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Changelog

All notable changes to Pipecat will be documented in this file.

The format is based on Keep a Changelog, and this project adheres to Semantic Versioning.

[0.0.52] - 2024-12-24

Added

  • Constructor arguments for GoogleLLMService to directly set tools and tool_config.

  • Smart turn detection example (22d-natural-conversation-gemini-audio.py) that leverages Gemini 2.0 capabilities (). (see https://x.com/kwindla/status/1870974144831275410)

  • Added DailyTransport.send_dtmf() to send dial-out DTMF tones.

  • Added DailyTransport.sip_call_transfer() to forward SIP and PSTN calls to another address or number. For example, transfer a SIP call to a different SIP address or transfer a PSTN phone number to a different PSTN phone number.

  • Added DailyTransport.sip_refer() to transfer incoming SIP/PSTN calls from outside Daily to another SIP/PSTN address.

  • Added an auto_mode input parameter to ElevenLabsTTSService. auto_mode is set to True by default. Enabling this setting disables the chunk schedule and all buffers, which reduces latency.

  • Added KoalaFilter which implement on device noise reduction using Koala Noise Suppression. (see https://picovoice.ai/platform/koala/)

  • Added CerebrasLLMService for Cerebras integration with an OpenAI-compatible interface. Added foundational example 14k-function-calling-cerebras.py.

  • Pipecat now supports Python 3.13. We had a dependency on the audioop package which was deprecated and now removed on Python 3.13. We are now using audioop-lts (https://github.com/AbstractUmbra/audioop) to provide the same functionality.

  • Added timestamped conversation transcript support:

    • New TranscriptProcessor factory provides access to user and assistant transcript processors.
    • UserTranscriptProcessor processes user speech with timestamps from transcription.
    • AssistantTranscriptProcessor processes assistant responses with LLM context timestamps.
    • Messages emitted with ISO 8601 timestamps indicating when they were spoken.
    • Supports all LLM formats (OpenAI, Anthropic, Google) via standard message format.
    • New examples: 28a-transcription-processor-openai.py, 28b-transcription-processor-anthropic.py, and 28c-transcription-processor-gemini.py.

  • Add support for more languages to ElevenLabs (Arabic, Croatian, Filipino, Tamil) and PlayHT (Afrikans, Albanian, Amharic, Arabic, Bengali, Croatian, Galician, Hebrew, Mandarin, Serbian, Tagalog, Urdu, Xhosa).

Changed

  • PlayHTTTSService uses the new v4 websocket API, which also fixes an issue where text inputted to the TTS didn't return audio.

  • The default model for ElevenLabsTTSService is now eleven_flash_v2_5.

  • OpenAIRealtimeBetaLLMService now takes a model parameter in the constructor.

  • Updated the default model for the OpenAIRealtimeBetaLLMService.

  • Room expiration (exp) in DailyRoomProperties is now optional (None) by default instead of automatically setting a 5-minute expiration time. You must explicitly set expiration time if desired.

Deprecated

  • AWSTTSService is now deprecated, use PollyTTSService instead.

Fixed

  • Fixed token counting in GoogleLLMService. Tokens were summed incorrectly (double-counted in many cases).

  • Fixed an issue that could cause the bot to stop talking if there was a user interruption before getting any audio from the TTS service.

  • Fixed an issue that would cause ParallelPipeline to handle EndFrame incorrectly causing the main pipeline to not terminate or terminate too early.

  • Fixed an audio stuttering issue in FastPitchTTSService.

  • Fixed a BaseOutputTransport issue that was causing non-audio frames being processed before the previous audio frames were played. This will allow, for example, sending a frame A after a TTSSpeakFrame and the frame A will only be pushed downstream after the audio generated from TTSSpeakFrame has been spoken.

  • Fixed a DeepgramSTTService issue that was causing language to be passed as an object instead of a string resulting in the connection to fail.

[0.0.51] - 2024-12-16

Fixed

  • Fixed an issue in websocket-based TTS services that was causing infinite reconnections (Cartesia, ElevenLabs, PlayHT and LMNT).

[0.0.50] - 2024-12-11

Added

  • Added GeminiMultimodalLiveLLMService. This is an integration for Google's Gemini Multimodal Live API, supporting:

    • Real-time audio and video input processing
    • Streaming text responses with TTS
    • Audio transcription for both user and bot speech
    • Function calling
    • System instructions and context management
    • Dynamic parameter updates (temperature, top_p, etc.)

  • Added AudioTranscriber utility class for handling audio transcription with Gemini models.

  • Added new context classes for Gemini:

    • GeminiMultimodalLiveContext
    • GeminiMultimodalLiveUserContextAggregator
    • GeminiMultimodalLiveAssistantContextAggregator
    • GeminiMultimodalLiveContextAggregatorPair

  • Added new foundational examples for GeminiMultimodalLiveLLMService:

    • 26-gemini-multimodal-live.py
    • 26a-gemini-multimodal-live-transcription.py
    • 26b-gemini-multimodal-live-video.py
    • 26c-gemini-multimodal-live-video.py

  • Added SimliVideoService. This is an integration for Simli AI avatars. (see https://www.simli.com)

  • Added NVIDIA Riva's FastPitchTTSService and ParakeetSTTService. (see https://www.nvidia.com/en-us/ai-data-science/products/riva/)

  • Added IdentityFilter. This is the simplest frame filter that lets through all incoming frames.

  • New STTMuteStrategy called FUNCTION_CALL which mutes the STT service during LLM function calls.

