 Merge pull request #1922 from pipecat-ai/aleix/output-transport-cleanup
output transports cleanup
2025-05-29 14:06:17 -07:00
Aleix Conchillo Flaqué
ca7b0650c2 examples: capture camera or screen. allow setting framerate 2025-05-29 13:16:44 -07:00
Aleix Conchillo Flaqué
67dd146038 output transports cleanup 2025-05-29 13:16:44 -07:00
Aleix Conchillo Flaqué
fb66df2efd use uvloop as the new event loop on Linux and macOS 2025-05-29 11:24:21 -07:00
Aleix Conchillo Flaqué
2395ca0057 Merge pull request #1921 from pipecat-ai/aleix/transcription-frame-result
add STT result field to TranscriptionFrame/InterimTranscriptionFrame
2025-05-29 11:22:38 -07:00
Aleix Conchillo Flaqué
d203789490 add STT result field to TranscriptionFrame/InterimTranscriptionFrame 2025-05-29 11:21:44 -07:00
Aleix Conchillo Flaqué
7ea0e31cd4 Merge pull request #1924 from pipecat-ai/aleix/new-examples-package
add new pipecat.examples package and make runner public
2025-05-29 11:18:22 -07:00
Mark Backman
d3bf13a503 Merge pull request #1917 from pipecat-ai/mb/fix-twilio-chatbot-client
fix: update websocket_client to use FrameProcessorSetup.task_manager
2025-05-29 14:05:27 -04:00
Mark Backman
ea91970499 Update the twilio-chatbot client to work with the updated server, which requires call_sid 2025-05-29 14:02:49 -04:00
Mark Backman
803b3f2cc4 fix: update websocket_client to use FrameProcessorSetup 2025-05-29 14:02:47 -04:00
Aleix Conchillo Flaqué
1788ba6c5c add new pipecat.examples package and make runner public 2025-05-29 10:43:12 -07:00
Philipp Eisen
5209bd3d9f Fix implemetation of ttfb_ms metric 2025-05-29 19:24:04 +02:00
Aleix Conchillo Flaqué
cb9178f1ec Merge pull request #1914 from pipecat-ai/aleix/base-output-daily-handle-dtmf-frames
output transport can now handle DTMF keypress
2025-05-29 09:35:01 -07:00
Aleix Conchillo Flaqué
5676920a6a BaseOutputTransport/DailyTransport: allow sending DTMF keypress 2025-05-29 09:33:29 -07:00
Aleix Conchillo Flaqué
513221d9fd added OutputDTMFUrgentFrame 2025-05-29 09:32:57 -07:00
Mark Backman
a33d0b4b53 Merge pull request #1904 from pipecat-ai/mb/aws-bedrock-no-op-tool
Add no_op tool to AWSBedrockLLMService
2025-05-29 10:29:19 -04:00
Mark Backman
bee242b781 Safer check when using the no_operation tool 2025-05-29 10:25:20 -04:00
Mark Backman
fa1c98ff29 Add no_op tool to AWSBedrockLLMService 2025-05-29 10:25:19 -04:00
Mark Backman
ae3a7d9bed Merge pull request #1920 from alexflorensa/alexflorensa/set-deepgram-model-name
fix(deepgram-stt): set model name to Deepgram STT
2025-05-29 09:41:56 -04:00
alexflorensa
0c2efb312c fix(deepgram-stt) Set model name to Deepgram STT 2025-05-29 12:08:16 +02:00
Aleix Conchillo Flaqué
cf8eeaab0b Merge pull request #1909 from pipecat-ai/aleix/daily-expose-transcription-functions
DailyTransport: expose start_transcription/stop_transcription
2025-05-28 17:06:40 -07:00
Vanessa Pyne
2f8cb3ce76 Merge pull request #1804 from pipecat-ai/vp-text-webrtc-chatbot
add text chatbot example using small webrtc transport
2025-05-28 14:25:42 -05:00
vipyne
821da723c0 update SmallWebRTCTransport text examples with new run_example 2025-05-28 13:54:45 -05:00
Filipi Fuchter
575b97ba60 Some improvements and cleanups in the SmallWebRTCTransport text examples. 2025-05-28 13:54:45 -05:00
vipyne
cc0819b709 examples: add text and audio over webrtc transport
update filenames

add high level comments to 41* examples
2025-05-28 13:54:45 -05:00
vipyne
318d6f042b examples: add text chatbot example using small webrtc transport
examples: send small webrtc UI updates in RTVIServerMessageFrame

add explanation about RTVI server messages being specific to
small-webrtc-prebuilt UI
2025-05-28 13:54:45 -05:00
Aleix Conchillo Flaqué
05ae3a1703 DailyTransport: expose start_transcription/stop_transcription 2025-05-28 11:21:26 -07:00
Aleix Conchillo Flaqué
8e54805e62 Merge pull request #1908 from pipecat-ai/aleix/add-manifest-file-to-reduce-sdist
add MANIFEST.in to reduce sdist tarball size
2025-05-28 10:15:38 -07:00
Aleix Conchillo Flaqué
64399a72f3 add MANIFEST.in to reduce sdist tarball size 2025-05-28 10:09:39 -07:00
Aleix Conchillo Flaqué
6c33f0b0bd Merge pull request #1907 from pipecat-ai/aleix/pipecat-0.0.68
update CHANGELOG for pipecat 0.0.68
2025-05-28 09:41:34 -07:00
Aleix Conchillo Flaqué
aca304b395 update CHANGELOG for pipecat 0.0.68 2025-05-28 09:08:08 -07:00
Aleix Conchillo Flaqué
db5c9e67be Merge pull request #1906 from pipecat-ai/aleix/daily-python-0.19.1
pyproject: update daily-python to 0.19.1
2025-05-28 09:05:13 -07:00
Aleix Conchillo Flaqué
2313cec792 pyproject: update daily-python to 0.19.1 2025-05-28 00:36:40 -07:00
Aleix Conchillo Flaqué
83acaf692a Merge pull request #1890 from pipecat-ai/aleix/examples-multi-transport
add support for multiple transports to foundational examples
2025-05-28 00:27:01 -07:00
Aleix Conchillo Flaqué
e9aeb2662b scripts: allow specifying a name for the test run 2025-05-28 00:22:55 -07:00
Aleix Conchillo Flaqué
356f4039e4 scripts: allow storing logs for release evals 2025-05-27 21:10:22 -07:00
Aleix Conchillo Flaqué
736c7f1f30 scripts: allow storing audio for release evals 2025-05-27 18:09:25 -07:00
Aleix Conchillo Flaqué
2994448036 introduce release evals
This is an initial attempt to implement evals for all (or most) of our
foundational examples. Before we release, we want to make sure all of them work
and reply properly. Until now this has been done manually, hopefully this will
be useful to speed up our release process.
2025-05-27 17:42:52 -07:00
Aleix Conchillo Flaqué
d476d9ea05 examples: remove "on_client_closed"
This has been replaced for "on_client_disconnected" in SmallWebRTCTransport to
match other transports and therefore it is not necessary anymore.
2025-05-27 17:42:52 -07:00
Filipi Fuchter
bf31bce440 Updated SmallWebRTCTransport to align with how other transports handle on_client_disconnected 2025-05-27 17:42:52 -07:00
Aleix Conchillo Flaqué
6393e89022 examples(foundational): update handle_signint depending on transport 2025-05-27 17:42:52 -07:00
Aleix Conchillo Flaqué
884268fce3 examples(foundational): allow running examples with twilio 2025-05-27 17:42:52 -07:00
Aleix Conchillo Flaqué
071a9307c9 transport(websocket): do not require a frame serializer 2025-05-27 17:42:52 -07:00
Aleix Conchillo Flaqué
2cdfaa0a82 examples(foundational): support multiple transports 2025-05-27 17:42:52 -07:00
Aleix Conchillo Flaqué
ecf878e14d DailyTransport: allow requesting video frames with any framerate 2025-05-27 17:42:50 -07:00
Aleix Conchillo Flaqué
4eed335bc7 PipelineTask: check if pipeline has already been cancelled 2025-05-27 17:40:39 -07:00
Aleix Conchillo Flaqué
2e57bb74d2 BaseObject: do not raise exception if event handler not registered 2025-05-27 17:40:35 -07:00
Aleix Conchillo Flaqué
0a39769cd0 Merge pull request #1901 from pipecat-ai/aleix/deepgram-sdk-4.1.0
pyproject: update deepgram-sdk to 4.1.0
2025-05-27 17:20:54 -07:00
Aleix Conchillo Flaqué
bdb6a9e5d1 Merge pull request #1903 from pipecat-ai/aleix/openpipe-4.50.0
pyproject: update openpipe to 4.50.0
2025-05-27 17:20:30 -07:00
Aleix Conchillo Flaqué
f88e0eb96d pyproject: update openpipe to 4.50.0 2025-05-27 15:22:55 -07:00
Aleix Conchillo Flaqué
0099f60d29 deepgram: fix an issue with user provided LiveOptions 2025-05-27 15:19:17 -07:00
Filipi da Silva Fuchter
eaf9f20c56 Merge pull request #1898 from pipecat-ai/tavus_transport_fixes
Fixing TavusTransport with some TTS services.
2025-05-27 16:33:19 -03:00
Aleix Conchillo Flaqué
e987c4741a pyproject: update deepgram-sdk to 4.1.0 2025-05-27 10:32:48 -07:00
Aleix Conchillo Flaqué
6242278abd Merge pull request #1895 from pipecat-ai/aleix/more-avoid-mutable-default-values
more avoiding mutable default constructor values
2025-05-27 10:08:42 -07:00
Mark Backman
30850a431a Merge pull request #1886 from pipecat-ai/mb/add-otel-llm-output
Add LLM response tracing to OTel tracing
2025-05-27 11:58:57 -04:00
Mark Backman
1e7407c042 Merge pull request #1899 from pipecat-ai/mb/genai-to-standard-messages
fix: Update GoogleLLMContext to_standard_messages to be compatible with google-genai
2025-05-27 11:57:44 -04:00
Mark Backman
6d94f31ff2 Update foundational example 33 to work with google-genai 2025-05-27 11:20:21 -04:00
Mark Backman
ebb3d1cfd3 fix: Update GoogleLLMContext to_standard_messages to be compatible with google-genai 2025-05-27 11:20:21 -04:00
Filipi Fuchter
acce9489d7 Creating the silence based on the chunk size. 2025-05-27 11:26:34 -03:00
Filipi Fuchter
3d442620f9 Removing not used imports. 2025-05-27 10:42:06 -03:00
Martin Schweiger
f1d7eb8565 final touches 2025-05-27 21:29:50 +08:00
Filipi Fuchter
798b935ff6 The remaining audio should not be sent as done. 2025-05-27 10:28:40 -03:00
Filipi Fuchter
3039a1444e Refactoring the TavusVideoService to match the same the behavior of the bot started speaking and bot stopped speaking. 2025-05-27 10:26:41 -03:00
Mark Backman
aa7d15beb3 fix: move LLM call outside tracing try block to prevent double execution 2025-05-26 22:06:31 -04:00
Filipi Fuchter
2b3d2cb342 Fixing the issue when using the TavusVideoService with some TTS services. 2025-05-26 22:55:20 -03:00
Filipi Fuchter
5a58357429 Fixing the issue when using the TavusTransport with some TTS services. 2025-05-26 18:34:52 -03:00
Mark Backman
366add2536 Merge pull request #1878 from pipecat-ai/mb/add-plivo-serializer
Add PlivoFrameSerializer
2025-05-26 11:07:13 -04:00
Mark Backman
e13c9fd42e Add PlivoFrameSerializer 2025-05-26 11:00:01 -04:00
Mark Backman
2a6c01f634 Merge pull request #1885 from pipecat-ai/mb/update-langfuse-example 2025-05-26 10:14:02 -04:00
Mark Backman
bf29722e78 Merge pull request #1884 from pipecat-ai/mb/readme-otel-twilio-telnyx 2025-05-26 10:13:23 -04:00
Mark Backman
db227ad15f Merge pull request #1897 from pipecat-ai/mb/fix-websocket-client-example
Fix mismatched html tag in websocket client example
2025-05-26 09:26:53 -04:00
Mark Backman
514716042b Fix mismatched html tag in websocket client example 2025-05-26 08:25:03 -04:00
Aleix Conchillo Flaqué
7a767e680c more avoiding mutable default constructor values 2025-05-25 21:00:35 -07:00
Martin Schweiger
320b52eb1e Merge branch 'm-ods/assemblyai-universal-streaming' of https://github.com/m-ods/pipecat into m-ods/assemblyai-universal-streaming 2025-05-26 09:13:08 +08:00
Martin Schweiger
428cee75c5 Add User-Agent header to AssemblyAI websocket connection 2025-05-26 09:10:55 +08:00
Martin Schweiger
5479a55b2c Add websockets dependency to assemblyai extra 2025-05-26 09:08:56 +08:00
Mark Backman
d1f2a5d04f Merge pull request #1868 from aristid/google-streaming-tts
Add Google streaming TTS as a base TTS service
2025-05-24 12:42:47 -04:00
aristid
09ba319f3e Merge branch 'main' into google-streaming-tts 2025-05-24 17:16:22 +02:00
ezun-kim
3da711ba8b Fix SSE server connection handling for MCP client
### Summary
This PR improves the MCP (Model Context Protocol) client's SSE (Server-Sent Events) server connection handling by replacing the generic string parameter with a proper `SseServerParameters` class.

### Changes
- **Breaking Change**: Changed `server_params` type from `Union[StdioServerParameters, str]` to `Union[StdioServerParameters, SseServerParameters]`
- Added import for `SseServerParameters` from `mcp.client.session_group`
- Updated SSE client connection to use structured parameters instead of a simple URL string
- Fixed error message to correctly reflect the expected parameter types
- Improved logging by changing info-level log to debug-level for consistency

### Details

#### Before
The SSE client connection only accepted a URL string:
```python
async with self._client(self._server_params) as (read, write):
```

#### After
Now properly unpacks SSE server parameters:
```python
async with self._client(
    url=self._server_params.url,
    headers=self._server_params.headers,
    timeout=self._server_params.timeout,
    sse_read_timeout=self._server_params.sse_read_timeout
) as (read, write):
```

### Benefits
- **Type Safety**: Stronger type checking with dedicated `SseServerParameters` class
- **Extended Configuration**: Support for custom headers (authentication), timeouts, and SSE-specific settings
- **Better Error Messages**: Clear type error messages when incorrect parameters are provided
- **Improved Debugging**: Debug logging of SSE server parameters for troubleshooting

### Migration Guide
Users need to update their SSE server initialization:
```python
# Before
client = MCPClient("https://example.com/sse")

# After
from mcp.client.session_group import SseServerParameters
client = MCPClient(SseServerParameters(
    url="https://example.com/sse",
    headers={"Authorization": "Bearer token"},
    timeout=30,
    sse_read_timeout=60
))
```

### Testing
- [ ] Tested with StdioServerParameters (unchanged behavior)
- [ ] Tested with SseServerParameters with various configurations
- [ ] Verified error handling for invalid parameter types