  • DeepgramSTTService now exposes two event handlers on_speech_started and on_utterance_end that could be used to implement interruptions. See new example examples/foundational/07c-interruptible-deepgram-vad.py.

  • Added GroqLLMService, GrokLLMService, and NimLLMService for Groq, Grok, and NVIDIA NIM API integration, with an OpenAI-compatible interface.

  • New examples demonstrating function calling with Groq, Grok, Azure OpenAI, Fireworks, and NVIDIA NIM: 14f-function-calling-groq.py, 14g-function-calling-grok.py, 14h-function-calling-azure.py, 14i-function-calling-fireworks.py, and 14j-function-calling-nvidia.py.

  • In order to obtain the audio stored by the AudioBufferProcessor you can now also register an on_audio_data event handler. The on_audio_data handler will be called every time buffer_size (a new constructor argument) is reached. If buffer_size is 0 (default) you need to manually get the audio as before using AudioBufferProcessor.merge_audio_buffers().

@audiobuffer.event_handler("on_audio_data")
async def on_audio_data(processor, audio, sample_rate, num_channels):
    await save_audio(audio, sample_rate, num_channels)
  • Added a new RTVI message called disconnect-bot, which when handled pushes an EndFrame to trigger the pipeline to stop.

Changed

  • STTMuteFilter now supports multiple simultaneous muting strategies.

  • XTTSService language now defaults to Language.EN.

  • SoundfileMixer doesn't resample input files anymore to avoid startup delays. The sample rate of the provided sound files now need to match the sample rate of the output transport.

  • Input frames (audio, image and transport messages) are now system frames. This means they are processed immediately by all processors instead of being queued internally.

  • Expanded the transcriptions.language module to support a superset of languages.

  • Updated STT and TTS services with language options that match the supported languages for each service.

  • Updated the AzureLLMService to use the OpenAILLMService. Updated the api_version to 2024-09-01-preview.

  • Updated the FireworksLLMService to use the OpenAILLMService. Updated the default model to accounts/fireworks/models/firefunction-v2.

  • Updated the simple-chatbot example to include a Javascript and React client example, using RTVI JS and React.

Removed

  • Removed AppFrame. This was used as a special user custom frame, but there's actually no use case for that.

Fixed

  • Fixed a ParallelPipeline issue that would cause system frames to be queued.

  • Fixed FastAPIWebsocketTransport so it can work with binary data (e.g. using the protobuf serializer).

  • Fixed an issue in CartesiaTTSService that could cause previous audio to be received after an interruption.

  • Fixed Cartesia, ElevenLabs, LMNT and PlayHT TTS websocket reconnection. Before, if an error occurred no reconnection was happening.

  • Fixed a BaseOutputTransport issue that was causing audio to be discarded after an EndFrame was received.

  • Fixed an issue in WebsocketServerTransport and FastAPIWebsocketTransport that would cause a busy loop when using audio mixer.

  • Fixed a DailyTransport and LiveKitTransport issue where connections were being closed in the input transport prematurely. This was causing frames queued inside the pipeline being discarded.

  • Fixed an issue in DailyTransport that would cause some internal callbacks to not be executed.

  • Fixed an issue where other frames were being processed while a CancelFrame was being pushed down the pipeline.

  • AudioBufferProcessor now handles interruptions properly.

  • Fixed a WebsocketServerTransport issue that would prevent interruptions with TwilioSerializer from working.

  • DailyTransport.capture_participant_video now allows capturing user's screen share by simply passing video_source="screenVideo".

  • Fixed Google Gemini message handling to properly convert appended messages to Gemini's required format.

  • Fixed an issue with FireworksLLMService where chat completions were failing by removing the stream_options from the chat completion options.

[0.0.49] - 2024-11-17

Added

  • Added RTVI on_bot_started event which is useful in a single turn interaction.

  • Added DailyTransport events dialin-connected, dialin-stopped, dialin-error and dialin-warning. Needs daily-python >= 0.13.0.

  • Added RimeHttpTTSService and the 07q-interruptible-rime.py foundational example.

  • Added STTMuteFilter, a general-purpose processor that combines STT muting and interruption control. When active, it prevents both transcription and interruptions during bot speech. The processor supports multiple strategies: FIRST_SPEECH (mute only during bot's first speech), ALWAYS (mute during all bot speech), or CUSTOM (using provided callback).

  • Added STTMuteFrame, a control frame that enables/disables speech transcription in STT services.

[0.0.48] - 2024-11-10 "Antonio release"

Added

  • There's now an input queue in each frame processor. When you call FrameProcessor.push_frame() this will internally call FrameProcessor.queue_frame() on the next processor (upstream or downstream) and the frame will be internally queued (except system frames). Then, the queued frames will get processed. With this input queue it is also possible for FrameProcessors to block processing more frames by calling FrameProcessor.pause_processing_frames(). The way to resume processing frames is by calling FrameProcessor.resume_processing_frames().

  • Added audio filter NoisereduceFilter.

  • Introduce input transport audio filters (BaseAudioFilter). Audio filters can be used to remove background noises before audio is sent to VAD.

  • Introduce output transport audio mixers (BaseAudioMixer). Output transport audio mixers can be used, for example, to add background sounds or any other audio mixing functionality before the output audio is actually written to the transport.

  • Added GatedOpenAILLMContextAggregator. This aggregator keeps the last received OpenAI LLM context frame and it doesn't let it through until the notifier is notified.