---

This is a necessary change to support production-ready SSE connections with proper authentication and timeout handling.
2025-05-24 22:35:57 +09:00
Mark Backman
6f524fb816 Add LLM response to OTel tracing 2025-05-24 09:15:39 -04:00
unknown
d3e2a9e5c0 Change default voice and fix formatting 2025-05-24 15:14:39 +02:00
Mark Backman
b4cd7d7941 Langfuse OTel env.example improvements 2025-05-24 08:17:10 -04:00
Mark Backman
cd03b91115 Update README: Add OTel, Add serializers 2025-05-24 07:23:59 -04:00
Aleix Conchillo Flaqué
f86d002ceb Merge pull request #1881 from pipecat-ai/aleix/daily-input-audio-and-video-task-fix
daily input audio and video task fix
2025-05-23 19:39:25 -07:00
Aleix Conchillo Flaqué
940926b5ec TavusVideoService: no need to enable audio/video outputs 2025-05-23 19:29:34 -07:00
Aleix Conchillo Flaqué
85c096df0b DailyTransport: create audio/video input tasks when input flag is enabled 2025-05-23 19:28:18 -07:00
Filipi da Silva Fuchter
76d93522ac Merge pull request #1820 from pipecat-ai/tavus_video_service
Tavus improvements
2025-05-23 23:11:00 -03:00
Filipi Fuchter
31492831cc Updating the changelog and readme to reflect the Tavus changes. 2025-05-23 23:04:04 -03:00
Filipi Fuchter
8221dd594e Creating a Tavus example using the DailyTransport. 2025-05-23 23:03:40 -03:00
Filipi Fuchter
6346ca1a84 Creating a Tavus example using the SmallWebRTCTransport. 2025-05-23 23:03:24 -03:00
Filipi Fuchter
4a3404883f Creating a Tavus example using the new TavusTransport. 2025-05-23 23:03:16 -03:00
Filipi Fuchter
1ebca35313 Queuing the app messages if the SmallWebrtcTransport is not connected yet. 2025-05-23 23:03:04 -03:00
Filipi Fuchter
e0d1381f87 Refactoring the TavusVideoService to allow to work with any transport. 2025-05-23 23:02:49 -03:00
Filipi Fuchter
86e6841569 Creating TavusTransport and TavusTransportClient. 2025-05-23 23:02:37 -03:00
Aleix Conchillo Flaqué
28b7a92a00 Merge pull request #1880 from pipecat-ai/aleix/daily-resize-event-loop-fix
BaseOutputTransport: don't block event loop during image resize
2025-05-23 18:32:00 -07:00
Aleix Conchillo Flaqué
4db5b18694 BaseOutputTransport: don't block event loop during image resize 2025-05-23 18:30:28 -07:00
Aleix Conchillo Flaqué
a628e921c0 Merge pull request #1879 from pipecat-ai/aleix/daily-fix-video-task
DailyTransport: fix video task variable
2025-05-23 17:56:08 -07:00
Aleix Conchillo Flaqué
6ca6ff37c9 DailyTransport: fix video task variable 2025-05-23 17:54:25 -07:00
Aleix Conchillo Flaqué
456db3710a Merge pull request #1828 from pipecat-ai/aleix/daily-use-audio-renderers
DailyTransport: replace virtual speaker and microphones
2025-05-23 13:31:51 -07:00
Aleix Conchillo Flaqué
50f024c6f9 LiveKitTransport: use UserAudioRawFrame instead of InputAudioRawFrame 2025-05-23 11:27:53 -07:00
Aleix Conchillo Flaqué
a4de75a8c0 Merge pull request #1867 from pipecat-ai/aleix/user-bot-latency-log-observer
observers: added UserBotLatencyLogObserver
2025-05-23 09:23:03 -07:00
Aleix Conchillo Flaqué
88e8fcdaca observers: added UserBotLatencyLogObserver 2025-05-23 09:17:53 -07:00
unknown
bfe9952c9a Remove sleep(0), add doc string etc. 2025-05-23 12:11:08 +02:00
Martin Schweiger
7f568e3e7e Merge branch 'main' into m-ods/assemblyai-universal-streaming 2025-05-23 17:39:00 +08:00
Martin Schweiger
9b8800ac1d update stt.py 2025-05-23 17:32:31 +08:00
Martin Schweiger
fd53712567 add models for new streaming service 2025-05-23 17:32:12 +08:00
Aleix Conchillo Flaqué
7f74c2465c SileroVADAnalyzer: improve non-matching sample rate log 2025-05-23 01:47:09 -07:00
Aleix Conchillo Flaqué
30d67a78eb examples(chatbot-audio-recording): use same sample rate to avoid downsampling 2025-05-23 01:47:09 -07:00
Aleix Conchillo Flaqué
c3cfd1f0ce DailyTransport: process audio, video and event callbacks in separate tasks 2025-05-23 01:47:09 -07:00
Aleix Conchillo Flaqué
69ac70eed8 DailyTransport: replace virtual microphone with custom microphone track 2025-05-23 01:47:09 -07:00
Aleix Conchillo Flaqué
fcf49e79cc DailyTransport: use participant audio renderers instead of virtual speaker 2025-05-23 01:47:09 -07:00
Aleix Conchillo Flaqué
8d4894846d pyproject: update to daily-python 0.19.0 2025-05-23 01:47:09 -07:00
Vanessa Pyne
a809b710c5 Merge pull request #1844 from pipecat-ai/vp-docsinreadme
add docs link at top of readme
2025-05-22 21:52:18 -05:00
vipyne
f6289e9db2 add docs link at top of readme 2025-05-22 21:51:29 -05:00
Mark Backman
26b4c4df22 Merge pull request #1870 from pipecat-ai/mb/gemini-2.5-flash-update
Update GeminiMultimodalLiveLLMService to use Gemini 2.5 Flash Native …
2025-05-22 18:19:55 -04:00
Mark Backman
f3a9844295 Merge pull request #1860 from pipecat-ai/mb/organize-otel-demos
Reorganize OpenTelemetry demos, add top-level README
2025-05-22 18:15:20 -04:00
Mark Backman
692821bdae Merge pull request #1873 from pipecat-ai/mb/readme-sarvam
Add SarvamTTSService to README
2025-05-22 18:14:40 -04:00
Mark Backman
ee143d5b3a Update GeminiMultimodalLiveLLMService to use Gemini 2.5 Flash Native Audio Dialog model 2025-05-22 18:13:41 -04:00
Mark Backman
7e178a634a Merge pull request #1871 from pipecat-ai/mb/claude-sonnet-4-update
Update default model for Anthropic to Claude Sonnet 4
2025-05-22 18:12:47 -04:00
Mark Backman
fe88a3d80b Add SarvamTTSService to README 2025-05-22 18:11:11 -04:00
Mark Backman
a196eac290 Merge pull request #1872 from pipecat-ai/mb/add-sarvam-tts
Add SarvamTTSService
2025-05-22 18:02:36 -04:00
Mark Backman
3c819955a2 Add SarvamTTSService 2025-05-22 16:23:08 -04:00
Mark Backman
ca0d7bbbed Update default model for Anthropic to Claude Sonnet 4 2025-05-22 15:13:33 -04:00
Mark Backman
f93bd1e817 Merge pull request #1864 from pipecat-ai/mb/fix-11lab-set-model-voice
Fix: ElevenLabsTTSService, change voice and model
2025-05-22 14:36:24 -04:00
Mark Backman
415bc6ca0a Fix: ElevenLabsTTSService, change voice and model 2025-05-22 14:28:50 -04:00
Mark Backman
8543c8d11d Merge pull request #1869 from pipecat-ai/mb/update-readme-nova-sonic
Add AWS Nova Sonic to README
2025-05-22 14:07:35 -04:00
Mark Backman
bf5ad64575 Add AWS Nova Sonic to README 2025-05-22 14:03:28 -04:00
unknown
d42d02d809 Add Google streaming TTS as a base TTS service. Rename non-streaming service to GoogleHttpTTSService. 2025-05-22 11:21:06 +02:00
Aleix Conchillo Flaqué
0718f79ff2 Merge pull request #1866 from pipecat-ai/aleix/base-observers-are-base-objects
BaseObserver: inherit from BaseObject so we can have events
2025-05-21 16:07:38 -07:00
Aleix Conchillo Flaqué
9bbce225ce BaseObserver: inherit from BaseObject so we can have events 2025-05-21 16:04:44 -07:00
Mark Backman
fb35fd6d71 Merge pull request #1859 from pipecat-ai/mb/otel-attribute-naming
Update OTel attribute names
2025-05-21 12:10:15 -04:00
Mark Backman
b4fd92aed6 Merge pull request #1862 from marctorsoc/clean-links-in-md-text-filter
Add link cleaning in MD text filter
2025-05-21 09:20:27 -04:00
Mark Backman
36931825b3 Merge pull request #1854 from sklinglernv/fix/elevenlab-tts
fix(elevenlabs tts): message parameter naming
2025-05-21 09:17:29 -04:00
marc.torsoc
ca35299dcd add link cleaning and a test for it 2025-05-21 12:08:53 +02:00
Severin Klingler
e74b900914 revert most of the changes except keyword naming fix 2025-05-21 09:24:03 +02:00
Mark Backman
25115668a7 Reorganize OpenTelemetry demos, add top-level README 2025-05-20 23:30:46 -04:00
Mark Backman
fb94db3e64 Update to use GenAI naming 2025-05-20 22:56:02 -04:00
Mark Backman
c4778e770e Merge pull request #1835 from marcklingen/langfuse-tracing
Add examples/open-telemetry-tracing-langfuse
2025-05-20 18:22:55 -04:00
Mark Backman
3860cdf97b Update OTel attribute names 2025-05-20 18:00:46 -04:00
Aleix Conchillo Flaqué
f3aec0c4ac Merge pull request #1829 from pipecat-ai/aleix/pipeline-task-add-observer
PipelineTask: add add_observer()
2025-05-20 13:18:24 -07:00
Aleix Conchillo Flaqué
d333094149 PipelineTask: add add_observer() and remove_observer() 2025-05-20 13:16:06 -07:00
Aleix Conchillo Flaqué
609ff4e66c Merge pull request #1841 from pipecat-ai/aleix/base-text-aggregator-async
make BaseTextAggregator and BaseTextFilter functions async
2025-05-20 13:13:54 -07:00
Aleix Conchillo Flaqué
cbccbcd9e7 BaseTextFilter: make functions async 2025-05-20 13:11:44 -07:00
Aleix Conchillo Flaqué
54b1d7fcc1 BaseTextAggregator: make functions async 2025-05-20 13:11:42 -07:00
Aleix Conchillo Flaqué
54388c0d9b Merge pull request #1850 from pipecat-ai/aleix/transcription-message-user-id
TranscriptionMessage: add user_id field
2025-05-20 13:10:42 -07:00
Aleix Conchillo Flaqué
228c866aaa Merge pull request #1857 from pipecat-ai/aleix/avoid-mutable-default-values
avoid mutable default constructor values
2025-05-20 13:10:24 -07:00
Aleix Conchillo Flaqué
a09bd648af avoid mutable default constructor values 2025-05-20 11:59:28 -07:00
Vanessa Pyne
3e4ae61c75 Merge pull request #1856 from pipecat-ai/vp-mcp-debug
mcp: fix typo in tool call response
2025-05-20 13:59:11 -05:00
vipyne
7655c432c2 mcp: fix typo in tool call response 2025-05-20 11:16:59 -05:00
Aleix Conchillo Flaqué
25dd651757 TranscriptionMessage: add user_id field 2025-05-19 15:47:54 -07:00
Mark Backman
462aecea3e Merge pull request #1839 from pipecat-ai/mb/cartesia-speed
Add support for Cartesia's speed parameter, update clients and APIs, deprecate emotion
2025-05-19 17:57:25 -04:00
Mark Backman
5f37df790b Merge pull request #1848 from pipecat-ai/mb/fix-word-wrangler-transport-params
Fix: Add audio_in_enabled to Word Wrangler TransportParams
2025-05-19 17:52:05 -04:00
Mark Backman
8e4e03541c Update CHANGELOG 2025-05-19 17:51:27 -04:00
Aleix Conchillo Flaqué
c1252fc7eb Merge pull request #1840 from pipecat-ai/aleix/base-object-dont-create-tasks
BaseObject: don't create event handler tasks if none is registered
2025-05-19 14:12:31 -07:00
Mark Backman
ed1077cc9a Fix: Add audio_in_enabled to Word Wrangler TransportParams 2025-05-19 15:53:29 -04:00
Mark Backman
4c761a7b22 Merge pull request #1847 from pipecat-ai/mb/update-otel
Keep span identifiers in attributes only
2025-05-19 14:37:42 -04:00
Mark Backman
9bc3df7803 Keep span identifiers in attributes only 2025-05-19 12:25:13 -04:00
Aleix Conchillo Flaqué
5e5060a6fe BaseObject: don't create event handler tasks if none is registered 2025-05-19 09:24:56 -07:00
Aleix Conchillo Flaqué
2b66eddaa1 Merge pull request #1830 from pipecat-ai/aleix/pipeline-task-frame-events
PipelineTask: add new started/stopped/ended/cancelled events
2025-05-19 08:32:28 -07:00
Mark Backman
916b9d6c6d Add an example for CartesiaHttpTTSService 2025-05-19 11:31:47 -04:00
Mark Backman
bd09ccd608 Update CartesiaHttpTTSService to work with the new cartesia 2.0 client 2025-05-19 11:31:28 -04:00
Mark Backman
682f8e4d45 Bump the cartesia_version for CartesiaTTSService, and cartesia package for CartesiaHttpTTSService 2025-05-19 11:10:03 -04:00
Mark Backman
c9d0af9ee0 Deprecate emotion, add new speed parameter 2025-05-19 09:43:24 -04:00
Severin Klingler
e1299d59bf fix(elevenlabs tts): Fix message paramter naming and make use of contexts to send out TTSStoppedFrames() 2025-05-19 15:22:13 +02:00
Mark Backman
61da6437ea Merge pull request #1834 from pipecat-ai/mb/gemini-live-tokens
Fix: Make LLMTokenUsage more robust
2025-05-19 09:04:07 -04:00
Marc Klingen
798705469b Update README.md 2025-05-18 21:11:20 +02:00
Marc Klingen
459a753de3 add reference to main otel example 2025-05-18 19:56:12 +02:00
Marc Klingen
1092ce70b3 add video of langfuse trace 2025-05-18 19:46:38 +02:00
Marc Klingen
9511c189bd revert original folder 2025-05-18 19:42:13 +02:00
Marc Klingen
66fea9e2ee create new example folder 2025-05-18 19:41:17 +02:00
Marc Klingen
69ae83516e use http exporter 2025-05-18 19:11:06 +02:00
Mark Backman
144ea36c81 Fix: Make LLMTokenUsage more robust 2025-05-18 07:41:16 -04:00
Mark Backman
7a8ab9a900 Merge pull request #1672 from golbin/main
Use "use_original_timestamps" only for sonic-2 model
2025-05-18 07:24:58 -04:00
Jin Kim
c4b35055b4 Update CHANGELOG.md 2025-05-18 16:54:29 +09:00
Jin Kim
a4c04e7c17 Opt out Sonic models from use_original_timestamps 2025-05-18 16:52:37 +09:00
Jin Kim
a6f7e7fc30 Merge branch 'pipecat-ai:main' into main 2025-05-18 16:48:24 +09:00
Aleix Conchillo Flaqué
d5ebc883b3 PipelineTask: add new started/stopped/ended/cancelled events 2025-05-17 22:46:22 -07:00
Mark Backman
deb43df0a4 Merge pull request #1824 from pipecat-ai/mb/gemini-live-transcribe-user-audio
Update GeminiMultimodalLiveLLMService to use Gemini's user transcription
2025-05-16 22:51:04 -04:00
Mattie Ruth
88e472b3f1 Update Modal Readme (#1825) 2025-05-16 17:40:57 -04:00
Mark Backman
f59fb8167d Merge pull request #1784 from thsunkid/thu/handle-transcript-gpt4o-audio
Handle audio transcript from gpt-4o-audio and clean up logs
2025-05-16 13:20:16 -04:00
Mark Backman
fac6f526f7 Add comments and docstrings 2025-05-16 10:54:50 -04:00
Mark Backman
2f78d74ce6 Change Gemini Live to use Gemini provided usage metrics 2025-05-16 10:53:01 -04:00
Mark Backman
d3942dda52 Gemini Live to transcribe user audio 2025-05-16 10:53:01 -04:00
Mark Backman
c00e9a8d3a Merge pull request #1819 from kaikato/lmnt-model-langs
LmntTTSService: add model param and additional languages
2025-05-16 08:49:55 -04:00
kaikato
c3b95767f3 LmntTTSService: add model param and additional languages 2025-05-16 04:24:57 +00:00
Mark Backman
90f27a3090 Merge pull request #1816 from pipecat-ai/mb/add-minimax-tts
Add MiniMax TTS
2025-05-15 18:05:13 -04:00
Mark Backman
b6f09defc9 Add MiniMax TTS 2025-05-15 18:02:29 -04:00
Aleix Conchillo Flaqué
172813bcfb Merge pull request #1815 from pipecat-ai/aleix/remove-silerovad-processor
remove SileroVAD() frame processor
2025-05-15 13:44:44 -07:00
Aleix Conchillo Flaqué
95c25efab7 remove SileroVAD() frame processor 2025-05-15 11:55:20 -07:00
Aleix Conchillo Flaqué
a51af35024 Merge pull request #1814 from pipecat-ai/aleix/examples-dependabot-05142025
examples: updates for dependabot 05/14/2025
2025-05-15 11:38:45 -07:00
Mark Backman
119fd5ba7d Merge pull request #1025 from fatwang2/main
added hailuo tts service
2025-05-15 14:29:24 -04:00
Aleix Conchillo Flaqué
0718a812bd examples: updates for dependabot 05/14/2025 2025-05-14 22:51:08 -07:00
Mark Backman
3814501b48 Merge pull request #1811 from pipecat-ai/mb/dont-require-tracing-dep
Fix: Resolve an issue where tracing imports were required
2025-05-14 12:35:47 -04:00
Mark Backman
7a5205dbda Fix: Resolve an issue where tracing imports were required 2025-05-14 12:29:08 -04:00
Thu Nguyen
15a5028d23 Revert log changes 2025-05-14 22:28:25 +08:00
Thu Nguyen
fee2648ac0 Handle audio transcript from gpt-4o-audio and clean up logs 2025-05-14 13:02:22 +07:00
Varun Singh
04c02c9a20 Merge pull request #1810 from pipecat-ai/vr000m-receiving-custom-sip-headers
added handling for sipHeaders
2025-05-13 23:02:14 -07:00
Varun Singh
0ff7195a83 Update README.md
updating docs
2025-05-13 19:08:43 -04:00
Varun Singh
3b91aa013a added handling for sipHeaders 2025-05-13 16:00:05 -07:00
Mark Backman
50f6235edb Add support for OpenTelemetry tracing (#1729)
* Also added TurnTrackingObserver, TurnTraceObserver, foundational 29, open-telemetry-example
2025-05-13 17:18:11 -04:00
Aleix Conchillo Flaqué
6f4d94f91b Merge pull request #1800 from pipecat-ai/aleix/frame-processors-setup
introduce frame processors setup
2025-05-13 13:18:06 -07:00
Aleix Conchillo Flaqué
83a4c7d443 RTVIProcessor: remove unused code 2025-05-13 11:26:37 -07:00
Aleix Conchillo Flaqué
8171fec925 SmallWebRTCConnection: complain if av package not found 2025-05-13 11:26:37 -07:00
Aleix Conchillo Flaqué
175f352ea7 add FrameProcessor.setup() to setup processors before StartFrame 2025-05-13 11:26:35 -07:00
Filipi da Silva Fuchter
5290161ac4 Merge pull request #1746 from pipecat-ai/simple_chatbot-react-native
Simple chatbot: React Native client
2025-05-13 10:48:09 -03:00
Filipi Fuchter
8762019ed7 Not setting the local audio level when the user stopped speaking. 2025-05-13 10:46:30 -03:00
Filipi Fuchter
61a59fa158 Fixing useNavigation typescript warning. 2025-05-13 10:36:39 -03:00
Filipi Fuchter
55eea20c8e Renaming expo environment variable 2025-05-13 10:32:27 -03:00
kompfner
9a621f0c54 Merge pull request #1805 from pipecat-ai/pk/aws-nova-sonic-aggregate-user-transcription-text
AWS Nova Sonic service - aggregate user transcription text; it was fr…
2025-05-13 09:13:58 -04:00
Paul Kompfner
55fc24e933 AWS Nova Sonic service - aggregate user transcription text; it was fragmented across many conversation history messages before 2025-05-13 09:13:28 -04:00
Filipi da Silva Fuchter
b14608f09b Merge pull request #1799 from pipecat-ai/daily_audio_source
Using audio source for capturing Daily's participant audio
2025-05-13 08:15:10 -03:00
Mark Backman
4a25c57337 Merge pull request #1806 from pipecat-ai/aleix/run-test-observers
tests: allow passing observers to run_test()
2025-05-12 22:10:44 -04:00
Aleix Conchillo Flaqué
f800e35ccb tests: allow passing observers to run_test() 2025-05-12 17:53:02 -07:00
Vanessa Pyne
12d49a9b9d Merge pull request #1801 from pipecat-ai/vp-fix-typo
update examples
2025-05-12 15:33:56 -05:00
vipyne
b25b251a44 update examples 2025-05-12 14:07:17 -05:00
Mattie Ruth
64b2a75a94 Update Modal App: (#1755)
* Update Modal App:

Updated Modal App to include:

1. Latest Modal API usage
2. Ability to launch different Pipecat pipelines, much like the
   simple chatbot example
3. Ability to choose which pipeline is launched via the
   /connect endpoint
4. Added a pipeline option for connecting to a self-hosted LLM
   on Modal
5. Improved READMEs
6. Added a web client for interacting with the Modal deployment

tmp

* Update README
2025-05-12 12:45:43 -05:00
Aleix Conchillo Flaqué
b33a60f3a5 Merge pull request #1793 from pipecat-ai/khk/deepgram-async-fix
Fix Deepgram TTS streaming
2025-05-12 09:59:46 -07:00
Filipi Fuchter
d22dbb1a6d Fixing ruff format. 2025-05-12 10:36:21 -03:00
Filipi Fuchter
983199a6cd New example capturing the audio from the participant using the custom audio source. 2025-05-12 10:18:43 -03:00
Filipi Fuchter
133d7ee33a Fixing the default audio source for capture_participant_audio 2025-05-12 10:16:32 -03:00
Mark Backman
0bd888afc7 Merge pull request #1796 from nikp06/patch-1
Wrong deprecation warning when importing ai_services.py
2025-05-12 09:12:48 -04:00
nikp06
537bd1c58d Update ai_services.py
fix: correct deprecation warning format in ai_services module
2025-05-12 12:01:13 +02:00
Kwindla Hultman Kramer
5ef519fe2c Fix Deepgram TTS to use stream_raw() 2025-05-11 15:40:31 -07:00
Mark Backman
20498fb47f Merge pull request #1790 from AngeloGiacco/angelo/fix-api-key
[elevenlabs tts ] fix api key
2025-05-10 19:16:27 -04:00
Angelo Giacco
b57dfb3b5d fix lint 2025-05-10 16:36:26 +01:00
Angelo Giacco
0355ed4aa1 move api key to ws header 2025-05-10 16:34:01 +01:00
Angelo Giacco
1e76cc7bdc fix: elevenlabs api key 2025-05-10 16:09:20 +01:00
Vanessa Pyne
18c0374126 Merge pull request #1785 from pipecat-ai/vp-small-filenmae-change
39-aws-nova-sonic.py -> 40-aws-nova-sonic.py
2025-05-09 12:19:09 -05:00
Aleix Conchillo Flaqué
7072fba7e7 Merge pull request #1780 from pipecat-ai/aleix/deprecate-google-generativeai
GoogleLLMService: deprecate google-generativeai
2025-05-09 09:18:30 -07:00
Aleix Conchillo Flaqué
3d702a5c39 minor examples cleanup 2025-05-09 09:16:10 -07:00
Aleix Conchillo Flaqué
f31efa42c9 GoogleLLMService: deprecate google-generativeai 2025-05-09 09:14:43 -07:00
vipyne
74b369ff20 39-aws-nova-sonic.py -> 40-aws-nova-sonic.py 2025-05-09 08:30:59 -05:00
Filipi Fuchter
46eed0a59a Bumping to use the latest version of @pipecat-ai/react-native-daily-transport, and removing code not needed. 2025-05-08 18:18:00 -03:00
kompfner
9643296e29 Merge pull request #1779 from pipecat-ai/pk/aws-nova-sonic-missing-params-export
Add missing `Params` export to AWS Nova Sonic module
2025-05-08 16:04:38 -04:00
Paul Kompfner
c83c5b5a34 Add missing Params export to AWS Nova Sonic module 2025-05-08 15:23:25 -04:00
Filipi Fuchter
277e2d7fc0 Merge branch 'main' into simple_chatbot-react-native 2025-05-08 09:03:16 -03:00
Filipi Fuchter
56ca7360ae Fixing versions 2025-05-05 19:11:59 -03:00
Filipi Fuchter
d5ab3251f0 Bumping the dependencies, updating readme, adding .gitignore. 2025-05-05 18:43:04 -03:00
Filipi Fuchter
915c284420 Fixing readme 2025-05-05 18:32:04 -03:00
Filipi Fuchter
40154824e8 Creating a RN example for simple-chatbot 2025-05-05 18:17:39 -03:00
Jin Kim
cf2f249f8a Use "use_original_timestamps" only for sonic-2 model 2025-04-27 19:33:14 +09:00
fatwang2
8cda4512ad Merge branch 'pipecat-ai:main' into main 2025-02-06 10:50:25 +08:00
fatwang2
fc90bdc638 changed to HailuoHttpTTSService 2025-01-19 09:43:48 +08:00
fatwang2
5a88165a26 Merge branch 'pipecat-ai:main' into main 2025-01-19 09:40:08 +08:00
fatwang2
3466842cd4 add hailuo tts service 2025-01-17 12:46:05 +08:00
617 changed files with 44103 additions and 8899 deletions