  • Added WakeNotifierFilter. This processor expects a list of frame types and will execute a given callback predicate when a frame of any of those type is being processed. If the callback returns true the notifier will be notified.

  • Added NullFilter. A null filter doesn't push any frames upstream or downstream. This is usually used to disable one of the pipelines in ParallelPipeline.

  • Added EventNotifier. This can be used as a very simple synchronization feature between processors.

  • Added TavusVideoService. This is an integration for Tavus digital twins. (see https://www.tavus.io/)

  • Added DailyTransport.update_subscriptions(). This allows you to have fine grained control of what media subscriptions you want for each participant in a room.

  • Added audio filter KrispFilter.

Changed

  • The following DailyTransport functions are now async which means they need to be awaited: start_dialout, stop_dialout, start_recording, stop_recording, capture_participant_transcription and capture_participant_video.

  • Changed default output sample rate to 24000. This changes all TTS service to output to 24000 and also the default output transport sample rate. This improves audio quality at the cost of some extra bandwidth.

  • AzureTTSService now uses Azure websockets instead of HTTP requests.

  • The previous AzureTTSService HTTP implementation is now AzureHttpTTSService.

Fixed

  • Websocket transports (FastAPI and Websocket) now synchronize with time before sending data. This allows for interruptions to just work out of the box.

  • Improved bot speaking detection for all TTS services by using actual bot audio.

  • Fixed an issue that was generating constant bot started/stopped speaking frames for HTTP TTS services.

  • Fixed an issue that was causing stuttering with AWS TTS service.

  • Fixed an issue with PlayHTTTSService, where the TTFB metrics were reporting very small time values.

  • Fixed an issue where AzureTTSService wasn't initializing the specified language.

Other

  • Add 23-bot-background-sound.py foundational example.

  • Added a new foundational example 22-natural-conversation.py. This example shows how to achieve a more natural conversation detecting when the user ends statement.

[0.0.47] - 2024-10-22

Added

  • Added AssemblyAISTTService and corresponding foundational examples 07o-interruptible-assemblyai.py and 13d-assemblyai-transcription.py.

  • Added a foundational example for Gladia transcription: 13c-gladia-transcription.py

Changed

  • Updated GladiaSTTService to use the V2 API.

  • Changed DailyTransport transcription model to nova-2-general.

Fixed

  • Fixed an issue that would cause an import error when importing SileroVADAnalyzer from the old package pipecat.vad.silero.

  • Fixed enable_usage_metrics to control LLM/TTS usage metrics separately from enable_metrics.

[0.0.46] - 2024-10-19

Added

  • Added audio_passthrough parameter to STTService. If enabled it allows audio frames to be pushed downstream in case other processors need them.

  • Added input parameter options for PlayHTTTSService and PlayHTHttpTTSService.

Changed

  • Changed DeepgramSTTService model to nova-2-general.

  • Moved SileroVAD audio processor to processors.audio.vad.

  • Module utils.audio is now audio.utils. A new resample_audio function has been added.

  • PlayHTTTSService now uses PlayHT websockets instead of HTTP requests.

  • The previous PlayHTTTSService HTTP implementation is now PlayHTHttpTTSService.

  • PlayHTTTSService and PlayHTHttpTTSService now use a voice_engine of PlayHT3.0-mini, which allows for multi-lingual support.

  • Renamed OpenAILLMServiceRealtimeBeta to OpenAIRealtimeBetaLLMService to match other services.

Deprecated

  • LLMUserResponseAggregator and LLMAssistantResponseAggregator are mostly deprecated, use OpenAILLMContext instead.

  • The vad package is now deprecated and audio.vad should be used instead. The avd package will get removed in a future release.

Fixed

  • Fixed an issue that would cause an error if no VAD analyzer was passed to LiveKitTransport params.

  • Fixed SileroVAD processor to support interruptions properly.

Other

  • Added examples/foundational/07-interruptible-vad.py. This is the same as 07-interruptible.py but using the SileroVAD processor instead of passing the VADAnalyzer in the transport.

[0.0.45] - 2024-10-16

Changed

  • Metrics messages have moved out from the transport's base output into RTVI.

[0.0.44] - 2024-10-15

Added

  • Added support for OpenAI Realtime API with the new OpenAILLMServiceRealtimeBeta processor. (see https://platform.openai.com/docs/guides/realtime/overview)

  • Added RTVIBotTranscriptionProcessor which will send the RTVI bot-transcription protocol message. These are TTS text aggregated (into sentences) messages.

  • Added new input params to the MarkdownTextFilter utility. You can set filter_code to filter code from text and filter_tables to filter tables from text.

  • Added CanonicalMetricsService. This processor uses the new AudioBufferProcessor to capture conversation audio and later send it to Canonical AI. (see https://canonical.chat/)

  • Added AudioBufferProcessor. This processor can be used to buffer mixed user and bot audio. This can later be saved into an audio file or processed by some audio analyzer.

  • Added on_first_participant_joined event to LiveKitTransport.

Changed

  • LLM text responses are now logged properly as unicode characters.

  • UserStartedSpeakingFrame, UserStoppedSpeakingFrame, BotStartedSpeakingFrame, BotStoppedSpeakingFrame, BotSpeakingFrame and UserImageRequestFrame are now based from SystemFrame

Fixed

  • Merge RTVIBotLLMProcessor/RTVIBotLLMTextProcessor and RTVIBotTTSProcessor/RTVIBotTTSTextProcessor to avoid out of order issues.

  • Fixed an issue in RTVI protocol that could cause a bot-llm-stopped or bot-tts-stopped message to be sent before a bot-llm-text or bot-tts-text message.

  • Fixed DeepgramSTTService constructor settings not being merged with default ones.

  • Fixed an issue in Daily transport that would cause tasks to be hanging if urgent transport messages were being sent from a transport event handler.

  • Fixed an issue in BaseOutputTransport that would cause EndFrame to be pushed downed too early and call FrameProcessor.cleanup() before letting the transport stop properly.

[0.0.43] - 2024-10-10

Added

  • Added a new util called MarkdownTextFilter which is a subclass of a new base class called BaseTextFilter. This is a configurable utility which is intended to filter text received by TTS services.

  • Added new RTVIUserLLMTextProcessor. This processor will send an RTVI user-llm-text message with the user content's that was sent to the LLM.

Changed

  • TransportMessageFrame doesn't have an urgent field anymore, instead there's now a TransportMessageUrgentFrame which is a SystemFrame and therefore skip all internal queuing.

  • For TTS services, convert inputted languages to match each service's language format

Fixed

  • Fixed an issue where changing a language with the Deepgram STT service wouldn't apply the change. This was fixed by disconnecting and reconnecting when the language changes.

[0.0.42] - 2024-10-02

Added

  • SentryMetrics has been added to report frame processor metrics to Sentry. This is now possible because FrameProcessorMetrics can now be passed to FrameProcessor.

  • Added Google TTS service and corresponding foundational example 07n-interruptible-google.py

  • Added AWS Polly TTS support and 07m-interruptible-aws.py as an example.

  • Added InputParams to Azure TTS service.

  • Added LivekitTransport (audio-only for now).

  • RTVI 0.2.0 is now supported.

  • All FrameProcessors can now register event handlers.

tts = SomeTTSService(...)

@tts.event_handler("on_connected"):
async def on_connected(processor):
  ...

  • Added AsyncGeneratorProcessor. This processor can be used together with a FrameSerializer as an async generator. It provides a generator() function that returns an AsyncGenerator and that yields serialized frames.

  • Added EndTaskFrame and CancelTaskFrame. These are new frames that are meant to be pushed upstream to tell the pipeline task to stop nicely or immediately respectively.

  • Added configurable LLM parameters (e.g., temperature, top_p, max_tokens, seed) for OpenAI, Anthropic, and Together AI services along with corresponding setter functions.

  • Added sample_rate as a constructor parameter for TTS services.

  • Pipecat has a pipeline-based architecture. The pipeline consists of frame processors linked to each other. The elements traveling across the pipeline are called frames.

    To have a deterministic behavior the frames traveling through the pipeline should always be ordered, except system frames which are out-of-band frames. To achieve that, each frame processor should only output frames from a single task.

    In this version all the frame processors have their own task to push frames. That is, when push_frame() is called the given frame will be put into an internal queue (with the exception of system frames) and a frame processor task will push it out.

  • Added pipeline clocks. A pipeline clock is used by the output transport to know when a frame needs to be presented. For that, all frames now have an optional pts field (prensentation timestamp). There's currently just one clock implementation SystemClock and the pts field is currently only used for TextFrames (audio and image frames will be next).

  • A clock can now be specified to PipelineTask (defaults to SystemClock). This clock will be passed to each frame processor via the StartFrame.

  • Added CartesiaHttpTTSService.

  • DailyTransport now supports setting the audio bitrate to improve audio quality through the DailyParams.audio_out_bitrate parameter. The new default is 96kbps.

  • DailyTransport now uses the number of audio output channels (1 or 2) to set mono or stereo audio when needed.

  • Interruptions support has been added to TwilioFrameSerializer when using FastAPIWebsocketTransport.

  • Added new LmntTTSService text-to-speech service. (see https://www.lmnt.com/)

  • Added TTSModelUpdateFrame, TTSLanguageUpdateFrame, STTModelUpdateFrame, and STTLanguageUpdateFrame frames to allow you to switch models, language and voices in TTS and STT services.

  • Added new transcriptions.Language enum.

Changed

  • Context frames are now pushed downstream from assistant context aggregators.

  • Removed Silero VAD torch dependency.

  • Updated individual update settings frame classes into a single ServiceUpdateSettingsFrame class.

  • We now distinguish between input and output audio and image frames. We introduce InputAudioRawFrame, OutputAudioRawFrame, InputImageRawFrame and OutputImageRawFrame (and other subclasses of those). The input frames usually come from an input transport and are meant to be processed inside the pipeline to generate new frames. However, the input frames will not be sent through an output transport. The output frames can also be processed by any frame processor in the pipeline and they are allowed to be sent by the output transport.

  • ParallelTask has been renamed to SyncParallelPipeline. A SyncParallelPipeline is a frame processor that contains a list of different pipelines to be executed concurrently. The difference between a SyncParallelPipeline and a ParallelPipeline is that, given an input frame, the SyncParallelPipeline will wait for all the internal pipelines to complete. This is achieved by making sure the last processor in each of the pipelines is synchronous (e.g. an HTTP-based service that waits for the response).

  • StartFrame is back a system frame to make sure it's processed immediately by all processors. EndFrame stays a control frame since it needs to be ordered allowing the frames in the pipeline to be processed.

  • Updated MoondreamService revision to 2024-08-26.

  • CartesiaTTSService and ElevenLabsTTSService now add presentation timestamps to their text output. This allows the output transport to push the text frames downstream at almost the same time the words are spoken. We say "almost" because currently the audio frames don't have presentation timestamp but they should be played at roughly the same time.