View File

@@ -6,11 +6,13 @@ on:
- main
paths:
- "examples/simple-chatbot/client/android/**"
- "examples/p2p-webrtc/video-transform/client/android/**"
pull_request:
branches:
- "**"
paths:
- "examples/simple-chatbot/client/android/**"
- "examples/p2p-webrtc/video-transform/client/android/**"
workflow_dispatch:
inputs:
sdk_git_ref:
@@ -23,7 +25,7 @@ concurrency:
jobs:
sdk:
name: "Simple chatbot demo"
name: "Demo apps"
runs-on: ubuntu-latest
steps:
- name: Checkout repo
@@ -37,12 +39,22 @@ jobs:
distribution: 'temurin'
java-version: '17'
- name: Build demo app
- name: "Example app: Simple Chatbot"
working-directory: examples/simple-chatbot/client/android
run: ./gradlew :simple-chatbot-client:assembleDebug
- name: Upload demo APK
- name: Upload Simple Chatbot APK
uses: actions/upload-artifact@v4
with:
name: Simple Chatbot Android Client
path: examples/simple-chatbot/client/android/simple-chatbot-client/build/outputs/apk/debug/simple-chatbot-client-debug.apk
- name: "Example app: Small WebRTC Client"
working-directory: examples/p2p-webrtc/video-transform/client/android
run: ./gradlew :small-webrtc-client:assembleDebug
- name: Upload Small WebRTC APK
uses: actions/upload-artifact@v4
with:
name: Small WebRTC Android Client
path: examples/p2p-webrtc/video-transform/client/android/small-webrtc-client/build/outputs/apk/debug/small-webrtc-client-debug.apk

View File

@@ -5,6 +5,501 @@ All notable changes to **Pipecat** will be documented in this file.
The format is based on [Keep a Changelog](https://keepachangelog.com/en/1.0.0/),
and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html).
## [0.0.72] - 2025-06-26
### Added
- Added logging and improved error handling to help diagnose and prevent potential
Pipeline freezes.
- Added `WatchdogQueue`, `WatchdogPriorityQueue`, `WatchdogEvent` and
`WatchdogAsyncIterator`. These helper utilities reset watchdog timers
appropriately before they expire. When watchdog timers are disabled, the
utilities behave as standard counterparts without side effects.
- Introduce task watchdog timers. Watchdog timers are used to detect if a
Pipecat task is taking longer than expected (by default 5 seconds). Watchdog
timers are disabled by default and can be enabled globally by passing
`enable_watchdog_timers` argument to `PipelineTask` constructor. It is
possible to change the default watchdog timer timeout by using the
`watchdog_timeout` argument. You can also log how long it takes to reset the
watchdog timers which is done with the `enable_watchdog_logging`. You can
control all these settings per each frame processor or even per task. That is,
you can set `enable_watchdog_timers`, `enable_watchdog_logging` and
`watchdog_timeout` when creating any frame processor through their constructor
arguments or when you create a task with `FrameProcessor.create_task()`. Note
that watchdog timers only work with Pipecat tasks and will not work if you use
`asycio.create_task()` or similar.
- Added `lexicon_names` parameter to `AWSPollyTTSService.InputParams`.
- Added reconnection logic and audio buffer management to `GladiaSTTService`.
- The `TurnTrackingObserver` now ends a turn upon observing an `EndFrame` or
`CancelFrame`.
- Added Polish support to `AWSTranscribeSTTService`.
- Added new frames `FrameProcessorPauseFrame` and `FrameProcessorResumeFrame`
which allow pausing and resuming frame processing for a given frame
processor. These are control frames, so they are ordered. Pausing frame
processor will keep old frames in the internal queues until resume takes
place. Frames being pushed while a frame processor is paused will be pushed to
the queues. When frame processing is resumed all queued frames will be
processed in order. Also added `FrameProcessorPauseUrgentFrame` and
`FrameProcessorResumeUrgentFrame` which are system frames and therefore they
have high priority.
- Added a property called `has_function_calls_in_progress` in
`LLMAssistantContextAggregator` that exposes whether a function call is in
progress.
- Added `SambaNovaLLMService` which provides llm api integration with an
OpenAI-compatible interface.
- Added `SambaNovaTTSService` which provides speech-to-text functionality using
SambaNovas's (whisper) API.
- Add fundational examples for function calling and transcription
`14s-function-calling-sambanova.py`, `13g-sambanova-transcription.py`
### Changed
- `HeartbeatFrame`s are now control frames. This will make it easier to detect
pipeline freezes. Previously, heartbeat frames were system frames which meant
they were not get queued with other frames, making it difficult to detect
pipeline stalls.
- Updated `OpenAIRealtimeBetaLLMService` to accept `language` in the
`InputAudioTranscription` class for all models.
- Updated the default model for `OpenAIRealtimeBetaLLMService` to
`gpt-4o-realtime-preview-2025-06-03`.
- The `PipelineParams` arg `allow_interruptions` now defaults to `True`.
- `TavusTransport` and `TavusVideoService` now send audio to Tavus using WebRTC
audio tracks instead of `app-messages` over WebSocket. This should improve the
overall audio quality.
- Upgraded `daily-python` to 0.19.3.
### Fixed
- Fixed an issue that would cause heartbeat frames to be sent before processors
were started.
- Fixed an event loop blocking issue when using `SentryMetrics`.
- Fixed an issue in `FastAPIWebsocketClient` to ensure proper disconnection
when the websocket is already closed.
- Fixed an issue where the `UserStoppedSpeakingFrame` was not received if the
transport was not receiving new audio frames.
- Fixed an edge case where if the user interrupted the bot but no new aggregation
was received, the bot would not resume speaking.
- Fixed an issue with `TelnyxFrameSerializer` where it would throw an exception
when the user hung up the call.
- Fixed an issue with `ElevenLabsTTSService` where the context was not being
closed.
- Fixed function calling in `AWSNovaSonicLLMService`.
- Fixed an issue that would cause multiple `PipelineTask.on_idle_timeout`
events to be triggered repeatedly.
- Fixed an issue that was causing user and bot speech to not be synchronized
during recordings.
- Fixed an issue where voice settings weren't applied to ElevenLabsTTSService.
- Fixed an issue with `GroqTTSService` where it was not properly parsing the
WAV file header.
- Fixed an issue with `GoogleSTTService` where it was constantly reconnecting
before starting to receive audio from the user.
- Fixed an issue where `GoogleLLMService`'s TTFB value was incorrect.
### Deprecated
- `AudioBufferProcessor` parameter `user_continuos_stream` is deprecated.
### Other
- Rename `14e-function-calling-gemini.py` to `14e-function-calling-google.py`.
## [0.0.71] - 2025-06-10
### Added
- Adds a parameter called `additional_span_attributes` to PipelineTask that
lets you add any additional attributes you'd like to the conversation span.
### Fixed
- Fixed an issue with `CartesiaSTTService` initialization.
## [0.0.70] - 2025-06-10
### Added
- Added `ExotelFrameSerializer` to handle telephony calls via Exotel.
- Added the option `informal` to `TranslationConfig` on Gladia config.
Allowing to force informal language forms when available.
- Added `CartesiaSTTService` which is a websocket based implementation to
transcribe audio. Added a foundational example in
`13f-cartesia-transcription.py`
- Added an `websocket` example, showing how to use the new Pipecat client
`WebsocketTransport` to connect with Pipecat `FastAPIWebsocketTransport` or
`WebsocketServerTransport`.
- Added language support to `RimeHttpTTSService`. Extended languages to include
German and French for both `RimeTTSService` and `RimeHttpTTSService`.
### Changed
- Upgraded `daily-python` to 0.19.2.
- Make `PipelineTask.add_observer()` synchronous. This allows callers to call it
before doing the work of running the `PipelineTask` (i.e. without invoking
`PipelineTask.set_event_loop()` first).
- Pipecat 0.0.69 forced `uvloop` event loop on Linux on macOS. Unfortunately,
this is causing issue in some systems. So, `uvloop` is not enabled by default
anymore. If you want to use `uvloop` you can just set the `asyncio` event
policy before starting your agent with:
```python
asyncio.set_event_loop_policy(uvloop.EventLoopPolicy())
```
### Fixed
- Fixed an issue with various TTS services that would cause audio glitches at
the start of every bot turn.
- Fixed an `ElevenLabsTTSService` issue where a context warning was printed
when pushing a `TTSSpeakFrame`.
- Fixed an `AssemblyAISTTService` issue that could cause unexpected behavior
when yielding empty `Frame()`s.
- Fixed an issue where `OutputAudioRawFrame.transport_destination` was being
reset to `None` instead of retaining its intended value before sending the
audio frame to `write_audio_frame`.
- Fixed a typo in Livekit transport that prevented initialization.
## [0.0.69] - 2025-06-02 "AI Engineer World's Fair release" ✨
### Added
- Added a new frame `FunctionCallsStartedFrame`. This frame is pushed both
upstream and downstream from the LLM service to indicate that one or more
function calls are going to be executed.
- Added LLM services `on_function_calls_started` event. This event will be
triggered when the LLM service receives function calls from the model and is
going to start executing them.
- Function calls can now be executed sequentially (in the order received in the
completion) by passing `run_in_parallel=False` when creating your LLM
service. By default, if the LLM completion returns 2 or more function calls
they run concurrently. In both cases, concurrently and sequentially, a new LLM
completion will run when the last function call finishes.
- Added OpenTelemetry tracing for `GeminiMultimodalLiveLLMService` and
`OpenAIRealtimeBetaLLMService`.
- Added initial support for interruption strategies, which determine if the user
should interrupt the bot while the bot is speaking. Interruption strategies
can be based on factors such as audio volume or the number of words spoken by
the user. These can be specified via the new `interruption_strategies` field
in `PipelineParams`. A new `MinWordsInterruptionStrategy` strategy has been
introduced which triggers an interruption if the user has spoken a minimum
number of words. If no interruption strategies are specified, the normal
interruption behavior applies. If multiple strategies are provided, the first
one that evaluates to true will trigger the interruption.
- `BaseInputTransport` now handles `StopFrame`. When a `StopFrame` is received
the transport will pause sending frames downstream until a new `StartFrame` is
received. This allows the transport to be reused (keeping the same connection)
in a different pipeline.
- Updated AssemblyAI STT service to support their latest streaming
speech-to-text model with improved transcription latency and endpointing.
- You can now access STT service results through the new
`TranscriptionFrame.result` and `InterimTranscriptionFrame.result` field. This
is useful in case you use some specific settings for the STT and you want to
access the STT results.
- The examples runner is now public from the `pipecat.examples` package. This
allows everyone to build their own examples and run them easily.
- It is now possible to push `OutputDTMFFrame` or `OutputDTMFUrgentFrame` with
`DailyTransport`. This will be sent properly if a Daily dial-out connection
has been established.
- Added `OutputDTMFUrgentFrame` to send a DTMF keypress quickly. The previous
`OutputDTMFFrame` queues the keypress with the rest of data frames.
- Added `DTMFAggregator`, which aggregates keypad presses into
`TranscriptionFrame`s. Aggregation occurs after a timeout, termination key
press, or user interruption. You can specify the prefix of the
`TranscriptionFrame`.
- Added new functions `DailyTransport.start_transcription()` and
`DailyTransport.stop_transcription()` to be able to start and stop Daily
transcription dynamically (maybe with different settings).
### Changed
- Reverted the default model for `GeminiMultimodalLiveLLMService` back to
`models/gemini-2.0-flash-live-001`.
`gemini-2.5-flash-preview-native-audio-dialog` has inconsistent performance.
You can opt in to using this model by setting the `model` arg.
- Function calls are now cancelled by default if there's an interruption. To
disable this behavior you can set `cancel_on_interruption=False` when
registering the function call. Since function calls are executed as tasks you
can tell if a function call has been cancelled by catching the
`asyncio.CancelledError` exception (and don't forget to raise it again!).
- Updated OpenTelemetry tracing attribute `metrics.ttfb_ms` to `metrics.ttfb`.
The attribute reports TTFB in seconds.
### Deprecated
- `DailyTransport.send_dtmf()` is deprecated, push an `OutputDTMFFrame` or an
`OutputDTMFUrgentFrame` instead.
### Fixed
- Fixed an issue with `ElevenLabsTTSService` where long responses would
continue generating output even after an interruption.
- Fixed an issue with the `OpenAILLMContext` where non-Roman characters were
being incorrectly encoded as Unicode escape sequences. This was a logging
issue and did not impact the actual conversation.
- In `AWSBedrockLLMService`, worked around a possible bug in AWS Bedrock where
a `toolConfig` is required if there has been previous tool use in the
messages array. This workaround includes a no_op factory function call is
used to satisfy the requirement.
- Fixed `WebsocketClientTransport` to use `FrameProcessorSetup.task_manager`
instead of `StartFrame.task_manager`.
### Performance
- Use `uvloop` as the new event loop on Linux and macOS systems.
## [0.0.68] - 2025-05-28
### Added
- Added `GoogleHttpTTSService` which uses Google's HTTP TTS API.
- Added `TavusTransport`, a new transport implementation compatible with any
Pipecat pipeline. When using the `TavusTransport`the Pipecat bot will
connect in the same room as the Tavus Avatar and the user.
- Added `PlivoFrameSerializer` to support Plivo calls. A full running example
has also been added to `examples/plivo-chatbot`.
- Added `UserBotLatencyLogObserver`. This is an observer that logs the latency
between when the user stops speaking and when the bot starts speaking. This
gives you an initial idea on how quickly the AI services respond.
- Added `SarvamTTSService`, which implements Sarvam AI's TTS API:
https://docs.sarvam.ai/api-reference-docs/text-to-speech/convert.
- Added `PipelineTask.add_observer()` and `PipelineTask.remove_observer()` to
allow mangaging observers at runtime. This is useful for cases where the task
is passed around to other code components that might want to observe the
pipeline dynamically.
- Added `user_id` field to `TranscriptionMessage`. This allows identifying the
user in a multi-user scenario. Note that this requires that
`TranscriptionFrame` has the `user_id` properly set.
- Added new `PipelineTask` event handlers `on_pipeline_started`,
`on_pipeline_stopped`, `on_pipeline_ended` and `on_pipeline_cancelled`, which
correspond to the `StartFrame`, `StopFrame`, `EndFrame` and `CancelFrame`
respectively.
- Added additional languages to `LmntTTSService`. Languages include: `hi`,
`id`, `it`, `ja`, `nl`, `pl`, `ru`, `sv`, `th`, `tr`, `uk`, `vi`.
- Added a `model` parameter to the `LmntTTSService` constructor, allowing
switching between LMNT models.
- Added `MiniMaxHttpTTSService`, which implements MiniMax's T2A API for TTS.
Learn more: https://www.minimax.io/platform_overview
- A new function `FrameProcessor.setup()` has been added to allow setting up
frame processors before receiving a `StartFrame`. This is what's happening
internally: `FrameProcessor.setup()` is called, `StartFrame` is pushed from
the beginning of the pipeline, your regular pipeline operations, `EndFrame`
or `CancelFrame` are pushed from the beginning of the pipeline and finally
`FrameProcessor.cleanup()` is called.
- Added support for OpenTelemetry tracing in Pipecat. This initial
implementation includes:
- A `setup_tracing` method where you can specify your OpenTelemetry exporter
- Service decorators for STT (`@traced_stt`), LLM (`@traced_llm`), and TTS
(`@traced_tts`) which trace the execution and collect properties and
metrics (TTFB, token usage, character counts, etc.)
- Class decorators that provide execution tracking; these are generic and can
be used for service tracking as needed
- Spans that help track traces on a per conversations and turn basis:
```
conversation-uuid
├── turn-1
│ ├── stt_deepgramsttservice
│ ├── llm_openaillmservice
│ └── tts_cartesiattsservice
...
└── turn-n
└── ...
```
By default, Pipecat has implemented service decorators to trace execution of
STT, LLM, and TTS services. You can enable tracing by setting
`enable_tracing` to `True` in the PipelineTask.
- Added `TurnTrackingObserver`, which tracks the start and end of a user/bot
turn pair and emits events `on_turn_started` and `on_turn_stopped`
corresponding to the start and end of a turn, respectively.
- Allow passing observers to `run_test()` while running unit tests.
### Changed
- Upgraded `daily-python` to 0.19.1.
- ⚠️ Updated `SmallWebRTCTransport` to align with how other transports handle
`on_client_disconnected`. Now, when the connection is closed and no reconnection
is attempted, `on_client_disconnected` is called instead of `on_client_close`. The
`on_client_close` callback is no longer used, use `on_client_disconnected` instead.
- Check if `PipelineTask` has already been cancelled.
- Don't raise an exception if event handler is not registered.
- Upgraded `deepgram-sdk` to 4.1.0.
- Updated `GoogleTTSService` to use Google's streaming TTS API. The default
voice also updated to `en-US-Chirp3-HD-Charon`.
- ⚠️ Refactored the `TavusVideoService`, so it acts like a proxy, sending audio
to Tavus and receiving both audio and video. This will make
`TavusVideoService` usable with any Pipecat pipeline and with any transport.
This is a **breaking change**, check the
`examples/foundational/21a-tavus-layer-small-webrtc.py` to see how to use it.
- `DailyTransport` now uses custom microphone audio tracks instead of virtual
microphones. Now, multiple Daily transports can be used in the same process.
- `DailyTransport` now captures audio from individual participants instead of
the whole room. This allows identifying audio frames per participant.
- Updated the default model for `AnthropicLLMService` to
`claude-sonnet-4-20250514`.
- Updated the default model for `GeminiMultimodalLiveLLMService` to
`models/gemini-2.5-flash-preview-native-audio-dialog`.
- `BaseTextFilter` methods `filter()`, `update_settings()`,
`handle_interruption()` and `reset_interruption()` are now async.
- `BaseTextAggregator` methods `aggregate()`, `handle_interruption()` and
`reset()` are now async.
- The API version for `CartesiaTTSService` and `CartesiaHttpTTSService` has
been updated. Also, the `cartesia` dependency has been updated to 2.x.
- `CartesiaTTSService` and `CartesiaHttpTTSService` now support Cartesia's new
`speed` parameter which accepts values of `slow`, `normal`, and `fast`.
- `GeminiMultimodalLiveLLMService` now uses the user transcription and usage
metrics provided by Gemini Live.
- `GoogleLLMService` has been updated to use `google-genai` instead of the
deprecated `google-generativeai`.
### Deprecated
- In `CartesiaTTSService` and `CartesiaHttpTTSService`, `emotion` has been
deprecated by Cartesia. Pipecat is following suit and deprecating `emotion`
as well.
### Removed
- Since `GeminiMultimodalLiveLLMService` now transcribes it's own audio, the
`transcribe_user_audio` arg has been removed. Audio is now transcribed
automatically.
- Removed `SileroVAD` frame processor, just use `SileroVADAnalyzer`
instead. Also removed, `07a-interruptible-vad.py` example.
### Fixed
- Fixed a `DailyTransport` issue that was not allow capturing video frames if
framerate was greater than zero.
- Fixed a `DeegramSTTService` connection issue when the user provided their own
`LiveOptions`.
- Fixed a `DailyTransport` issue that would cause images needing resize to block
the event loop.
- Fixed an issue with `ElevenLabsTTSService` where changing the model or voice
while the service is running wasn't working.
- Fixed an issue that would cause multiple instances of the same class to behave
incorrectly if any of the given constructor arguments defaulted to a mutable
value (e.g. lists, dictionaries, objects).
- Fixed an issue with `CartesiaTTSService` where `TTSTextFrame` messages weren't
being emitted when the model was set to `sonic`. This resulted in the
assistant context not being updated with assistant messages.
### Performance
- `DailyTransport`: process audio, video and events in separate tasks.
- Don't create event handler tasks if no user event handlers have been
registered.
### Other
- It is now possible to run all (or most) foundational example with multiple
transports. By default, they run with P2P (Peer-To-Peer) WebRTC so you can try
everything locally. You can also run them with Daily or even with a Twilio
phone number.
- Added foundation examples `07y-interruptible-minimax.py` and
`07z-interruptible-sarvam.py`to show how to use the `MiniMaxHttpTTSService`
and `SarvamTTSService`, respectively.
- Added an `open-telemetry-tracing` example, showing how to setup tracing. The
example also includes Jaeger as an open source OpenTelemetry client to review
traces from the example runs.
- Added foundational example `29-turn-tracking-observer.py` to show how to use
the `TurnTrackingObserver`.
## [0.0.67] - 2025-05-07
### Added