  • DailyTransport.on_joined event now returns the full session data instead of just the participant.

  • CartesiaTTSService is now a subclass of TTSService.

  • DeepgramSTTService is now a subclass of STTService.

  • WhisperSTTService is now a subclass of SegmentedSTTService. A SegmentedSTTService is a STTService where the provided audio is given in a big chunk (i.e. from when the user starts speaking until the user stops speaking) instead of a continous stream.

Fixed

  • Fixed OpenAI multiple function calls.

  • Fixed a Cartesia TTS issue that would cause audio to be truncated in some cases.

  • Fixed a BaseOutputTransport issue that would stop audio and video rendering tasks (after receiving and EndFrame) before the internal queue was emptied, causing the pipeline to finish prematurely.

  • StartFrame should be the first frame every processor receives to avoid situations where things are not initialized (because initialization happens on StartFrame) and other frames come in resulting in undesired behavior.

Performance

  • obj_id() and obj_count() now use itertools.count avoiding the need of threading.Lock.

Other

[0.0.41] - 2024-08-22

Added

  • Added LivekitFrameSerializer audio frame serializer.

Fixed

  • Fix FastAPIWebsocketOutputTransport variable name clash with subclass.

  • Fix an AnthropicLLMService issue with empty arguments in function calling.

Other

  • Fixed studypal example errors.

[0.0.40] - 2024-08-20

Added

  • VAD parameters can now be dynamicallt updated using the VADParamsUpdateFrame.

  • ErrorFrame has now a fatal field to indicate the bot should exit if a fatal error is pushed upstream (false by default). A new FatalErrorFrame that sets this flag to true has been added.

  • AnthropicLLMService now supports function calling and initial support for prompt caching. (see https://www.anthropic.com/news/prompt-caching)

  • ElevenLabsTTSService can now specify ElevenLabs input parameters such as output_format.

  • TwilioFrameSerializer can now specify Twilio's and Pipecat's desired sample rates to use.

  • Added new on_participant_updated event to DailyTransport.

  • Added DailyRESTHelper.delete_room_by_name() and DailyRESTHelper.delete_room_by_url().

  • Added LLM and TTS usage metrics. Those are enabled when PipelineParams.enable_usage_metrics is True.

  • AudioRawFrames are now pushed downstream from the base output transport. This allows capturing the exact words the bot says by adding an STT service at the end of the pipeline.

  • Added new GStreamerPipelineSource. This processor can generate image or audio frames from a GStreamer pipeline (e.g. reading an MP4 file, and RTP stream or anything supported by GStreamer).

  • Added TransportParams.audio_out_is_live. This flag is False by default and it is useful to indicate we should not synchronize audio with sporadic images.

  • Added new BotStartedSpeakingFrame and BotStoppedSpeakingFrame control frames. These frames are pushed upstream and they should wrap BotSpeakingFrame.

  • Transports now allow you to register event handlers without decorators.

Changed

  • Support RTVI message protocol 0.1. This includes new messages, support for messages responses, support for actions, configuration, webhooks and a bunch of new cool stuff. (see https://docs.rtvi.ai/)

  • SileroVAD dependency is now imported via pip's silero-vad package.

  • ElevenLabsTTSService now uses eleven_turbo_v2_5 model by default.

  • BotSpeakingFrame is now a control frame.

  • StartFrame is now a control frame similar to EndFrame.

  • DeepgramTTSService now is more customizable. You can adjust the encoding and sample rate.

Fixed

  • TTSStartFrame and TTSStopFrame are now sent when TTS really starts and stops. This allows for knowing when the bot starts and stops speaking even with asynchronous services (like Cartesia).

  • Fixed AzureSTTService transcription frame timestamps.

  • Fixed an issue with DailyRESTHelper.create_room() expirations which would cause this function to stop working after the initial expiration elapsed.

  • Improved EndFrame and CancelFrame handling. EndFrame should end things gracefully while a CancelFrame should cancel all running tasks as soon as possible.

  • Fixed an issue in AIService that would cause a yielded None value to be processed.

  • RTVI's bot-ready message is now sent when the RTVI pipeline is ready and a first participant joins.

  • Fixed a BaseInputTransport issue that was causing incoming system frames to be queued instead of being pushed immediately.

  • Fixed a BaseInputTransport issue that was causing start/stop interruptions incoming frames to not cancel tasks and be processed properly.

Other

  • Added studypal example (from to the Cartesia folks!).

  • Most examples now use Cartesia.

  • Added examples foundational/19a-tools-anthropic.py, foundational/19b-tools-video-anthropic.py and foundational/19a-tools-togetherai.py.

  • Added examples foundational/18-gstreamer-filesrc.py and foundational/18a-gstreamer-videotestsrc.py that show how to use GStreamerPipelineSource

  • Remove requests library usage.

  • Cleanup examples and use DailyRESTHelper.

[0.0.39] - 2024-07-23

Fixed

  • Fixed a regression introduced in 0.0.38 that would cause Daily transcription to stop the Pipeline.

[0.0.38] - 2024-07-23

Added

  • Added force_reload, skip_validation and trust_repo to SileroVAD and SileroVADAnalyzer. This allows caching and various GitHub repo validations.

  • Added send_initial_empty_metrics flag to PipelineParams to request for initial empty metrics (zero values). True by default.

Fixed

  • Fixed initial metrics format. It was using the wrong keys name/time instead of processor/value.

  • STT services should be using ISO 8601 time format for transcription frames.

  • Fixed an issue that would cause Daily transport to show a stop transcription error when actually none occurred.

[0.0.37] - 2024-07-22

Added

  • Added RTVIProcessor which implements the RTVI-AI standard. See https://github.com/rtvi-ai

  • Added BotInterruptionFrame which allows interrupting the bot while talking.

  • Added LLMMessagesAppendFrame which allows appending messages to the current LLM context.

  • Added LLMMessagesUpdateFrame which allows changing the LLM context for the one provided in this new frame.

  • Added LLMModelUpdateFrame which allows updating the LLM model.

  • Added TTSSpeakFrame which causes the bot say some text. This text will not be part of the LLM context.

  • Added TTSVoiceUpdateFrame which allows updating the TTS voice.

Removed

  • We remove the LLMResponseStartFrame and LLMResponseEndFrame frames. These were added in the past to properly handle interruptions for the LLMAssistantContextAggregator. But the LLMContextAggregator is now based on LLMResponseAggregator which handles interruptions properly by just processing the StartInterruptionFrame, so there's no need for these extra frames any more.

Fixed

  • Fixed an issue with StatelessTextTransformer where it was pushing a string instead of a TextFrame.

  • TTSService end of sentence detection has been improved. It now works with acronyms, numbers, hours and others.

  • Fixed an issue in TTSService that would not properly flush the current aggregated sentence if an LLMFullResponseEndFrame was found.

Performance

  • CartesiaTTSService now uses websockets which improves speed. It also leverages the new Cartesia contexts which maintains generated audio prosody when multiple inputs are sent, therefore improving audio quality a lot.

[0.0.36] - 2024-07-02

Added

  • Added GladiaSTTService. See https://docs.gladia.io/chapters/speech-to-text-api/pages/live-speech-recognition

  • Added XTTSService. This is a local Text-To-Speech service. See https://github.com/coqui-ai/TTS

  • Added UserIdleProcessor. This processor can be used to wait for any interaction with the user. If the user doesn't say anything within a given timeout a provided callback is called.

  • Added IdleFrameProcessor. This processor can be used to wait for frames within a given timeout. If no frame is received within the timeout a provided callback is called.

  • Added new frame BotSpeakingFrame. This frame will be continuously pushed upstream while the bot is talking.

  • It is now possible to specify a Silero VAD version when using SileroVADAnalyzer or SileroVAD.

  • Added AysncFrameProcessor and AsyncAIService. Some services like DeepgramSTTService need to process things asynchronously. For example, audio is sent to Deepgram but transcriptions are not returned immediately. In these cases we still require all frames (except system frames) to be pushed downstream from a single task. That's what AsyncFrameProcessor is for. It creates a task and all frames should be pushed from that task. So, whenever a new Deepgram transcription is ready that transcription will also be pushed from this internal task.

  • The MetricsFrame now includes processing metrics if metrics are enabled. The processing metrics indicate the time a processor needs to generate all its output. Note that not all processors generate these kind of metrics.

Changed

  • WhisperSTTService model can now also be a string.

  • Added missing * keyword separators in services.

Fixed

  • WebsocketServerTransport doesn't try to send frames anymore if serializers returns None.

  • Fixed an issue where exceptions that occurred inside frame processors were being swallowed and not displayed.

  • Fixed an issue in FastAPIWebsocketTransport where it would still try to send data to the websocket after being closed.

Other

  • Added Fly.io deployment example in examples/deployment/flyio-example.

  • Added new 17-detect-user-idle.py example that shows how to use the new UserIdleProcessor.

[0.0.35] - 2024-06-28

Changed

  • FastAPIWebsocketParams now require a serializer.

  • TwilioFrameSerializer now requires a streamSid.

Fixed

  • Silero VAD number of frames needs to be 512 for 16000 sample rate or 256 for 8000 sample rate.

[0.0.34] - 2024-06-25

Fixed

  • Fixed an issue with asynchronous STT services (Deepgram and Azure) that could interruptions to ignore transcriptions.

  • Fixed an issue introduced in 0.0.33 that would cause the LLM to generate shorter output.

[0.0.33] - 2024-06-25

Changed

  • Upgraded to Cartesia's new Python library 1.0.0. CartesiaTTSService now expects a voice ID instead of a voice name (you can get the voice ID from Cartesia's playground). You can also specify the audio sample_rate and encoding instead of the previous output_format.

Fixed

  • Fixed an issue with asynchronous STT services (Deepgram and Azure) that could cause static audio issues and interruptions to not work properly when dealing with multiple LLMs sentences.

  • Fixed an issue that could mix new LLM responses with previous ones when handling interruptions.

  • Fixed a Daily transport blocking situation that occurred while reading audio frames after a participant left the room. Needs daily-python >= 0.10.1.

[0.0.32] - 2024-06-22

Added

  • Allow specifying a DeepgramSTTService url which allows using on-prem Deepgram.

  • Added new FastAPIWebsocketTransport. This is a new websocket transport that can be integrated with FastAPI websockets.

  • Added new TwilioFrameSerializer. This is a new serializer that knows how to serialize and deserialize audio frames from Twilio.

  • Added Daily transport event: on_dialout_answered. See https://reference-python.daily.co/api_reference.html#daily.EventHandler

  • Added new AzureSTTService. This allows you to use Azure Speech-To-Text.

Performance

  • Convert BaseOutputTransport and BaseOutputTransport to fully use asyncio and remove the use of threads.