View File

@@ -41,36 +41,76 @@ We use Ruff for code linting and formatting. Please ensure your code passes all
We follow Google-style docstrings with these specific conventions:
- Class docstrings should fully document all parameters used in `__init__`
- We don't require separate docstrings for `__init__` methods when parameters are documented in the class docstring
- Property methods should have docstrings explaining their purpose and return value
**Regular Classes:**
Example of correctly documented class:
- Class docstring describes the class purpose and documents all `__init__` parameters in an `Args:` section
- No separate `__init__` docstring needed
- All public methods must have docstrings with `Args:` and `Returns:` sections as appropriate
**Dataclasses:**
- Class docstring describes the purpose and documents all fields in a `Parameters:` section
- No `__init__` docstring (auto-generated)
**Properties:**
- Must have docstrings with `Returns:` section
**Abstract Methods:**
- Must have docstrings explaining what subclasses should implement
#### Examples:
```python
class MyClass:
"""Class description.
Additional details about the class.
# Regular class
class MyService(BaseService):
"""Description of what the service does.
Args:
param1: Description of first parameter.
param2: Description of second parameter.
param1: Description of param1.
param2: Description of param2. Defaults to True.
**kwargs: Additional arguments passed to parent.
"""
def __init__(self, param1, param2):
# No docstring required here as parameters are documented above
self.param1 = param1
self.param2 = param2
def __init__(self, param1: str, param2: bool = True, **kwargs):
# No docstring - parameters documented above
super().__init__(**kwargs)
@property
def some_property(self) -> str:
"""Get the formatted property value.
def sample_rate(self) -> int:
"""Get the current sample rate.
Returns:
A string representation of the property.
The sample rate in Hz.
"""
return f"Property: {self.param1}"
return self._sample_rate
async def process_data(self, data: str) -> bool:
"""Process the provided data.
Args:
data: The data to process.
Returns:
True if processing succeeded.
"""
pass
# Dataclass
@dataclass
class ConfigParams:
"""Configuration parameters for the service.
Parameters:
host: The host address.
port: The port number. Defaults to 8080.
timeout: Connection timeout in seconds.
"""
host: str
port: int = 8080
timeout: float = 30.0
```
# Contributor Covenant Code of Conduct

4
MANIFEST.in Normal file
View File

@@ -0,0 +1,4 @@
prune docs
prune examples
prune scripts
prune tests

View File

@@ -8,6 +8,8 @@
**Pipecat** is an open-source Python framework for building real-time voice and multimodal conversational agents. Orchestrate audio and video, AI services, different transports, and conversation pipelines effortlessly—so you can focus on what makes your agent unique.
> Want to dive right in? [Install Pipecat](https://docs.pipecat.ai/getting-started/installation) then try the [quickstart](https://docs.pipecat.ai/getting-started/quickstart).
## 🚀 What You Can Build
- **Voice Assistants** natural, streaming conversations with AI
@@ -49,18 +51,19 @@ You can connect to Pipecat from any platform using our official SDKs:
## 🧩 Available services
| Category | Services |
|---------------------|-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------|
| Speech-to-Text | [AssemblyAI](https://docs.pipecat.ai/server/services/stt/assemblyai), [AWS](https://docs.pipecat.ai/server/services/stt/aws), [Azure](https://docs.pipecat.ai/server/services/stt/azure), [Deepgram](https://docs.pipecat.ai/server/services/stt/deepgram), [Fal Wizper](https://docs.pipecat.ai/server/services/stt/fal), [Gladia](https://docs.pipecat.ai/server/services/stt/gladia), [Google](https://docs.pipecat.ai/server/services/stt/google), [Groq (Whisper)](https://docs.pipecat.ai/server/services/stt/groq), [OpenAI (Whisper)](https://docs.pipecat.ai/server/services/stt/openai), [Parakeet (NVIDIA)](https://docs.pipecat.ai/server/services/stt/parakeet), [Ultravox](https://docs.pipecat.ai/server/services/stt/ultravox), [Whisper](https://docs.pipecat.ai/server/services/stt/whisper) |
| LLMs | [Anthropic](https://docs.pipecat.ai/server/services/llm/anthropic), [AWS](https://docs.pipecat.ai/server/services/llm/aws), [Azure](https://docs.pipecat.ai/server/services/llm/azure), [Cerebras](https://docs.pipecat.ai/server/services/llm/cerebras), [DeepSeek](https://docs.pipecat.ai/server/services/llm/deepseek), [Fireworks AI](https://docs.pipecat.ai/server/services/llm/fireworks), [Gemini](https://docs.pipecat.ai/server/services/llm/gemini), [Grok](https://docs.pipecat.ai/server/services/llm/grok), [Groq](https://docs.pipecat.ai/server/services/llm/groq), [NVIDIA NIM](https://docs.pipecat.ai/server/services/llm/nim), [Ollama](https://docs.pipecat.ai/server/services/llm/ollama), [OpenAI](https://docs.pipecat.ai/server/services/llm/openai), [OpenRouter](https://docs.pipecat.ai/server/services/llm/openrouter), [Perplexity](https://docs.pipecat.ai/server/services/llm/perplexity), [Qwen](https://docs.pipecat.ai/server/services/llm/qwen), [Together AI](https://docs.pipecat.ai/server/services/llm/together) |
| Text-to-Speech | [AWS](https://docs.pipecat.ai/server/services/tts/aws), [Azure](https://docs.pipecat.ai/server/services/tts/azure), [Cartesia](https://docs.pipecat.ai/server/services/tts/cartesia), [Deepgram](https://docs.pipecat.ai/server/services/tts/deepgram), [ElevenLabs](https://docs.pipecat.ai/server/services/tts/elevenlabs), [FastPitch (NVIDIA)](https://docs.pipecat.ai/server/services/tts/fastpitch), [Fish](https://docs.pipecat.ai/server/services/tts/fish), [Google](https://docs.pipecat.ai/server/services/tts/google), [LMNT](https://docs.pipecat.ai/server/services/tts/lmnt), [Neuphonic](https://docs.pipecat.ai/server/services/tts/neuphonic), [OpenAI](https://docs.pipecat.ai/server/services/tts/openai), [Piper](https://docs.pipecat.ai/server/services/tts/piper), [PlayHT](https://docs.pipecat.ai/server/services/tts/playht), [Rime](https://docs.pipecat.ai/server/services/tts/rime), [XTTS](https://docs.pipecat.ai/server/services/tts/xtts) |
| Speech-to-Speech | [Gemini Multimodal Live](https://docs.pipecat.ai/server/services/s2s/gemini), [OpenAI Realtime](https://docs.pipecat.ai/server/services/s2s/openai) |
| Transport | [Daily (WebRTC)](https://docs.pipecat.ai/server/services/transport/daily), [FastAPI Websocket](https://docs.pipecat.ai/server/services/transport/fastapi-websocket), [SmallWebRTCTransport](https://docs.pipecat.ai/server/services/transport/small-webrtc), [WebSocket Server](https://docs.pipecat.ai/server/services/transport/websocket-server), Local |
| Video | [Tavus](https://docs.pipecat.ai/server/services/video/tavus), [Simli](https://docs.pipecat.ai/server/services/video/simli) |
| Memory | [mem0](https://docs.pipecat.ai/server/services/memory/mem0) |
| Vision & Image | [fal](https://docs.pipecat.ai/server/services/image-generation/fal), [Google Imagen](https://docs.pipecat.ai/server/services/image-generation/fal), [Moondream](https://docs.pipecat.ai/server/services/vision/moondream) |
| Audio Processing | [Silero VAD](https://docs.pipecat.ai/server/utilities/audio/silero-vad-analyzer), [Krisp](https://docs.pipecat.ai/server/utilities/audio/krisp-filter), [Koala](https://docs.pipecat.ai/server/utilities/audio/koala-filter), [Noisereduce](https://docs.pipecat.ai/server/utilities/audio/noisereduce-filter) |
| Analytics & Metrics | [Sentry](https://docs.pipecat.ai/server/services/analytics/sentry) |
| Category | Services |
| ------------------- | --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| Speech-to-Text | [AssemblyAI](https://docs.pipecat.ai/server/services/stt/assemblyai), [AWS](https://docs.pipecat.ai/server/services/stt/aws), [Azure](https://docs.pipecat.ai/server/services/stt/azure), [Cartesia](https://docs.pipecat.ai/server/services/stt/cartesia), [Deepgram](https://docs.pipecat.ai/server/services/stt/deepgram), [Fal Wizper](https://docs.pipecat.ai/server/services/stt/fal), [Gladia](https://docs.pipecat.ai/server/services/stt/gladia), [Google](https://docs.pipecat.ai/server/services/stt/google), [Groq (Whisper)](https://docs.pipecat.ai/server/services/stt/groq), [OpenAI (Whisper)](https://docs.pipecat.ai/server/services/stt/openai), [Parakeet (NVIDIA)](https://docs.pipecat.ai/server/services/stt/parakeet), [SambaNova (Whisper)](https://docs.pipecat.ai/server/services/stt/sambanova) [Ultravox](https://docs.pipecat.ai/server/services/stt/ultravox), [Whisper](https://docs.pipecat.ai/server/services/stt/whisper) |
| LLMs | [Anthropic](https://docs.pipecat.ai/server/services/llm/anthropic), [AWS](https://docs.pipecat.ai/server/services/llm/aws), [Azure](https://docs.pipecat.ai/server/services/llm/azure), [Cerebras](https://docs.pipecat.ai/server/services/llm/cerebras), [DeepSeek](https://docs.pipecat.ai/server/services/llm/deepseek), [Fireworks AI](https://docs.pipecat.ai/server/services/llm/fireworks), [Gemini](https://docs.pipecat.ai/server/services/llm/gemini), [Grok](https://docs.pipecat.ai/server/services/llm/grok), [Groq](https://docs.pipecat.ai/server/services/llm/groq), [NVIDIA NIM](https://docs.pipecat.ai/server/services/llm/nim), [Ollama](https://docs.pipecat.ai/server/services/llm/ollama), [OpenAI](https://docs.pipecat.ai/server/services/llm/openai), [OpenRouter](https://docs.pipecat.ai/server/services/llm/openrouter), [Perplexity](https://docs.pipecat.ai/server/services/llm/perplexity), [Qwen](https://docs.pipecat.ai/server/services/llm/qwen), [SambaNova](https://docs.pipecat.ai/server/services/llm/sambanova) [Together AI](https://docs.pipecat.ai/server/services/llm/together) |
| Text-to-Speech | [AWS](https://docs.pipecat.ai/server/services/tts/aws), [Azure](https://docs.pipecat.ai/server/services/tts/azure), [Cartesia](https://docs.pipecat.ai/server/services/tts/cartesia), [Deepgram](https://docs.pipecat.ai/server/services/tts/deepgram), [ElevenLabs](https://docs.pipecat.ai/server/services/tts/elevenlabs), [FastPitch (NVIDIA)](https://docs.pipecat.ai/server/services/tts/fastpitch), [Fish](https://docs.pipecat.ai/server/services/tts/fish), [Google](https://docs.pipecat.ai/server/services/tts/google), [LMNT](https://docs.pipecat.ai/server/services/tts/lmnt), [MiniMax](https://docs.pipecat.ai/server/services/tts/minimax), [Neuphonic](https://docs.pipecat.ai/server/services/tts/neuphonic), [OpenAI](https://docs.pipecat.ai/server/services/tts/openai), [Piper](https://docs.pipecat.ai/server/services/tts/piper), [PlayHT](https://docs.pipecat.ai/server/services/tts/playht), [Rime](https://docs.pipecat.ai/server/services/tts/rime), [Sarvam](https://docs.pipecat.ai/server/services/tts/sarvam), [XTTS](https://docs.pipecat.ai/server/services/tts/xtts) |
| Speech-to-Speech | [AWS Nova Sonic](https://docs.pipecat.ai/server/services/s2s/aws), [Gemini Multimodal Live](https://docs.pipecat.ai/server/services/s2s/gemini), [OpenAI Realtime](https://docs.pipecat.ai/server/services/s2s/openai) |
| Transport | [Daily (WebRTC)](https://docs.pipecat.ai/server/services/transport/daily), [FastAPI Websocket](https://docs.pipecat.ai/server/services/transport/fastapi-websocket), [SmallWebRTCTransport](https://docs.pipecat.ai/server/services/transport/small-webrtc), [WebSocket Server](https://docs.pipecat.ai/server/services/transport/websocket-server), Local |
| Serializers | [Plivo](https://docs.pipecat.ai/server/utilities/serializers/plivo), [Twilio](https://docs.pipecat.ai/server/utilities/serializers/twilio), [Telnyx](https://docs.pipecat.ai/server/utilities/serializers/telnyx) |
| Video | [Tavus](https://docs.pipecat.ai/server/services/video/tavus), [Simli](https://docs.pipecat.ai/server/services/video/simli) |
| Memory | [mem0](https://docs.pipecat.ai/server/services/memory/mem0) |
| Vision & Image | [fal](https://docs.pipecat.ai/server/services/image-generation/fal), [Google Imagen](https://docs.pipecat.ai/server/services/image-generation/fal), [Moondream](https://docs.pipecat.ai/server/services/vision/moondream) |
| Audio Processing | [Silero VAD](https://docs.pipecat.ai/server/utilities/audio/silero-vad-analyzer), [Krisp](https://docs.pipecat.ai/server/utilities/audio/krisp-filter), [Koala](https://docs.pipecat.ai/server/utilities/audio/koala-filter), [Noisereduce](https://docs.pipecat.ai/server/utilities/audio/noisereduce-filter) |
| Analytics & Metrics | [OpenTelemetry](https://docs.pipecat.ai/server/utilities/opentelemetry), [Sentry](https://docs.pipecat.ai/server/services/analytics/sentry) |
📚 [View full services documentation →](https://docs.pipecat.ai/server/services/supported-services)

View File

@@ -3,11 +3,11 @@ coverage~=7.6.12
grpcio-tools~=1.67.1
pip-tools~=7.4.1
pre-commit~=4.0.1
pyright~=1.1.397
pyright~=1.1.400
pytest~=8.3.4
pytest-asyncio~=0.25.3
pytest-aiohttp==1.1.0
ruff~=0.11.1
ruff~=0.11.13
setuptools~=70.0.0
setuptools_scm~=8.1.0
python-dotenv~=1.0.1

View File

@@ -1,5 +1,6 @@
import logging
import sys
from datetime import datetime
from pathlib import Path
# Configure logging
@@ -13,7 +14,8 @@ sys.path.insert(0, str(project_root / "src"))
# Project information
project = "pipecat-ai"
copyright = "2024, Daily"
current_year = datetime.now().year
copyright = f"2024-{current_year}, Daily" if current_year > 2024 else "2024, Daily"
author = "Daily"
# General configuration
@@ -27,15 +29,14 @@ extensions = [
# Napoleon settings
napoleon_google_docstring = True
napoleon_numpy_docstring = False
napoleon_include_init_with_doc = True
napoleon_include_init_with_doc = False
# AutoDoc settings
autodoc_default_options = {
"members": True,
"member-order": "bysource",
"special-members": "__init__",
"undoc-members": True,
"exclude-members": "__weakref__",
"exclude-members": "__weakref__,__init__",
"no-index": True,
"show-inheritance": True,
}
@@ -145,6 +146,28 @@ autodoc_mock_imports = [
"transformers.AutoFeatureExtractor",
# Also add specific classes that are imported
"AutoFeatureExtractor",
# Sentry dependencies
"sentry_sdk",
# AWS Nova Sonic dependencies
"aws_sdk_bedrock_runtime",
"aws_sdk_bedrock_runtime.client",
"aws_sdk_bedrock_runtime.config",
"aws_sdk_bedrock_runtime.models",
"smithy_aws_core",
"smithy_aws_core.credentials_resolvers",
"smithy_aws_core.credentials_resolvers.static",
"smithy_aws_core.identity",
"smithy_core",
"smithy_core.aio",
"smithy_core.aio.eventstream",
# MCP dependencies (you may already have these)
"mcp",
"mcp.client",
"mcp.client.session_group",
"mcp.client.sse",
"mcp.client.stdio",
"mcp.ClientSession",
"mcp.StdioServerParameters",
]
# HTML output settings
@@ -249,6 +272,9 @@ def clean_title(title: str) -> str:
"playht": "PlayHT",
"xtts": "XTTS",
"lmnt": "LMNT",
"stt": "STT",
"tts": "TTS",
"llm": "LLM",
}
# Check if the entire title is a special case

View File

@@ -42,6 +42,7 @@ pipecat-ai[openai]
pipecat-ai[qwen]
pipecat-ai[remote-smart-turn]
# pipecat-ai[riva] # Mocked
pipecat-ai[sambanova]
pipecat-ai[silero]
pipecat-ai[simli]
pipecat-ai[soundfile]

View File

@@ -95,9 +95,22 @@ OPENROUTER_API_KEY=...
PIPER_BASE_URL=...
# Smart turn
LOCAL_SMART_TURN_MODEL_PATH=
LOCAL_SMART_TURN_MODEL_PATH=...
FAL_SMART_TURN_API_KEY=...
# Twilio
TWILIO_ACCOUNT_SID=
TWILIO_AUTH_TOKEN=
TWILIO_ACCOUNT_SID=...
TWILIO_AUTH_TOKEN=...
# MiniMax
MINIMAX_API_KEY=...
MINIMAX_GROUP_ID=...
# Sarvam AI
SARVAM_API_KEY=...
# SambaNova
SAMBANOVA_API_KEY=...
# Sentry
SENTRY_DSN=...

View File

@@ -12,7 +12,7 @@
"@daily-co/daily-js": "0.74.0"
},
"devDependencies": {
"vite": "^6.0.9"
"vite": "^6.3.5"
}
},
"node_modules/@babel/runtime": {
@@ -999,9 +999,9 @@
}
},
"node_modules/vite": {
"version": "6.3.3",
"resolved": "https://registry.npmjs.org/vite/-/vite-6.3.3.tgz",
"integrity": "sha512-5nXH+QsELbFKhsEfWLkHrvgRpTdGJzqOZ+utSdmPTvwHmvU6ITTm3xx+mRusihkcI8GeC7lCDyn3kDtiki9scw==",
"version": "6.3.5",
"resolved": "https://registry.npmjs.org/vite/-/vite-6.3.5.tgz",
"integrity": "sha512-cZn6NDFE7wdTpINgs++ZJ4N49W2vRp8LCKrn3Ob1kYNtOo21vfDoaV5GzBfLU4MovSAB8uNRm4jgzVQZ+mBzPQ==",
"dev": true,
"dependencies": {
"esbuild": "^0.25.0",

View File

@@ -12,7 +12,7 @@
"license": "ISC",
"description": "",
"devDependencies": {
"vite": "^6.0.9"
"vite": "^6.3.5"
},
"dependencies": {
"@daily-co/daily-js": "0.74.0"

View File

@@ -128,7 +128,15 @@ async def main():
]
)
task = PipelineTask(pipeline, params=PipelineParams(allow_interruptions=True))
task = PipelineTask(
pipeline,
params=PipelineParams(
audio_in_sample_rate=16000,
audio_out_sample_rate=16000,
enable_metrics=True,
enable_usage_metrics=True,
),
)
@audiobuffer.event_handler("on_audio_data")
async def on_audio_data(buffer, audio, sample_rate, num_channels):

View File

@@ -71,6 +71,8 @@ async def main():
params=PipelineParams(
audio_in_sample_rate=16000,
audio_out_sample_rate=16000,
enable_metrics=True,
enable_usage_metrics=True,
),
)

View File

@@ -148,10 +148,8 @@ async def main():
params=PipelineParams(
audio_in_sample_rate=16000,
audio_out_sample_rate=16000,
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
observers=[TranscriptionLogObserver()],
)

View File

@@ -75,7 +75,13 @@ async def main(room_url: str, token: str):
]
)
task = PipelineTask(pipeline, params=PipelineParams(allow_interruptions=True))
task = PipelineTask(
pipeline,
params=PipelineParams(
enable_metrics=True,
enable_usage_metrics=True,
),
)
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):

View File

@@ -1,3 +1,6 @@
# Modal clone
modal-examples
# Python
__pycache__/
*.py[cod]

View File

@@ -1,37 +1,91 @@
# Deploying Pipecat to Modal.com
Barebones deployment example for [modal.com](https://www.modal.com)
Deployment example for [modal.com](https://www.modal.com). This example demonstrates how to deploy a FastAPI webapp to Modal with an RTVI compatible `/connect` endpoint that launches a Pipecat pipeline in a separate Modal container and returns a room/token for the client to join. This example also supports providing a parameter to the `/connect` endpoint for specifying which Pipecat pipeline to launch; openai, gemini, or vllm. The vllm pipeline points to a self-hosted OpenAI compatible LLM, using a llama model (neuralmagic/Meta-Llama-3.1-8B-Instruct-quantized.w4a16), deployed to Modal.
1. Install dependencies
![](diagram.jpg)
```bash
python -m venv venv
source venv/bin/active # or OS equivalent
pip install -r requirements.txt
```
# Running this Example
2. Setup .env
## Install the Modal CLI
```bash
cp env.example .env
```
Setup a Modal account and install it on your machine if you have not already, following their easy 3 steps in their [Getting Started Guide](https://modal.com/docs/guide#getting-started)
Alternatively, you can configure your Modal app to use [secrets](https://modal.com/docs/guide/secrets)
## Deploy a self-serve LLM
3. Test the app locally
1. Deploy Modal's OpenAI-compatible LLM service:
```bash
modal serve app.py
```
```bash
git clone https://github.com/modal-labs/modal-examples
cd modal-examples
modal deploy 06_gpu_and_ml/llm-serving/vllm_inference.py
```
Refer to Modal's guide and example for [Deploying an OpenAI-compatible LLM service with vLLM](https://modal.com/docs/examples/vllm_inference) for more details.
2. Take note of the endpoint URL from the previous step, which will look like:
```
https://{your-workspace}--example-vllm-openai-compatible-serve.modal.run
```
You'll need this for the `bot_vllm.py` file in the next section.
**Note:** The default Modal LLM example uses Llama-3.1 and will shut down after 15 minutes of inactivity. Cold starts take 5-10 minutes. To prepare the service, we recommend visiting the `/docs` endpoint (`https://<Modal workspace>--example-vllm-openai-compatible-serve.modal.run/docs`) for your deployed LLM and wait for it to fully load before connecting your client.
## Deploy FastAPI App and Pipecat pipeline to Modal
1. Setup environment variables
```bash
cd server
cp env.example .env
# Modify .env to provide your service API Keys
```
Alternatively, you can configure your Modal app to use [secrets](https://modal.com/docs/guide/secrets)
2. Update the `modal_url` in `server/src/bot_vllm.py` to point to the url produced from the self-serve llm deploy, mentioned above.
3. From within the `server` directory, test the app locally:
```bash
modal serve app.py
```
4. Deploy to production
```bash
modal deploy app.py
```
```bash
modal deploy app.py
```
## Configuration options
5. Note the endpoint URL produced from this deployment. It will look like:
This app sets some sensible defaults for reducing cold starts, such as `minkeep_warm=1`, which will keep at least 1 warm instance ready for your bot function.
```bash
https://{your-workspace}--pipecat-modal-fastapi-app.modal.run
```
It has been configured to only allow a concurrency of 1 (`max_inputs=1`) as each user will require their own running function.
You'll need this URL for the client's `app.js` configuration mentioned in its README.
## Launch your bots on Modal
### Option 1: Direct Link
Simply click on the url displayed after running the server or deploy step to launch an agent and be redirected to a Daily room to talk with the launched bot. This will use the OpenAI pipeline.
### Option 2: Connect via an RTVI Client
Follow the instructions provided in the [client folder's README](client/javascript/README.md) for building and running a custom client that connects to your Modal endpoint. The provided client provides a dropdown for choosing which bot pipeline to run.
# Navigating your llm, server, and Pipecat logs
In your [Modal dashboard](https://modal.com/apps), you should have two Apps listed under Live Apps:
1. `example-vllm-openai-compatible`: This App contains the containers and logs used to run your self-hosted LLM. There will be just one App Function listed: `serve`. Click on this function to view logs for your LLM.
2. `pipecat-modal`: This App contains the containers and logs used to run your `connect` endpoints and Pipecat pipelines. It will list two App Functions:
1. `fastapi_app`: This function is running the endpoints that your client will interact with and initiate starting a new pipeline (`/`, `/connect`, `/status`). Click on this function to see logs for each endpoint hit.
2. `bot_runner`: This function handles launching and running a bot pipeline. Click on this function to get a list of all pipeline runs and access each run's logs.
# Modal + Pipecat Tips
- In most other Pipecat examples, we use `Popen` to launch the pipeline process from the `/connect` endpoint. In this example, we use a Modal function instead. This allows us to run the pipelines using a separately defined Modal image as well as run each pipeline in an isolated container.
- For the FastAPI and most common Pipecat Pipeline containers, a default `debian_slim` CPU-only should be all that's required to run. GPU containers are needed for self-hosted services.
- To minimize cold starts of the pipeline and reduce latency for users, set `min_containers=1` on the Modal Function that launches the pipeline to ensure at least one warm instance of your function is always available.
- For next steps on running a self-hosted llm and reducing latency, check out all of [Modal's LLM examples](https://modal.com/docs/examples/vllm_inference).

View File

@@ -1,80 +0,0 @@
#
# Copyright (c) 20242025, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import os
import aiohttp
import modal
from bot import _voice_bot_process
from fastapi import HTTPException
from fastapi.responses import JSONResponse
from loguru import logger
MAX_SESSION_TIME = 15 * 60 # 15 minutes
app = modal.App("pipecat-modal")
image = modal.Image.debian_slim(python_version="3.12").pip_install_from_requirements(
"requirements.txt"
)
@app.function(
image=image,
cpu=1.0,
secrets=[modal.Secret.from_dotenv()],
keep_warm=1,
enable_memory_snapshot=True,
max_inputs=1, # Do not reuse instances across requests
retries=0,
)
def launch_bot_process(room_url: str, token: str):
_voice_bot_process(room_url, token)
@app.function(
image=image,
secrets=[modal.Secret.from_dotenv()],
)
@modal.web_endpoint(method="POST")
async def start():
from pipecat.transports.services.helpers.daily_rest import (
DailyRESTHelper,
DailyRoomParams,
)
logger.info("Request received")
async with aiohttp.ClientSession() as session:
daily_rest_helper = DailyRESTHelper(
daily_api_key=os.getenv("DAILY_API_KEY", ""),
daily_api_url=os.getenv("DAILY_API_URL", "https://api.daily.co/v1"),
aiohttp_session=session,
)
# Create new Daily room
room = await daily_rest_helper.create_room(DailyRoomParams())
if not room.url:
raise HTTPException(
status_code=500,
detail="Unable to create room",
)
logger.info(f"Created room: {room.url}")
# Create bot token for room
token = await daily_rest_helper.get_token(room.url, MAX_SESSION_TIME)
if not token:
raise HTTPException(status_code=500, detail=f"Failed to get token for room: {room.url}")
logger.info(f"Bot token created: {token}")
# Spawn a new bot process
launch_bot_process.spawn(room_url=room.url, token=token)
# Return room URL to the user to join
# Note: in production, you would want to return a token to the user
return JSONResponse(content={"room_url": room.url, token: token})

View File

@@ -1,95 +0,0 @@
#
# Copyright (c) 20242025, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import os
import sys
from dotenv import load_dotenv
from loguru import logger
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def main(room_url: str, token: str):
transport = DailyTransport(
room_url,
token,
"bot",
DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
transcription_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY", ""), voice_id="71a7ad14-091c-4e8e-a314-022ece01c121"
)
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
messages = [
{
"role": "system",
"content": "You are a helpful LLM in a WebRTC call. Your goal is to demonstrate your capabilities in a succinct way. Your output will be converted to audio so don't include special characters in your answers. Respond to what the user said in a creative and helpful way.",
},
]
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
transport.input(),
context_aggregator.user(),
llm,
tts,
transport.output(),
context_aggregator.assistant(),
]
)
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await transport.capture_participant_transcription(participant["id"])
messages.append({"role": "system", "content": "Please introduce yourself to the user."})
await task.queue_frames([context_aggregator.user().get_context_frame()])
@transport.event_handler("on_participant_left")
async def on_participant_left(transport, participant, reason):
await task.cancel()
runner = PipelineRunner()
await runner.run(task)
def _voice_bot_process(room_url: str, token: str):
asyncio.run(main(room_url, token))

View File

@@ -0,0 +1 @@
node_modules

View File

@@ -0,0 +1,29 @@
# JavaScript Implementation
Basic implementation using the [Pipecat JavaScript SDK](https://docs.pipecat.ai/client/js/introduction).
## Setup
1. Deploy the Modal server. See the main [README](../../README).
2. Navigate to the `client/javascript` directory:
```bash
cd client/javascript
```
3. Modify the baseUrl in src/app.js to point to your deployed Modal endpoint
4. Install dependencies:
```bash
npm install
```
5. Run the client app:
```
npm run dev
```
6. Visit http://localhost:5173 in your browser.

View File

@@ -0,0 +1,49 @@
<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="UTF-8" />
<meta name="viewport" content="width=device-width, initial-scale=1.0" />
<title>AI Chatbot</title>
</head>
<body>
<div class="container">
<div class="status-bar">
<div class="status">
Status: <span id="connection-status">Disconnected</span>
</div>
<div class="controls">
<select id="bot-selector">
<option value="openai">OpenAI</option>
<option value="gemini">Gemini</option>
<option value="vllm">Llama</option>
</select>
<button id="connect-btn">Connect</button>
<button id="disconnect-btn" disabled>Disconnect</button>
</div>
</div>
<div class="main-content">
<div class="bot-container">
<div id="bot-video-container"></div>
<audio id="bot-audio" autoplay></audio>
</div>
</div>
<div class="device-bar">
<div class="device-controls">
<select id="device-selector"></select>
<button id="mic-toggle-btn">Mute Mic</button>
</div>
</div>
<div class="debug-panel">
<h3>Debug Info</h3>
<div id="debug-log"></div>
</div>
</div>
<script type="module" src="/src/app.js"></script>
<link rel="stylesheet" href="/src/style.css" />
</body>
</html>

File diff suppressed because it is too large Load Diff

View File

@@ -0,0 +1,21 @@
{
"name": "client",
"version": "1.0.0",
"main": "index.js",
"scripts": {
"dev": "vite",
"build": "vite build",
"preview": "vite preview"
},
"keywords": [],
"author": "",
"license": "ISC",
"description": "",
"devDependencies": {
"vite": "^6.3.5"
},
"dependencies": {
"@pipecat-ai/client-js": "^0.3.5",
"@pipecat-ai/daily-transport": "^0.3.10"
}
}

View File

@@ -0,0 +1,381 @@
/**
* Copyright (c) 20242025, Daily
*
* SPDX-License-Identifier: BSD 2-Clause License
*/
/**
* RTVI Client Implementation
*
* This client connects to an RTVI-compatible bot server using WebRTC (via Daily).
* It handles audio/video streaming and manages the connection lifecycle.
*
* Requirements:
* - A running RTVI bot server (defaults to http://localhost:7860)
* - The server must implement the /connect endpoint that returns Daily.co room credentials
* - Browser with WebRTC support
*/
import { RTVIClient, RTVIEvent } from '@pipecat-ai/client-js';
import { DailyTransport } from '@pipecat-ai/daily-transport';
/**
* ChatbotClient handles the connection and media management for a real-time
* voice and video interaction with an AI bot.
*/
class ChatbotClient {
constructor() {
// Initialize client state
this.rtviClient = null;
this.setupDOMElements();
this.initializeClientAndTransport();
this.setupEventListeners();
}
/**
* Set up references to DOM elements and create necessary media elements
*/
setupDOMElements() {
// Get references to UI control elements
this.connectBtn = document.getElementById('connect-btn');
this.disconnectBtn = document.getElementById('disconnect-btn');
this.statusSpan = document.getElementById('connection-status');
this.debugLog = document.getElementById('debug-log');
this.botVideoContainer = document.getElementById('bot-video-container');
this.deviceSelector = document.getElementById('device-selector');
// Create an audio element for bot's voice output
this.botAudio = document.createElement('audio');
this.botAudio.autoplay = true;
this.botAudio.playsInline = true;
document.body.appendChild(this.botAudio);
}
/**
* Set up event listeners for connect/disconnect buttons
*/
setupEventListeners() {
this.connectBtn.addEventListener('click', () => this.connect());
this.disconnectBtn.addEventListener('click', () => this.disconnect());
// Populate device selector
this.rtviClient.getAllMics().then((mics) => {
console.log('Available mics:', mics);
mics.forEach((device) => {
const option = document.createElement('option');
option.value = device.deviceId;
option.textContent = device.label || `Microphone ${device.deviceId}`;
this.deviceSelector.appendChild(option);
});
});
this.deviceSelector.addEventListener('change', (event) => {
const selectedDeviceId = event.target.value;
console.log('Selected device ID:', selectedDeviceId);
this.rtviClient.updateMic(selectedDeviceId);
});
// Handle mic mute/unmute toggle
const micToggleBtn = document.getElementById('mic-toggle-btn');
micToggleBtn.addEventListener('click', () => {
let micEnabled = this.rtviClient.isMicEnabled;
micToggleBtn.textContent = micEnabled ? 'Unmute Mic' : 'Mute Mic';
this.rtviClient.enableMic(!micEnabled);
// Add logic to mute/unmute the mic
if (micEnabled) {
console.log('Mic muted');
// Add code to mute the mic
} else {
console.log('Mic unmuted');
// Add code to unmute the mic
}
});
}
/**
* Set up the RTVI client and Daily transport
*/
async initializeClientAndTransport() {
// Initialize the RTVI client with a DailyTransport and our configuration
this.rtviClient = new RTVIClient({
transport: new DailyTransport(),
params: {
// REPLACE WITH YOUR MODAL URL ENDPOINT
baseUrl:
'https://<Modal workspace>--pipecat-modal-bot-launcher.modal.run',
endpoints: {
connect: '/connect',
},
requestData: {
bot_name: 'openai',
},
},
enableMic: true, // Enable microphone for user input
enableCam: false,
callbacks: {
// Handle connection state changes
onConnected: () => {
this.updateStatus('Connected');
this.connectBtn.disabled = true;
this.disconnectBtn.disabled = false;
this.log('Client connected');
},
onDisconnected: () => {
this.updateStatus('Disconnected');
this.connectBtn.disabled = false;
this.disconnectBtn.disabled = true;
this.log('Client disconnected');
},
// Handle transport state changes
onTransportStateChanged: (state) => {
this.updateStatus(`Transport: ${state}`);
this.log(`Transport state changed: ${state}`);
if (state === 'connecting') {
window.startTime = Date.now();
}
if (state === 'ready') {
this.setupMediaTracks();
console.warn('TIME TO BOT READY:', Date.now() - window.startTime);
}
},
// Handle bot connection events
onBotConnected: (participant) => {
this.log(`Bot connected: ${JSON.stringify(participant)}`);
},
onBotDisconnected: (participant) => {
this.log(`Bot disconnected: ${JSON.stringify(participant)}`);
},
onBotReady: (data) => {
this.log(`Bot ready: ${JSON.stringify(data)}`);
this.setupMediaTracks();
},
// Transcript events
onUserTranscript: (data) => {
// Only log final transcripts
if (data.final) {
this.log(`User: ${data.text}`);
}
},
onBotTranscript: (data) => {
this.log(`Bot: ${data.text}`);
},
// Error handling
onMessageError: (error) => {
console.log('Message error:', error);
},
onMicUpdated: (data) => {
console.log('Mic updated:', data);
this.deviceSelector.value = data.deviceId;
},
onError: (error) => {
console.log('Error:', JSON.stringify(error));
},
},
});
// Set up listeners for media track events
this.setupTrackListeners();
await this.rtviClient.initDevices();
window.client = this.rtviClient;
}
/**
* Add a timestamped message to the debug log
*/
log(message) {
const entry = document.createElement('div');
entry.textContent = `${new Date().toISOString()} - ${message}`;
// Add styling based on message type
if (message.startsWith('User: ')) {
entry.style.color = '#2196F3'; // blue for user
} else if (message.startsWith('Bot: ')) {
entry.style.color = '#4CAF50'; // green for bot
}
this.debugLog.appendChild(entry);
this.debugLog.scrollTop = this.debugLog.scrollHeight;
console.log(message);
}
/**
* Update the connection status display
*/
updateStatus(status) {
this.statusSpan.textContent = status;
this.log(`Status: ${status}`);
}
/**
* Check for available media tracks and set them up if present
* This is called when the bot is ready or when the transport state changes to ready
*/
setupMediaTracks() {
if (!this.rtviClient) return;
// Get current tracks from the client
const tracks = this.rtviClient.tracks();
// Set up any available bot tracks
if (tracks.bot?.audio) {
this.setupAudioTrack(tracks.bot.audio);
}
if (tracks.bot?.video) {
this.setupVideoTrack(tracks.bot.video);
}
}
/**
* Set up listeners for track events (start/stop)
* This handles new tracks being added during the session
*/
setupTrackListeners() {
if (!this.rtviClient) return;
// Listen for new tracks starting
this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
// Only handle non-local (bot) tracks
if (!participant?.local) {
if (track.kind === 'audio') {
this.setupAudioTrack(track);
} else if (track.kind === 'video') {
this.setupVideoTrack(track);
}
this.log(
`Track started event: ${track.kind} from ${
participant?.name || 'unknown'
}`
);
} else {
this.log('Local mic unmuted');
}
});
// Listen for tracks stopping
this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
if (participant.local) {
this.log('Local mic muted');
return;
}
this.log(
`Track stopped event: ${track.kind} from ${
participant?.name || 'unknown'
}`
);
});
}
/**
* Set up an audio track for playback
* Handles both initial setup and track updates
*/
setupAudioTrack(track) {
this.log('Setting up audio track');
// Check if we're already playing this track
if (this.botAudio.srcObject) {
const oldTrack = this.botAudio.srcObject.getAudioTracks()[0];
if (oldTrack?.id === track.id) return;
}
// Create a new MediaStream with the track and set it as the audio source
this.botAudio.srcObject = new MediaStream([track]);
}
/**
* Set up a video track for display
* Handles both initial setup and track updates
*/
setupVideoTrack(track) {
this.log('Setting up video track');
const videoEl = document.createElement('video');
videoEl.autoplay = true;
videoEl.playsInline = true;
videoEl.muted = true;
videoEl.style.width = '100%';
videoEl.style.height = '100%';
videoEl.style.objectFit = 'cover';
// Check if we're already displaying this track
if (this.botVideoContainer.querySelector('video')?.srcObject) {
const oldTrack = this.botVideoContainer
.querySelector('video')
.srcObject.getVideoTracks()[0];
if (oldTrack?.id === track.id) return;
}
// Create a new MediaStream with the track and set it as the video source
videoEl.srcObject = new MediaStream([track]);
this.botVideoContainer.innerHTML = '';
this.botVideoContainer.appendChild(videoEl);
}
/**
* Initialize and connect to the bot
* This sets up the RTVI client, initializes devices, and establishes the connection
*/
async connect() {
try {
const botSelector = document.getElementById('bot-selector');
const selectedBot = botSelector.value;
this.rtviClient.params.requestData.bot_name = selectedBot;
// Initialize audio/video devices
this.log('Initializing devices...');
await this.rtviClient.initDevices();
// Connect to the bot
this.log(`Connecting to bot: ${selectedBot}`);
await this.rtviClient.connect();
this.log('Connection complete');
} catch (error) {
// Handle any errors during connection
console.error('Connection error:', error);
this.log(`Error connecting: ${JSON.stringify(error.message)}`);
this.log(`Error stack: ${error.stack}`);
this.updateStatus('Error');
// Clean up if there's an error
if (this.rtviClient) {
try {
await this.rtviClient.disconnect();
} catch (disconnectError) {
this.log(`Error during disconnect: ${disconnectError.message}`);
}
}
}
}
/**
* Disconnect from the bot and clean up media resources
*/
async disconnect() {
if (this.rtviClient) {
try {
// Disconnect the RTVI client
await this.rtviClient.disconnect();
// Clean up audio
if (this.botAudio.srcObject) {
this.botAudio.srcObject.getTracks().forEach((track) => track.stop());
this.botAudio.srcObject = null;
}
// Clean up video
if (this.botVideoContainer.querySelector('video')?.srcObject) {
const video = this.botVideoContainer.querySelector('video');
video.srcObject.getTracks().forEach((track) => track.stop());
video.srcObject = null;
}
this.botVideoContainer.innerHTML = '';
} catch (error) {
this.log(`Error disconnecting: ${error.message}`);
}
}
}
}
// Initialize the client when the page loads
window.addEventListener('DOMContentLoaded', () => {
new ChatbotClient();
});