Other

  • Added twilio-chatbot. This is an example that shows how to integrate Twilio phone numbers with a Pipecat bot.

  • Updated 07f-interruptible-azure.py to use AzureLLMService, AzureSTTService and AzureTTSService.

[0.0.31] - 2024-06-13

Performance

  • Break long audio frames into 20ms chunks instead of 10ms.

[0.0.30] - 2024-06-13

Added

  • Added report_only_initial_ttfb to PipelineParams. This will make it so only the initial TTFB metrics after the user stops talking are reported.

  • Added OpenPipeLLMService. This service will let you run OpenAI through OpenPipe's SDK.

  • Allow specifying frame processors' name through a new name constructor argument.

  • Added DeepgramSTTService. This service has an ongoing websocket connection. To handle this, it subclasses AIService instead of STTService. The output of this service will be pushed from the same task, except system frames like StartFrame, CancelFrame or StartInterruptionFrame.

Changed

  • FrameSerializer.deserialize() can now return None in case it is not possible to desearialize the given data.

  • daily_rest.DailyRoomProperties now allows extra unknown parameters.

Fixed

  • Fixed an issue where DailyRoomProperties.exp always had the same old timestamp unless set by the user.

  • Fixed a couple of issues with WebsocketServerTransport. It needed to use push_audio_frame() and also VAD was not working properly.

  • Fixed an issue that would cause LLM aggregator to fail with small VADParams.stop_secs values.

  • Fixed an issue where BaseOutputTransport would send longer audio frames preventing interruptions.

Other

  • Added new 07h-interruptible-openpipe.py example. This example shows how to use OpenPipe to run OpenAI LLMs and get the logs stored in OpenPipe.

  • Added new dialin-chatbot example. This examples shows how to call the bot using a phone number.

[0.0.29] - 2024-06-07

Added

  • Added a new FunctionFilter. This filter will let you filter frames based on a given function, except system messages which should never be filtered.

  • Added FrameProcessor.can_generate_metrics() method to indicate if a processor can generate metrics. In the future this might get an extra argument to ask for a specific type of metric.

  • Added BasePipeline. All pipeline classes should be based on this class. All subclasses should implement a processors_with_metrics() method that returns a list of all FrameProcessors in the pipeline that can generate metrics.

  • Added enable_metrics to PipelineParams.

  • Added MetricsFrame. The MetricsFrame will report different metrics in the system. Right now, it can report TTFB (Time To First Byte) values for different services, that is the time spent between the arrival of a Frame to the processor/service until the first DataFrame is pushed downstream. If metrics are enabled an intial MetricsFrame with all the services in the pipeline will be sent.

  • Added TTFB metrics and debug logging for TTS services.

Changed

  • Moved ParallelTask to pipecat.pipeline.parallel_task.

Fixed

  • Fixed PlayHT TTS service to work properly async.

[0.0.28] - 2024-06-05

Fixed

  • Fixed an issue with SileroVADAnalyzer that would cause memory to keep growing indefinitely.

[0.0.27] - 2024-06-05

Added

  • Added DailyTransport.participants() and DailyTransport.participant_counts().

[0.0.26] - 2024-06-05

Added

  • Added OpenAITTSService.

  • Allow passing output_format and model_id to CartesiaTTSService to change audio sample format and the model to use.

  • Added DailyRESTHelper which helps you create Daily rooms and tokens in an easy way.

  • PipelineTask now has a has_finished() method to indicate if the task has completed. If a task is never ran has_finished() will return False.

  • PipelineRunner now supports SIGTERM. If received, the runner will be canceled.

Fixed

  • Fixed an issue where BaseInputTransport and BaseOutputTransport where stopping push tasks before pushing EndFrame frames could cause the bots to get stuck.

  • Fixed an error closing local audio transports.

  • Fixed an issue with Deepgram TTS that was introduced in the previous release.

  • Fixed AnthropicLLMService interruptions. If an interruption occurred, a user message could be appended after the previous user message. Anthropic does not allow that because it requires alternate user and assistant messages.

Performance

  • The BaseInputTransport does not pull audio frames from sub-classes any more. Instead, sub-classes now push audio frames into a queue in the base class. Also, DailyInputTransport now pushes audio frames every 20ms instead of 10ms.

  • Remove redundant camera input thread from DailyInputTransport. This should improve performance a little bit when processing participant videos.

  • Load Cartesia voice on startup.

[0.0.25] - 2024-05-31

Added

  • Added WebsocketServerTransport. This will create a websocket server and will read messages coming from a client. The messages are serialized/deserialized with protobufs. See examples/websocket-server for a detailed example.

  • Added function calling (LLMService.register_function()). This will allow the LLM to call functions you have registered when needed. For example, if you register a function to get the weather in Los Angeles and ask the LLM about the weather in Los Angeles, the LLM will call your function. See https://platform.openai.com/docs/guides/function-calling

  • Added new LangchainProcessor.

  • Added Cartesia TTS support (https://cartesia.ai/)

Fixed

  • Fixed SileroVAD frame processor.

  • Fixed an issue where camera_out_enabled would cause the highg CPU usage if no image was provided.

Performance

  • Removed unnecessary audio input tasks.

[0.0.24] - 2024-05-29

Added

  • Exposed on_dialin_ready for Daily transport SIP endpoint handling. This notifies when the Daily room SIP endpoints are ready. This allows integrating with third-party services like Twilio.

  • Exposed Daily transport on_app_message event.