View File

@@ -0,0 +1,135 @@
body {
margin: 0;
padding: 20px;
font-family: Arial, sans-serif;
background-color: #f0f0f0;
}
.container {
max-width: 1200px;
margin: 0 auto;
}
.status-bar,
.device-bar {
display: flex;
justify-content: space-between;
align-items: center;
padding: 10px;
background-color: #fff;
border-radius: 8px;
margin-bottom: 20px;
}
.controls,
.device-controls {
display: flex;
align-items: center;
gap: 10px; /* Adds spacing between elements */
}
.device-controls {
margin-left: auto;
}
.controls button,
.device-controls button {
padding: 8px 16px;
margin-left: 10px;
border: none;
border-radius: 4px;
cursor: pointer;
}
#bot-selector,
#device-selector {
padding: 8px 16px;
padding-right: 40px;
border: none;
border-radius: 4px;
background-color: #6c757d; /* Gray background */
color: white; /* White text */
cursor: pointer;
appearance: none; /* Removes default browser styling for dropdowns */
background-image: url("data:image/svg+xml,%3Csvg xmlns='http://www.w3.org/2000/svg' viewBox='0 0 24 24' fill='white'%3E%3Cpath d='M7 10l5 5 5-5z'/%3E%3C/svg%3E"); /* Custom arrow */
background-repeat: no-repeat;
background-position: right 8px center; /* Position the arrow */
}
#bot-selector:focus,
#device-selector:focus {
outline: none;
box-shadow: 0 0 4px rgba(0, 0, 0, 0.3); /* Add a subtle focus effect */
}
#connect-btn {
background-color: #4caf50;
color: white;
}
#disconnect-btn {
background-color: #f44336;
color: white;
}
#mic-toggle-btn {
}
button:disabled {
opacity: 0.5;
cursor: not-allowed;
}
.main-content {
background-color: #fff;
border-radius: 8px;
padding: 20px;
margin-bottom: 20px;
}
.bot-container {
display: flex;
flex-direction: column;
align-items: center;
}
#bot-video-container {
width: 640px;
height: 360px;
background-color: #e0e0e0;
border-radius: 8px;
margin: 20px auto;
overflow: hidden;
display: flex;
align-items: center;
justify-content: center;
}
#bot-video-container video {
width: 100%;
height: 100%;
object-fit: cover;
}
.debug-panel {
background-color: #fff;
border-radius: 8px;
padding: 20px;
}
.debug-panel h3 {
margin: 0 0 10px 0;
font-size: 16px;
font-weight: bold;
}
#debug-log {
height: 200px;
overflow-y: auto;
background-color: #f8f8f8;
padding: 10px;
border-radius: 4px;
font-family: monospace;
font-size: 12px;
line-height: 1.4;
}

Binary file not shown.

After

Width:  |  Height:  |  Size: 114 KiB

View File

@@ -1,3 +0,0 @@
DAILY_API_KEY=
OPENAI_API_KEY=
CARTESIA_API_KEY=

View File

@@ -1,4 +0,0 @@
python-dotenv==1.0.1
modal==0.71.3
pipecat-ai[daily,silero,cartesia,openai]
fastapi==0.115.6

View File

@@ -0,0 +1,307 @@
"""modal_example.
This module shows a simple example of how to deploy a bot using Modal and FastAPI.
It includes:
- FastAPI endpoints for starting agents and checking bot statuses.
- Dynamic loading of bot implementations.
- Use of a Daily transport for bot communication.
"""
#
# Copyright (c) 20242025, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import importlib
import os
from contextlib import asynccontextmanager
from typing import Any, Dict, Literal
import aiohttp
import modal
from fastapi import APIRouter, FastAPI, HTTPException
from fastapi.responses import JSONResponse, RedirectResponse
from pydantic import BaseModel
# container specifications for the FastAPI web server
web_image = (
modal.Image.debian_slim(python_version="3.13")
.pip_install_from_requirements("requirements.txt")
.pip_install("pipecat-ai[daily]")
.add_local_dir("src", remote_path="/root/src")
)
# container specifications for the Pipecat pipeline
bot_image = (
modal.Image.debian_slim(python_version="3.13")
.apt_install("ffmpeg")
.pip_install_from_requirements("requirements.txt")
.pip_install("pipecat-ai[daily,elevenlabs,openai,silero,google]")
.add_local_dir("src", remote_path="/root/src")
)
app = modal.App("pipecat-modal", secrets=[modal.Secret.from_dotenv()])
router = APIRouter()
bot_jobs = {}
daily_helpers = {}
# Names of all supported bot implementations
# These correspond to the bot files in the src directory
BotName = Literal["openai", "gemini", "vllm"]
def cleanup():
"""Cleanup function to terminate all bot processes.
Called during server shutdown.
"""
for entry in bot_jobs.values():
func = modal.FunctionCall.from_id(entry[0])
if func:
func.cancel()
def get_bot_file(bot_name: BotName) -> str:
"""Retrieve the bot file name corresponding to the provided bot_name.
Args:
bot_name (BotName): The name of the bot (e.g., 'openai', 'gemini', 'vllm').
Returns:
str: The file name corresponding to the bot implementation.
Raises:
ValueError: If the bot name is invalid or not supported.
"""
# bot_implementation = os.getenv("BOT_IMPLEMENTATION", "openai").lower().strip()
bot_implementation = bot_name.lower().strip()
if not bot_implementation:
bot_implementation = "openai"
if bot_implementation not in ["openai", "gemini", "vllm"]:
raise ValueError(
f"Invalid BOT_IMPLEMENTATION: {bot_implementation}. Must be 'openai' or 'gemini' or 'vllm'"
)
return f"bot_{bot_implementation}"
def get_runner(path: str, bot_file: str) -> callable:
"""Dynamically import the run_bot function based on the bot name.
Args:
path (str): The path to the bot files (e.g., 'src').
bot_file (str): The file name of the bot implementation (e.g., 'openai', 'gemini', 'vllm').
Returns:
function: The run_bot function from the specified bot module.
Raises:
ImportError: If the specified bot module or run_bot function is not found.
"""
try:
# Dynamically construct the module name
module_name = f"{path}.{bot_file}"
# Import the module
module = importlib.import_module(module_name)
# Get the run_bot function from the module
return getattr(module, "run_bot")
except (ImportError, AttributeError) as e:
raise ImportError(f"Failed to import run_bot from {module_name}: {e}")
async def create_room_and_token() -> tuple[str, str]:
"""Create a Daily room and generate an authentication token.
This function checks for existing room URL and token in the environment variables.
If not found, it creates a new room using the Daily API and generates a token for it.
Returns:
tuple[str, str]: A tuple containing the room URL and the authentication token.
Raises:
HTTPException: If room creation or token generation fails.
"""
from pipecat.transports.services.helpers.daily_rest import DailyRoomParams
room_url = os.getenv("DAILY_SAMPLE_ROOM_URL", None)
token = os.getenv("DAILY_SAMPLE_ROOM_TOKEN", None)
if not room_url:
room = await daily_helpers["rest"].create_room(DailyRoomParams())
if not room.url:
raise HTTPException(status_code=500, detail="Failed to create room")
room_url = room.url
token = await daily_helpers["rest"].get_token(room_url)
if not token:
raise HTTPException(status_code=500, detail=f"Failed to get token for room: {room_url}")
return room_url, token
@app.function(image=bot_image, min_containers=1)
async def bot_runner(room_url, token, bot_name: BotName = "openai"):
"""Launch the provided bot process, providing the given room URL and token for the bot to join.
Args:
room_url (str): The URL of the Daily room where the bot and client will communicate.
token (str): The authentication token for the room.
bot_name (BotName): The name of the bot implementation to use. Defaults to "openai".
Raises:
HTTPException: If the bot pipeline fails to start.
"""
try:
path = "src"
bot_file = get_bot_file(bot_name)
run_bot = get_runner(path, bot_file)
print(f"Starting bot process: {bot_file} -u {room_url} -t {token}")
await run_bot(room_url, token)
except Exception as e:
raise HTTPException(status_code=500, detail=f"Failed to start bot pipeline: {e}")
@asynccontextmanager
async def lifespan(app: FastAPI):
"""FastAPI lifespan manager that handles startup and shutdown tasks.
- Creates aiohttp session
- Initializes Daily API helper
- Cleans up resources on shutdown
"""
from pipecat.transports.services.helpers.daily_rest import DailyRESTHelper
aiohttp_session = aiohttp.ClientSession()
daily_helpers["rest"] = DailyRESTHelper(
daily_api_key=os.getenv("DAILY_API_KEY", ""),
daily_api_url=os.getenv("DAILY_API_URL", "https://api.daily.co/v1"),
aiohttp_session=aiohttp_session,
)
yield
await aiohttp_session.close()
cleanup()
class ConnectData(BaseModel):
"""Data provided by client to specify the bot pipeline.
Attributes:
bot_name (BotName): The name of the bot to connect to. Defaults to "openai".
"""
bot_name: BotName = "openai"
async def start(data: ConnectData):
"""Internal method to start a bot agent and return the room URL and token.
Args:
data (ConnectData): The data containing the bot name to use.
Returns:
tuple[str, str]: A tuple containing the room URL and token.
"""
room_url, token = await create_room_and_token()
launch_bot_func = modal.Function.from_name("pipecat-modal", "bot_runner")
function_id = launch_bot_func.spawn(room_url, token, data.bot_name)
bot_jobs[function_id] = (function_id, room_url)
return room_url, token
@router.get("/")
async def start_agent():
"""A user endpoint for launching a bot agent and redirecting to the created room URL.
This function retrieves the bot implementation from the environment,
starts the bot agent, and redirects the user to the room URL to
interact with the bot through a Daily Prebuilt Interface.
Returns:
RedirectResponse: A response that redirects to the room URL.
"""
bot_name = os.getenv("BOT_IMPLEMENTATION", "openai").lower().strip()
print(f"Starting bot: {bot_name}")
room_url, token = await start(ConnectData(bot_name=bot_name))
return RedirectResponse(room_url)
@router.post("/connect")
async def rtvi_connect(data: ConnectData) -> Dict[Any, Any]:
"""A user endpoint for launching a bot agent and retrieving the room/token credentials.
This function retrieves the bot implementation from the request, if provided,
starts the bot agent, and returns the room URL and token for the bot. This allows the
client to then connect to the bot using their own RTVI interface.
Args:
data (ConnectData): Optional. The data containing the bot name to use.
Returns:
Dict[Any, Any]: A dictionary containing the room URL and token.
"""
print(f"Starting bot: {data.bot_name}")
if data is None or not data.bot_name:
data.bot_name = os.getenv("BOT_IMPLEMENTATION", "openai").lower().strip()
room_url, token = await start(data)
return {"room_url": room_url, "token": token}
@router.get("/status/{fid}")
def get_status(fid: str):
"""Retrieve the status of a bot process by its function ID.
Args:
fid (str): The function ID of the bot process.
Returns:
JSONResponse: A JSON response containing the bot's status and result code.
Raises:
HTTPException: If the bot process with the given ID is not found.
"""
func = modal.FunctionCall.from_id(fid)
if not func:
raise HTTPException(status_code=404, detail=f"Bot with process id: {fid} not found")
try:
result = func.get(timeout=0)
return JSONResponse({"bot_id": fid, "status": "finished", "code": result})
except modal.exception.OutputExpiredError:
return JSONResponse({"bot_id": fid, "status": "finished", "code": 404})
except TimeoutError:
return JSONResponse({"bot_id": fid, "status": "running", "code": 202})
@app.function(image=web_image, min_containers=1)
@modal.concurrent(max_inputs=1)
@modal.asgi_app()
def fastapi_app():
"""Create and configure the FastAPI application.
This function initializes the FastAPI app with middleware, routes, and lifespan management.
It is decorated to be used as a Modal ASGI app.
"""
from fastapi.middleware.cors import CORSMiddleware
# Initialize FastAPI app
web_app = FastAPI(lifespan=lifespan)
web_app.add_middleware(
CORSMiddleware,
allow_origins=["*"],
allow_credentials=True,
allow_methods=["*"],
allow_headers=["*"],
)
# Include the endpoints from endpoints.py
web_app.include_router(router)
return web_app

View File

@@ -0,0 +1,14 @@
DAILY_API_KEY=
# determines which bot file to default to: 'openai', 'gemini', or 'vllm'
BOT_IMPLEMENTATION=openai
# needed for the openai bot pipeline
OPENAI_API_KEY=
ELEVENLABS_API_KEY=
# needed for the gemini live bot pipeline
GOOGLE_API_KEY=
# needed if you modified the API Key for your self-hosted LLM
VLLM_API_KEY=

View File

@@ -0,0 +1,2 @@
python-dotenv==1.0.1
modal==0.71.3

Binary file not shown.

After

Width:  |  Height:  |  Size: 759 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 884 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 876 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 881 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 866 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 874 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 882 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 885 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 888 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 890 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 898 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 836 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 903 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 908 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 908 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 905 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 903 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 866 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 849 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 866 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 866 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 864 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 858 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 875 KiB

Binary file not shown.

After

Width:  |  Height:  |  Size: 881 KiB

View File

@@ -0,0 +1,197 @@
#
# Copyright (c) 20242025, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
"""Gemini Bot Implementation.
This module implements a chatbot using Google's Gemini Multimodal Live model.
It includes:
- Real-time audio/video interaction through Daily
- Animated robot avatar
- Speech-to-speech model
The bot runs as part of a pipeline that processes audio/video frames and manages
the conversation flow using Gemini's streaming capabilities.
"""
import os
import sys
from dotenv import load_dotenv
from loguru import logger
from PIL import Image
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.audio.vad.vad_analyzer import VADParams
from pipecat.frames.frames import (
BotStartedSpeakingFrame,
BotStoppedSpeakingFrame,
Frame,
OutputImageRawFrame,
SpriteFrame,
)
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.processors.frameworks.rtvi import RTVIConfig, RTVIObserver, RTVIProcessor
from pipecat.services.gemini_multimodal_live.gemini import GeminiMultimodalLiveLLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
try:
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
except ValueError:
# Handle the case where logger is already initialized
pass
sprites = []
script_dir = os.path.dirname(__file__)
for i in range(1, 26):
# Build the full path to the image file
full_path = os.path.join(script_dir, f"assets/robot0{i}.png")
# Get the filename without the extension to use as the dictionary key
# Open the image and convert it to bytes
with Image.open(full_path) as img:
sprites.append(OutputImageRawFrame(image=img.tobytes(), size=img.size, format=img.format))
# Create a smooth animation by adding reversed frames
flipped = sprites[::-1]
sprites.extend(flipped)
# Define static and animated states
quiet_frame = sprites[0] # Static frame for when bot is listening
talking_frame = SpriteFrame(images=sprites) # Animation sequence for when bot is talking
class TalkingAnimation(FrameProcessor):
"""Manages the bot's visual animation states.
Switches between static (listening) and animated (talking) states based on
the bot's current speaking status.
"""
def __init__(self):
super().__init__()
self._is_talking = False
async def process_frame(self, frame: Frame, direction: FrameDirection):
"""Process incoming frames and update animation state.
Args:
frame: The incoming frame to process
direction: The direction of frame flow in the pipeline
"""
await super().process_frame(frame, direction)
# Switch to talking animation when bot starts speaking
if isinstance(frame, BotStartedSpeakingFrame):
if not self._is_talking:
await self.push_frame(talking_frame)
self._is_talking = True
# Return to static frame when bot stops speaking
elif isinstance(frame, BotStoppedSpeakingFrame):
await self.push_frame(quiet_frame)
self._is_talking = False
await self.push_frame(frame, direction)
async def run_bot(room_url: str, token: str):
"""Main bot execution function.
Sets up and runs the bot pipeline including:
- Daily video transport with specific audio parameters
- Gemini Live multimodal model integration
- Voice activity detection
- Animation processing
- RTVI event handling
"""
# Set up Daily transport with specific audio/video parameters for Gemini
transport = DailyTransport(
room_url,
token,
"Chatbot",
DailyParams(
audio_out_enabled=True,
camera_out_enabled=True,
camera_out_width=1024,
camera_out_height=576,
vad_enabled=True,
vad_audio_passthrough=True,
vad_analyzer=SileroVADAnalyzer(params=VADParams(stop_secs=0.5)),
),
)
# Initialize the Gemini Multimodal Live model
llm = GeminiMultimodalLiveLLMService(
api_key=os.getenv("GOOGLE_API_KEY"),
voice_id="Puck", # Aoede, Charon, Fenrir, Kore, Puck
transcribe_user_audio=True,
)
messages = [
{
"role": "user",
"content": "You are Chatbot, a friendly, helpful robot. Your goal is to demonstrate your capabilities in a succinct way. Your output will be converted to audio so don't include special characters in your answers. Respond to what the user said in a creative and helpful way, but keep your responses brief. Start by introducing yourself.",
},
]
# Set up conversation context and management
# The context_aggregator will automatically collect conversation context
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
ta = TalkingAnimation()
#
# RTVI events for Pipecat client UI
#
rtvi = RTVIProcessor(config=RTVIConfig(config=[]))
pipeline = Pipeline(
[
transport.input(),
rtvi,
context_aggregator.user(),
llm,
ta,
transport.output(),
context_aggregator.assistant(),
]
)
task = PipelineTask(
pipeline,
params=PipelineParams(
enable_metrics=True,
enable_usage_metrics=True,
),
observers=[RTVIObserver(rtvi)],
)
await task.queue_frame(quiet_frame)
@rtvi.event_handler("on_client_ready")
async def on_client_ready(rtvi):
await rtvi.set_bot_ready()
# Kick off the conversation
await task.queue_frames([context_aggregator.user().get_context_frame()])
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await transport.capture_participant_transcription(participant["id"])
@transport.event_handler("on_participant_left")
async def on_participant_left(transport, participant, reason):
print(f"Participant left: {participant}")
await task.cancel()
runner = PipelineRunner()
await runner.run(task)

View File

@@ -0,0 +1,225 @@
#
# Copyright (c) 20242025, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
"""OpenAI Bot Implementation.
This module implements a chatbot using OpenAI's GPT-4 model for natural language
processing. It includes:
- Real-time audio/video interaction through Daily
- Animated robot avatar
- Text-to-speech using ElevenLabs
- Support for both English and Spanish
The bot runs as part of a pipeline that processes audio/video frames and manages
the conversation flow.
"""
import os
import sys
from dotenv import load_dotenv
from loguru import logger
from PIL import Image
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import (
BotStartedSpeakingFrame,
BotStoppedSpeakingFrame,
Frame,
OutputImageRawFrame,
SpriteFrame,
)
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.processors.frameworks.rtvi import RTVIConfig, RTVIObserver, RTVIProcessor
from pipecat.services.elevenlabs.tts import ElevenLabsTTSService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
try:
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
except ValueError:
# Handle the case where logger is already initialized
pass
sprites = []
script_dir = os.path.dirname(__file__)
# Load sequential animation frames
for i in range(1, 26):
# Build the full path to the image file
full_path = os.path.join(script_dir, f"assets/robot0{i}.png")
# Get the filename without the extension to use as the dictionary key
# Open the image and convert it to bytes
with Image.open(full_path) as img:
sprites.append(OutputImageRawFrame(image=img.tobytes(), size=img.size, format=img.format))
# Create a smooth animation by adding reversed frames
flipped = sprites[::-1]
sprites.extend(flipped)
# Define static and animated states
quiet_frame = sprites[0] # Static frame for when bot is listening
talking_frame = SpriteFrame(images=sprites) # Animation sequence for when bot is talking
class TalkingAnimation(FrameProcessor):
"""Manages the bot's visual animation states.
Switches between static (listening) and animated (talking) states based on
the bot's current speaking status.
"""
def __init__(self):
super().__init__()
self._is_talking = False
async def process_frame(self, frame: Frame, direction: FrameDirection):
"""Process incoming frames and update animation state.
Args:
frame: The incoming frame to process
direction: The direction of frame flow in the pipeline
"""
await super().process_frame(frame, direction)
# Switch to talking animation when bot starts speaking
if isinstance(frame, BotStartedSpeakingFrame):
if not self._is_talking:
await self.push_frame(talking_frame)
self._is_talking = True
# Return to static frame when bot stops speaking
elif isinstance(frame, BotStoppedSpeakingFrame):
await self.push_frame(quiet_frame)
self._is_talking = False
await self.push_frame(frame, direction)
async def run_bot(room_url: str, token: str):
"""Main bot execution function.
Sets up and runs the bot pipeline including:
- Daily video transport
- Speech-to-text and text-to-speech services
- Language model integration
- Animation processing
- RTVI event handling
"""
# Set up Daily transport with video/audio parameters
transport = DailyTransport(
room_url,
token,
"Chatbot",
DailyParams(
audio_out_enabled=True,
camera_out_enabled=True,
camera_out_width=1024,
camera_out_height=576,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
transcription_enabled=True,
#
# Spanish
#
# transcription_settings=DailyTranscriptionSettings(
# language="es",
# tier="nova",
# model="2-general"
# )
),
)
# Initialize text-to-speech service
tts = ElevenLabsTTSService(
api_key=os.getenv("ELEVENLABS_API_KEY"),
#
# English
#
voice_id="SAz9YHcvj6GT2YYXdXww",
#
# Spanish
#
# model="eleven_multilingual_v2",
# voice_id="gD1IexrzCvsXPHUuT0s3",
)
# Initialize LLM service
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
messages = [
{
"role": "system",
#
# English
#
"content": "You are an incessant one-upper. Start by asking the user how their day is going.",
#
# Spanish
#
# "content": "Eres Chatbot, un amigable y útil robot. Tu objetivo es demostrar tus capacidades de una manera breve. Tus respuestas se convertiran a audio así que nunca no debes incluir caracteres especiales. Contesta a lo que el usuario pregunte de una manera creativa, útil y breve. Empieza por presentarte a ti mismo.",
},
]
# Set up conversation context and management
# The context_aggregator will automatically collect conversation context
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
ta = TalkingAnimation()
#
# RTVI events for Pipecat client UI
#
rtvi = RTVIProcessor(config=RTVIConfig(config=[]))
pipeline = Pipeline(
[
transport.input(),
rtvi,
context_aggregator.user(),
llm,
tts,
ta,
transport.output(),
context_aggregator.assistant(),
]
)
task = PipelineTask(
pipeline,
params=PipelineParams(
enable_metrics=True,
enable_usage_metrics=True,
),
observers=[RTVIObserver(rtvi)],
)
await task.queue_frame(quiet_frame)
@rtvi.event_handler("on_client_ready")
async def on_client_ready(rtvi):
await rtvi.set_bot_ready()
# Kick off the conversation
await task.queue_frames([context_aggregator.user().get_context_frame()])
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await transport.capture_participant_transcription(participant["id"])
@transport.event_handler("on_participant_left")
async def on_participant_left(transport, participant, reason):
print(f"Participant left: {participant}")
await task.cancel()
runner = PipelineRunner()
await runner.run(task)

View File

@@ -0,0 +1,238 @@
#
# Copyright (c) 20242025, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
"""OpenAI Bot Implementation.
This module implements a chatbot using OpenAI's GPT-4 model for natural language
processing. It includes:
- Real-time audio/video interaction through Daily
- Animated robot avatar
- Text-to-speech using ElevenLabs
- Support for both English and Spanish
The bot runs as part of a pipeline that processes audio/video frames and manages
the conversation flow.
"""
import os
import sys
from typing import List
from dotenv import load_dotenv
from loguru import logger
from openai.types.chat import ChatCompletionMessageParam
from PIL import Image
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import (
BotStartedSpeakingFrame,
BotStoppedSpeakingFrame,
Frame,
OutputImageRawFrame,
SpriteFrame,
)
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.processors.frameworks.rtvi import RTVIConfig, RTVIObserver, RTVIProcessor
from pipecat.services.elevenlabs.tts import ElevenLabsTTSService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
try:
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
except ValueError:
# Handle the case where logger is already initialized
pass
# REPLACE WITH YOUR MODAL URL ENDPOINT
modal_url = "https://<Modal workspace>--example-vllm-openai-compatible-serve.modal.run"
api_key = os.getenv("VLLM_API_KEY", "super-secret-key")
sprites = []
script_dir = os.path.dirname(__file__)
# Load sequential animation frames
for i in range(1, 26):
# Build the full path to the image file
full_path = os.path.join(script_dir, f"assets/robot0{i}.png")
# Get the filename without the extension to use as the dictionary key
# Open the image and convert it to bytes
with Image.open(full_path) as img:
sprites.append(OutputImageRawFrame(image=img.tobytes(), size=img.size, format=img.format))
# Create a smooth animation by adding reversed frames
flipped = sprites[::-1]
sprites.extend(flipped)
# Define static and animated states
quiet_frame = sprites[0] # Static frame for when bot is listening
talking_frame = SpriteFrame(images=sprites) # Animation sequence for when bot is talking
class TalkingAnimation(FrameProcessor):
"""Manages the bot's visual animation states.
Switches between static (listening) and animated (talking) states based on
the bot's current speaking status.
"""
def __init__(self):
super().__init__()
self._is_talking = False
async def process_frame(self, frame: Frame, direction: FrameDirection):
"""Process incoming frames and update animation state.
Args:
frame: The incoming frame to process
direction: The direction of frame flow in the pipeline
"""
await super().process_frame(frame, direction)
# Switch to talking animation when bot starts speaking
if isinstance(frame, BotStartedSpeakingFrame):
if not self._is_talking:
await self.push_frame(talking_frame)
self._is_talking = True
# Return to static frame when bot stops speaking
elif isinstance(frame, BotStoppedSpeakingFrame):
await self.push_frame(quiet_frame)
self._is_talking = False
await self.push_frame(frame, direction)
async def run_bot(room_url: str, token: str):
"""Main bot execution function.
Sets up and runs the bot pipeline including:
- Daily video transport
- Speech-to-text and text-to-speech services
- Language model integration
- Animation processing
- RTVI event handling
"""
# Set up Daily transport with video/audio parameters
transport = DailyTransport(
room_url,
token,
"Chatbot",
DailyParams(
audio_out_enabled=True,
camera_out_enabled=True,
camera_out_width=1024,
camera_out_height=576,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
transcription_enabled=True,
#
# Spanish
#
# transcription_settings=DailyTranscriptionSettings(
# language="es",
# tier="nova",
# model="2-general"
# )
),
)
# Initialize text-to-speech service
tts = ElevenLabsTTSService(
api_key=os.getenv("ELEVENLABS_API_KEY"),
#
# English
#
voice_id="D38z5RcWu1voky8WS1ja",
#
# Spanish
#
# model="eleven_multilingual_v2",
# voice_id="gD1IexrzCvsXPHUuT0s3",
)
# Initialize LLM service
llm = OpenAILLMService(
# To use OpenAI
api_key=api_key,
# Or, to use a local vLLM (or similar) api server
model="neuralmagic/Meta-Llama-3.1-8B-Instruct-quantized.w4a16",
base_url=f"{modal_url}/v1",
)
messages = [
{
"role": "system",
#
# English
#
"content": "You are a salesman for Modal, the cloud-native serverless Python computing platform.",
#
# Spanish
#
# "content": "Eres Chatbot, un amigable y útil robot. Tu objetivo es demostrar tus capacidades de una manera breve. Tus respuestas se convertiran a audio así que nunca no debes incluir caracteres especiales. Contesta a lo que el usuario pregunte de una manera creativa, útil y breve. Empieza por presentarte a ti mismo.",
},
]
# Set up conversation context and management
# The context_aggregator will automatically collect conversation context
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
ta = TalkingAnimation()
#
# RTVI events for Pipecat client UI
#
rtvi = RTVIProcessor(config=RTVIConfig(config=[]))
pipeline = Pipeline(
[
transport.input(),
rtvi,
context_aggregator.user(),
llm,
tts,
ta,
transport.output(),
context_aggregator.assistant(),
]
)
task = PipelineTask(
pipeline,
params=PipelineParams(
enable_metrics=True,
enable_usage_metrics=True,
),
observers=[RTVIObserver(rtvi)],
)
await task.queue_frame(quiet_frame)
@rtvi.event_handler("on_client_ready")
async def on_client_ready(rtvi):
await rtvi.set_bot_ready()
# Kick off the conversation
await task.queue_frames([context_aggregator.user().get_context_frame()])
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await transport.capture_participant_transcription(participant["id"])
@transport.event_handler("on_participant_left")
async def on_participant_left(transport, participant, reason):
print(f"Participant left: {participant}")
await task.cancel()
runner = PipelineRunner()
await runner.run(task)

View File

@@ -0,0 +1,84 @@
#
# Copyright (c) 20242025, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import argparse
import asyncio
import importlib
import os
def get_bot_file(arg_bot: str | None) -> str:
bot_implementation = arg_bot or os.getenv("BOT_IMPLEMENTATION", "openai").lower().strip()
if not bot_implementation:
bot_implementation = "openai"
if bot_implementation not in ["openai", "gemini", "vllm"]:
raise ValueError(
f"Invalid BOT_IMPLEMENTATION: {bot_implementation}. Must be 'openai' or 'gemini'"
)
return f"bot_{bot_implementation}"
def get_runner(bot_file: str):
"""Dynamically import the run_bot function based on the bot name.
Args:
bot_name (str): The name of the bot implementation (e.g., 'openai', 'gemini').
Returns:
function: The run_bot function from the specified bot module.
Raises:
ImportError: If the specified bot module or run_bot function is not found.
"""
try:
# Dynamically construct the module name
module_name = f"{bot_file}"
# Import the module
module = importlib.import_module(module_name)
# Get the run_bot function from the module
return getattr(module, "run_bot")
except (ImportError, AttributeError) as e:
raise ImportError(f"Failed to import run_bot from {module_name}: {e}")
def main():
"""Parse the args to launch the appropriate bot using the given room/token."""
parser = argparse.ArgumentParser(description="Daily AI SDK Bot Sample")
parser.add_argument(
"-u", "--url", type=str, required=False, help="URL of the Daily room to join"
)
parser.add_argument(
"-t",
"--token",
type=str,
required=False,
help="Daily room token",
)
parser.add_argument(
"-b",
"--bot",
type=str,
required=False,
help="Bot runner to use (e.g., openai, gemini)",
)
args, unknown = parser.parse_known_args()
url = args.url or os.getenv("DAILY_SAMPLE_ROOM_URL")
token = args.token or os.getenv("DAILY_SAMPLE_ROOM_TOKEN")
bot_file = get_bot_file(args.bot)
if not url:
raise Exception(
"No Daily room specified. use the -u/--url option from the command line, or set DAILY_SAMPLE_ROOM_URL in your environment to specify a Daily room URL."
)
run_bot = get_runner(bot_file)
asyncio.run(run_bot(url, token))
if __name__ == "__main__":
main()

View File

@@ -100,7 +100,28 @@ phone numbers with valid values for your use case.
### Dialin Request
The server will receive a request when a call is received from Daily.
The server will receive a request when a call is received from Daily.
The payload that the webhook received is as follows:
```json
{
// for dial-in from webhook
"To": "+14152251493",
"From": "+14158483432",
"callId": "string-contains-uuid",
"callDomain": "string-contains-uuid",
"sipHeaders": {
"X-My-Custom-Header": "value",
"x-caller": "+1234567890",
"x-called": "+1987654321",
},
}
```
The `To`, `From`, `callId`, `callDomain` fields are converted to
`snake_case` and mapped to `dialin_settings`. In addition, `sipHeader`
contains any custom SIP headers received by Daily on the SIP
interconnect address (`sip_uri`). These are headers sent from
Twilio or other external SIP platforms, for example, to send the
caller's phone number.
### Dialout Request
@@ -158,6 +179,7 @@ curl -X POST http://localhost:3000/api/dial \
"From": "+1987654321",
"callId": "call-uuid-123",
"callDomain": "domain-uuid-456",
"sipHeader": {},
"dialout_settings": [
{
"phoneNumber": "+1234567890",

View File

@@ -39,6 +39,11 @@ class RoomRequest(BaseModel):
None, description="A flag to perform voicemail or answeing-machine detection"
)
call_transfer: Optional[Dict[str, Any]] = Field(None, description="to initiate a call transfer")
sipHeaders: Optional[Dict[str, Any]] = Field(
None,
alias="sip_headers",
description="Custom SIP headers received from the external SIP provider",
)
class Config:
populate_by_name = True
@@ -57,6 +62,14 @@ class RoomRequest(BaseModel):
"callDomain": "string-contains-uuid"
These need to be remapped to dialin_settings
In addition, we may receive in the body that can be
sent to the bot as a custom field, sip_headers
"sipHeaders": {
"X-My-Custom-Header": "value",
"x-caller": "+14158483432",
"x-called": "+14152251493",
},
"dialout_settings": [
{"phoneNumber": "+14158483432", "callerId": "+14152251493"},
{"sipUri": "sip:username@sip.hostname"}
@@ -157,6 +170,7 @@ async def dial(request: RoomRequest, raw_request: Request):
"dialout_settings": request.dialout_settings,
"voicemail_detection": request.voicemail_detection,
"call_transfer": request.call_transfer,
"sip_headers": request.sipHeaders, # passing the SIP headers to the bot
},
}

View File

@@ -65,6 +65,7 @@ export default async function handler(req, res) {
From,
callId,
callDomain,
sipHeaders,
dialout_settings,
voicemail_detection,
call_transfer
@@ -117,6 +118,7 @@ export default async function handler(req, res) {
dialout_settings,
voicemail_detection,
call_transfer,
sip_headers: sipHeaders,
},
};

View File

@@ -67,10 +67,8 @@ async def main(transport: DailyTransport):
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)

View File

@@ -192,7 +192,6 @@ async def main(transport: DailyTransport):
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),

View File

@@ -16,23 +16,25 @@ from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineTask
from pipecat.services.piper.tts import PiperTTSService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(audio_out_enabled=True),
"twilio": lambda: FastAPIWebsocketParams(audio_out_enabled=True),
"webrtc": lambda: TransportParams(audio_out_enabled=True),
}
# Create a transport using the WebRTC connection
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_out_enabled=True,
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
# Create an HTTP session
async with aiohttp.ClientSession() as session:
@@ -47,12 +49,12 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
async def on_client_connected(transport, client):
await task.queue_frames([TTSSpeakFrame(f"Hello there!"), EndFrame()])
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -16,24 +16,25 @@ from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineTask
from pipecat.services.rime.tts import RimeHttpTTSService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(audio_out_enabled=True),
"twilio": lambda: FastAPIWebsocketParams(audio_out_enabled=True),
"webrtc": lambda: TransportParams(audio_out_enabled=True),
}
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
# Create a transport using the WebRTC connection
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_out_enabled=True,
),
)
# Create an HTTP session
async with aiohttp.ClientSession() as session:
tts = RimeHttpTTSService(
@@ -49,12 +50,12 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
async def on_client_connected(transport, client):
await task.queue_frames([TTSSpeakFrame(f"Hello there!"), EndFrame()])
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -15,23 +15,25 @@ from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineTask
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(audio_out_enabled=True),
"twilio": lambda: FastAPIWebsocketParams(audio_out_enabled=True),
"webrtc": lambda: TransportParams(audio_out_enabled=True),
}
# Create a transport using the WebRTC connection
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_out_enabled=True,
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
@@ -45,12 +47,12 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
async def on_client_connected(transport, client):
await task.queue_frames([TTSSpeakFrame(f"Hello there!"), EndFrame()])
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -77,37 +77,36 @@ async def configure_livekit():
async def main():
async with aiohttp.ClientSession() as session:
(url, token, room_name) = await configure_livekit()
(url, token, room_name) = await configure_livekit()
transport = LiveKitTransport(
url=url,
token=token,
room_name=room_name,
params=LiveKitParams(audio_out_enabled=True),
)
transport = LiveKitTransport(
url=url,
token=token,
room_name=room_name,
params=LiveKitParams(audio_out_enabled=True),
)
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="71a7ad14-091c-4e8e-a314-022ece01c121", # British Reading Lady
)
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="71a7ad14-091c-4e8e-a314-022ece01c121", # British Reading Lady
)
runner = PipelineRunner()
runner = PipelineRunner()
task = PipelineTask(Pipeline([tts, transport.output()]))
task = PipelineTask(Pipeline([tts, transport.output()]))
# Register an event handler so we can play the audio when the
# participant joins.
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant_id):
await asyncio.sleep(1)
await task.queue_frame(
TextFrame(
"Hello there! How are you doing today? Would you like to talk about the weather?"
)
# Register an event handler so we can play the audio when the
# participant joins.
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant_id):
await asyncio.sleep(1)
await task.queue_frame(
TextFrame(
"Hello there! How are you doing today? Would you like to talk about the weather?"
)
)
await runner.run(task)
await runner.run(task)
if __name__ == "__main__":

View File

@@ -15,23 +15,25 @@ from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineTask
from pipecat.services.riva.tts import FastPitchTTSService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(audio_out_enabled=True),
"twilio": lambda: FastAPIWebsocketParams(audio_out_enabled=True),
"webrtc": lambda: TransportParams(audio_out_enabled=True),
}
# Create a transport using the WebRTC connection
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_out_enabled=True,
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
tts = FastPitchTTSService(api_key=os.getenv("NVIDIA_API_KEY"))
@@ -42,12 +44,12 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
async def on_client_connected(transport, client):
await task.queue_frames([TTSSpeakFrame(f"Hello there!"), EndFrame()])
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -16,23 +16,25 @@ from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineTask
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(audio_out_enabled=True),
"twilio": lambda: FastAPIWebsocketParams(audio_out_enabled=True),
"webrtc": lambda: TransportParams(audio_out_enabled=True),
}
# Create a transport using the WebRTC connection
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_out_enabled=True,
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
@@ -55,12 +57,12 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
async def on_client_connected(transport, client):
await task.queue_frames([LLMMessagesFrame(messages), EndFrame()])
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -16,25 +16,31 @@ from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineTask
from pipecat.services.fal.image import FalImageGenService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
video_out_enabled=True,
video_out_width=1024,
video_out_height=1024,
),
"webrtc": lambda: TransportParams(
video_out_enabled=True,
video_out_width=1024,
video_out_height=1024,
),
}
# Create a transport using the WebRTC connection
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
video_out_enabled=True,
video_out_width=1024,
video_out_height=1024,
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
# Create an HTTP session
async with aiohttp.ClientSession() as session:
@@ -54,18 +60,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -15,25 +15,31 @@ from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.services.google.image import GoogleImageGenService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
video_out_enabled=True,
video_out_width=1024,
video_out_height=1024,
),
"webrtc": lambda: TransportParams(
video_out_enabled=True,
video_out_width=1024,
video_out_height=1024,
),
}
# Create a transport using the WebRTC connection
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
video_out_enabled=True,
video_out_width=1024,
video_out_height=1024,
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
imagegen = GoogleImageGenService(
api_key=os.getenv("GOOGLE_API_KEY"),
@@ -41,7 +47,10 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
Pipeline([imagegen, transport.output()]),
params=PipelineParams(enable_metrics=True),
params=PipelineParams(
enable_metrics=True,
enable_usage_metrics=True,
),
)
# Register an event handler so we can play the audio when the client joins
@@ -54,18 +63,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -5,10 +5,17 @@
#
import argparse
import asyncio
import os
from contextlib import asynccontextmanager
from typing import Dict
import uvicorn
from dotenv import load_dotenv
from fastapi import BackgroundTasks, FastAPI
from fastapi.responses import RedirectResponse
from loguru import logger
from pipecat_ai_small_webrtc_prebuilt.frontend import SmallWebRTCPrebuiltUI
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.pipeline.pipeline import Pipeline
@@ -20,14 +27,29 @@ from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.network.webrtc_connection import IceServer, SmallWebRTCConnection
load_dotenv(override=True)
app = FastAPI()
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
# Store connections by pc_id
pcs_map: Dict[str, SmallWebRTCConnection] = {}
ice_servers = [
IceServer(
urls="stun:stun.l.google.com:19302",
)
]
# Mount the frontend at /
app.mount("/client", SmallWebRTCPrebuiltUI)
async def run_example(webrtc_connection: SmallWebRTCConnection):
logger.info(f"Starting bot")
# Create a transport using the WebRTC connection
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
@@ -71,10 +93,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -88,10 +108,6 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
@@ -99,7 +115,58 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
await runner.run(task)
if __name__ == "__main__":
from run import main
@app.get("/", include_in_schema=False)
async def root_redirect():
return RedirectResponse(url="/client/")
main()
@app.post("/api/offer")
async def offer(request: dict, background_tasks: BackgroundTasks):
pc_id = request.get("pc_id")
if pc_id and pc_id in pcs_map:
pipecat_connection = pcs_map[pc_id]
logger.info(f"Reusing existing connection for pc_id: {pc_id}")
await pipecat_connection.renegotiate(
sdp=request["sdp"],
type=request["type"],
restart_pc=request.get("restart_pc", False),
)
else:
pipecat_connection = SmallWebRTCConnection(ice_servers)
await pipecat_connection.initialize(sdp=request["sdp"], type=request["type"])
@pipecat_connection.event_handler("closed")
async def handle_disconnected(webrtc_connection: SmallWebRTCConnection):
logger.info(f"Discarding peer connection for pc_id: {webrtc_connection.pc_id}")
pcs_map.pop(webrtc_connection.pc_id, None)
# Run example function with SmallWebRTC transport arguments.
background_tasks.add_task(run_example, pipecat_connection)
answer = pipecat_connection.get_answer()
# Updating the peer connection inside the map
pcs_map[answer["pc_id"]] = pipecat_connection
return answer
@asynccontextmanager
async def lifespan(app: FastAPI):
yield # Run app
coros = [pc.disconnect() for pc in pcs_map.values()]
await asyncio.gather(*coros)
pcs_map.clear()
if __name__ == "__main__":
parser = argparse.ArgumentParser(description="Pipecat Bot Runner")
parser.add_argument(
"--host", default="localhost", help="Host for HTTP server (default: localhost)"
)
parser.add_argument(
"--port", type=int, default=7860, help="Port for HTTP server (default: 7860)"
)
args = parser.parse_args()
uvicorn.run(app, host=args.host, port=args.port)

View File

@@ -9,11 +9,11 @@ import os
import sys
import aiohttp
from daily_runner import configure
from dotenv import load_dotenv
from loguru import logger
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.examples.daily_runner import configure
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
@@ -37,9 +37,9 @@ async def main():
token,
"Respond bot",
DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
transcription_enabled=True,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
@@ -75,10 +75,8 @@ async def main():
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)

View File

@@ -10,7 +10,6 @@ import json
import os
import sys
import aiohttp
from deepgram import LiveOptions
from dotenv import load_dotenv
from livekit import api
@@ -104,101 +103,101 @@ async def configure_livekit():
async def main():
async with aiohttp.ClientSession() as session:
(url, token, room_name) = await configure_livekit()
(url, token, room_name) = await configure_livekit()
transport = LiveKitTransport(
url=url,
token=token,
room_name=room_name,
params=LiveKitParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
transport = LiveKitTransport(
url=url,
token=token,
room_name=room_name,
params=LiveKitParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
stt = DeepgramSTTService(
api_key=os.getenv("DEEPGRAM_API_KEY"),
live_options=LiveOptions(
vad_events=True,
),
)
stt = DeepgramSTTService(
api_key=os.getenv("DEEPGRAM_API_KEY"),
live_options=LiveOptions(
vad_events=True,
),
)
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="71a7ad14-091c-4e8e-a314-022ece01c121", # British Reading Lady
)
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="71a7ad14-091c-4e8e-a314-022ece01c121", # British Reading Lady
)
messages = [
{
"role": "system",
"content": "You are a helpful LLM in a WebRTC call. "
"Your goal is to demonstrate your capabilities in a succinct way. "
"Your output will be converted to audio so don't include special characters in your answers. "
"Respond to what the user said in a creative and helpful way.",
},
]
messages = [
{
"role": "system",
"content": "You are a helpful LLM in a WebRTC call. "
"Your goal is to demonstrate your capabilities in a succinct way. "
"Your output will be converted to audio so don't include special characters in your answers. "
"Respond to what the user said in a creative and helpful way.",
},
]
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
runner = PipelineRunner()
runner = PipelineRunner()
task = PipelineTask(
Pipeline(
[
transport.input(),
stt,
context_aggregator.user(),
llm,
tts,
transport.output(),
context_aggregator.assistant(),
],
),
params=PipelineParams(
allow_interruptions=True, enable_metrics=True, enable_usage_metrics=True
),
)
task = PipelineTask(
Pipeline(
[
transport.input(),
stt,
context_aggregator.user(),
llm,
tts,
transport.output(),
context_aggregator.assistant(),
],
),
params=PipelineParams(
enable_metrics=True,
enable_usage_metrics=True,
),
)
# Register an event handler so we can play the audio when the
# participant joins.
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant_id):
await asyncio.sleep(1)
await task.queue_frame(
TextFrame(
"Hello there! How are you doing today? Would you like to talk about the weather?"
)
# Register an event handler so we can play the audio when the
# participant joins.
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant_id):
await asyncio.sleep(1)
await task.queue_frame(
TextFrame(
"Hello there! How are you doing today? Would you like to talk about the weather?"
)
)
# Register an event handler to receive data from the participant via text chat
# in the LiveKit room. This will be used to as transcription frames and
# interrupt the bot and pass it to llm for processing and
# then pass back to the participant as audio output.
@transport.event_handler("on_data_received")
async def on_data_received(transport, data, participant_id):
logger.info(f"Received data from participant {participant_id}: {data}")
# convert data from bytes to string
json_data = json.loads(data)
# Register an event handler to receive data from the participant via text chat
# in the LiveKit room. This will be used to as transcription frames and
# interrupt the bot and pass it to llm for processing and
# then pass back to the participant as audio output.
@transport.event_handler("on_data_received")
async def on_data_received(transport, data, participant_id):
logger.info(f"Received data from participant {participant_id}: {data}")
# convert data from bytes to string
json_data = json.loads(data)
await task.queue_frames(
[
BotInterruptionFrame(),
UserStartedSpeakingFrame(),
TranscriptionFrame(
user_id=participant_id,
timestamp=json_data["timestamp"],
text=json_data["message"],
),
UserStoppedSpeakingFrame(),
],
)
await task.queue_frames(
[
BotInterruptionFrame(),
UserStartedSpeakingFrame(),
TranscriptionFrame(
user_id=participant_id,
timestamp=json_data["timestamp"],
text=json_data["message"],
),
UserStoppedSpeakingFrame(),
],
)
await runner.run(task)
await runner.run(task)
if __name__ == "__main__":

View File

@@ -28,9 +28,8 @@ from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.services.cartesia.tts import CartesiaHttpTTSService
from pipecat.services.fal.image import FalImageGenService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
@@ -64,7 +63,26 @@ class MonthPrepender(FrameProcessor):
await self.push_frame(frame, direction)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_out_enabled=True,
video_out_enabled=True,
video_out_width=1024,
video_out_height=1024,
),
"webrtc": lambda: TransportParams(
audio_out_enabled=True,
video_out_enabled=True,
video_out_width=1024,
video_out_height=1024,
),
}
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
"""Run the Calendar Month Narration bot using WebRTC transport.
Args:
@@ -73,17 +91,6 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
"""
logger.info(f"Starting bot")
# Create a transport using the WebRTC connection
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_out_enabled=True,
video_out_enabled=True,
video_out_width=1024,
video_out_height=1024,
),
)
# Create an HTTP session for API calls
async with aiohttp.ClientSession() as session:
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
@@ -159,18 +166,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
# Run the pipeline
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -26,9 +26,9 @@ from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
@@ -53,17 +53,30 @@ class MetricsLogger(FrameProcessor):
await self.push_frame(frame, direction)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
@@ -117,17 +130,13 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -26,9 +26,8 @@ from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
@@ -68,20 +67,31 @@ class ImageSyncAggregator(FrameProcessor):
await self.push_frame(frame)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
video_out_enabled=True,
video_out_width=1024,
video_out_height=1024,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
video_out_enabled=True,
video_out_width=1024,
video_out_height=1024,
vad_analyzer=SileroVADAnalyzer(),
),
}
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
video_out_enabled=True,
video_out_width=1024,
video_out_height=1024,
vad_analyzer=SileroVADAnalyzer(),
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
@@ -123,10 +133,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -139,17 +147,13 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -10,37 +10,49 @@ import os
from dotenv import load_dotenv
from loguru import logger
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.processors.audio.vad.silero import SileroVAD
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.cartesia.tts import CartesiaHttpTTSService
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
vad = SileroVAD()
tts = CartesiaTTSService(
tts = CartesiaHttpTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="71a7ad14-091c-4e8e-a314-022ece01c121", # British Reading Lady
)
@@ -59,24 +71,21 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
pipeline = Pipeline(
[
transport.input(),
transport.input(), # Transport user input
stt,
vad,
context_aggregator.user(),
llm,
tts,
transport.output(),
context_aggregator.assistant(),
context_aggregator.user(), # User responses
llm, # LLM
tts, # TTS
transport.output(), # Transport bot output
context_aggregator.assistant(), # Assistant spoken responses
]
)
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -90,18 +99,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -18,25 +18,37 @@ from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
tts = CartesiaTTSService(
@@ -71,10 +83,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -88,18 +98,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -27,9 +27,9 @@ from pipecat.processors.aggregators.llm_response import (
from pipecat.processors.frameworks.langchain import LangchainProcessor
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
@@ -43,17 +43,30 @@ def get_session_history(session_id: str) -> BaseChatMessageHistory:
return message_store[session_id]
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
@@ -100,10 +113,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -120,18 +131,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -24,23 +24,34 @@ from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.deepgram.tts import DeepgramTTSService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
),
}
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
stt = DeepgramSTTService(
api_key=os.getenv("DEEPGRAM_API_KEY"),
@@ -76,10 +87,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -101,18 +110,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -18,24 +18,37 @@ from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.deepgram.tts import DeepgramTTSService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
@@ -68,10 +81,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -85,18 +96,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -19,24 +19,37 @@ from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.elevenlabs.tts import ElevenLabsHttpTTSService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
# Create an HTTP session
async with aiohttp.ClientSession() as session:
@@ -75,10 +88,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -92,18 +103,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -18,24 +18,37 @@ from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.elevenlabs.tts import ElevenLabsTTSService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
logger.info(f"Starting bot")
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
@@ -71,10 +84,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -88,18 +99,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -18,25 +18,37 @@ from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.services.playht.tts import PlayHTHttpTTSService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
tts = PlayHTHttpTTSService(
@@ -72,10 +84,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -89,18 +99,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -19,25 +19,37 @@ from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.services.playht.tts import PlayHTTTSService
from pipecat.transcriptions.language import Language
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
tts = PlayHTTTSService(
@@ -74,10 +86,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -91,18 +101,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -18,25 +18,37 @@ from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.azure.llm import AzureLLMService
from pipecat.services.azure.stt import AzureSTTService
from pipecat.services.azure.tts import AzureTTSService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
stt = AzureSTTService(
api_key=os.getenv("AZURE_SPEECH_API_KEY"),
region=os.getenv("AZURE_SPEECH_REGION"),
@@ -78,10 +90,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -95,18 +105,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -18,25 +18,37 @@ from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.services.openai.stt import OpenAISTTService
from pipecat.services.openai.tts import OpenAITTSService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
stt = OpenAISTTService(
api_key=os.getenv("OPENAI_API_KEY"),
model="gpt-4o-transcribe",
@@ -72,11 +84,9 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
audio_out_sample_rate=24000,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -90,18 +100,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -19,25 +19,37 @@ from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.openpipe.llm import OpenPipeLLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
tts = CartesiaTTSService(
@@ -77,10 +89,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -94,18 +104,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -19,25 +19,37 @@ from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.services.xtts.tts import XTTSService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
# Create an HTTP session
async with aiohttp.ClientSession() as session:
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
@@ -75,10 +87,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -92,18 +102,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -20,25 +20,37 @@ from pipecat.services.gladia.config import GladiaInputParams, LanguageConfig
from pipecat.services.gladia.stt import GladiaSTTService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transcriptions.language import Language
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
stt = GladiaSTTService(
api_key=os.getenv("GLADIA_API_KEY", ""),
params=GladiaInputParams(
@@ -80,10 +92,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -97,17 +107,13 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -18,25 +18,37 @@ from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.lmnt.tts import LmntTTSService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
tts = LmntTTSService(api_key=os.getenv("LMNT_API_KEY"), voice_id="morgan")
@@ -68,10 +80,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -85,18 +95,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -19,25 +19,37 @@ from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.groq.llm import GroqLLMService
from pipecat.services.groq.stt import GroqSTTService
from pipecat.services.groq.tts import GroqTTSService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
stt = GroqSTTService(api_key=os.getenv("GROQ_API_KEY"))
llm = GroqLLMService(
@@ -73,7 +85,6 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),
@@ -89,18 +100,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -17,25 +17,37 @@ from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.aws.llm import AWSBedrockLLMService
from pipecat.services.aws.stt import AWSTranscribeSTTService
from pipecat.services.aws.tts import AWSPollyTTSService
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
stt = AWSTranscribeSTTService()
tts = AWSPollyTTSService(
@@ -75,10 +87,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -92,18 +102,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

View File

@@ -19,25 +19,37 @@ from pipecat.services.google.llm import GoogleLLMService
from pipecat.services.google.stt import GoogleSTTService
from pipecat.services.google.tts import GoogleTTSService
from pipecat.transcriptions.language import Language
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.network.small_webrtc import SmallWebRTCTransport
from pipecat.transports.network.webrtc_connection import SmallWebRTCConnection
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.network.fastapi_websocket import FastAPIWebsocketParams
from pipecat.transports.services.daily import DailyParams
load_dotenv(override=True)
# We store functions so objects (e.g. SileroVADAnalyzer) don't get
# instantiated. The function will be called when the desired transport gets
# selected.
transport_params = {
"daily": lambda: DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"twilio": lambda: FastAPIWebsocketParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
"webrtc": lambda: TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
}
async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespace):
async def run_example(transport: BaseTransport, _: argparse.Namespace, handle_sigint: bool):
logger.info(f"Starting bot")
transport = SmallWebRTCTransport(
webrtc_connection=webrtc_connection,
params=TransportParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
stt = GoogleSTTService(
params=GoogleSTTService.InputParams(languages=Language.EN_US),
credentials=os.getenv("GOOGLE_TEST_CREDENTIALS"),
@@ -76,10 +88,8 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
task = PipelineTask(
pipeline,
params=PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
report_only_initial_ttfb=True,
),
)
@@ -93,18 +103,14 @@ async def run_bot(webrtc_connection: SmallWebRTCConnection, _: argparse.Namespac
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
@transport.event_handler("on_client_closed")
async def on_client_closed(transport, client):
logger.info(f"Client closed connection")
await task.cancel()
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=handle_sigint)
await runner.run(task)
if __name__ == "__main__":
from run import main
from pipecat.examples.run import main
main()
main(run_example, transport_params=transport_params)

Some files were not shown because too many files have changed in this diff Show More