  • Added Daily transport on_call_state_updated event.

  • Added Daily transport start_recording(), stop_recording and stop_dialout.

Changed

  • Added PipelineParams. This replaces the allow_interruptions argument in PipelineTask and will allow future parameters in the future.

  • Fixed Deepgram Aura TTS base_url and added ErrorFrame reporting.

  • GoogleLLMService api_key argument is now mandatory.

Fixed

  • Daily tranport dialin-ready doesn't not block anymore and it now handles timeouts.

  • Fixed AzureLLMService.

[0.0.23] - 2024-05-23

Fixed

  • Fixed an issue handling Daily transport dialin-ready event.

[0.0.22] - 2024-05-23

Added

[0.0.21] - 2024-05-22

Added

  • Added vision support to Anthropic service.

  • Added WakeCheckFilter which allows you to pass information downstream only if you say a certain phrase/word.

Changed

  • Filter has been renamed to FrameFilter and it's now under processors/filters.

Fixed

  • Fixed Anthropic service to use new frame types.

  • Fixed an issue in LLMUserResponseAggregator and UserResponseAggregator that would cause frames after a brief pause to not be pushed to the LLM.

  • Clear the audio output buffer if we are interrupted.

  • Re-add exponential smoothing after volume calculation. This makes sure the volume value being used doesn't fluctuate so much.

[0.0.20] - 2024-05-22

Added

  • In order to improve interruptions we now compute a loudness level using pyloudnorm. The audio coming WebRTC transports (e.g. Daily) have an Automatic Gain Control (AGC) algorithm applied to the signal, however we don't do that on our local PyAudio signals. This means that currently incoming audio from PyAudio is kind of broken. We will fix it in future releases.

Fixed

  • Fixed an issue where StartInterruptionFrame would cause LLMUserResponseAggregator to push the accumulated text causing the LLM respond in the wrong task. The StartInterruptionFrame should not trigger any new LLM response because that would be spoken in a different task.

  • Fixed an issue where tasks and threads could be paused because the executor didn't have more tasks available. This was causing issues when cancelling and recreating tasks during interruptions.

[0.0.19] - 2024-05-20

Changed

  • LLMUserResponseAggregator and LLMAssistantResponseAggregator internal messages are now exposed through the messages property.

Fixed

  • Fixed an issue where LLMAssistantResponseAggregator was not accumulating the full response but short sentences instead. If there's an interruption we only accumulate what the bot has spoken until now in a long response as well.

[0.0.18] - 2024-05-20

Fixed

  • Fixed an issue in DailyOuputTransport where transport messages were not being sent.

[0.0.17] - 2024-05-19

Added

  • Added google.generativeai model support, including vision. This new google service defaults to using gemini-1.5-flash-latest. Example in examples/foundational/12a-describe-video-gemini-flash.py.

  • Added vision support to openai service. Example in examples/foundational/12a-describe-video-gemini-flash.py.

  • Added initial interruptions support. The assistant contexts (or aggregators) should now be placed after the output transport. This way, only the completed spoken context is added to the assistant context.

  • Added VADParams so you can control voice confidence level and others.

  • VADAnalyzer now uses an exponential smoothed volume to improve speech detection. This is useful when voice confidence is high (because there's someone talking near you) but volume is low.

Fixed

  • Fixed an issue where TTSService was not pushing TextFrames downstream.

  • Fixed issues with Ctrl-C program termination.

  • Fixed an issue that was causing StopTaskFrame to actually not exit the PipelineTask.

[0.0.16] - 2024-05-16

Fixed

  • DailyTransport: don't publish camera and audio tracks if not enabled.

  • Fixed an issue in BaseInputTransport that was causing frames pushed downstream not pushed in the right order.

[0.0.15] - 2024-05-15

Fixed

  • Quick hot fix for receiving DailyTransportMessage.

[0.0.14] - 2024-05-15

Added

  • Added DailyTransport event on_participant_left.

  • Added support for receiving DailyTransportMessage.

Fixed

  • Images are now resized to the size of the output camera. This was causing images not being displayed.

  • Fixed an issue in DailyTransport that would not allow the input processor to shutdown if no participant ever joined the room.

  • Fixed base transports start and stop. In some situation processors would halt or not shutdown properly.

[0.0.13] - 2024-05-14

Changed

  • MoondreamService argument model_id is now model.

  • VADAnalyzer arguments have been renamed for more clarity.

Fixed

  • Fixed an issue with DailyInputTransport and DailyOutputTransport that could cause some threads to not start properly.

  • Fixed STTService. Add max_silence_secs and max_buffer_secs to handle better what's being passed to the STT service. Also add exponential smoothing to the RMS.

  • Fixed WhisperSTTService. Add no_speech_prob to avoid garbage output text.

[0.0.12] - 2024-05-14

Added

  • Added DailyTranscriptionSettings to be able to specify transcription settings much easier (e.g. language).

Other

  • Updated simple-chatbot with Spanish.

  • Add missing dependencies in some of the examples.

[0.0.11] - 2024-05-13

Added

  • Allow stopping pipeline tasks with new StopTaskFrame.

Changed

  • TTS, STT and image generation service now use AsyncGenerator.

Fixed

  • DailyTransport: allow registering for participant transcriptions even if input transport is not initialized yet.

Other

  • Updated storytelling-chatbot.

[0.0.10] - 2024-05-13

Added

  • Added Intel GPU support to MoondreamService.

  • Added support for sending transport messages (e.g. to communicate with an app at the other end of the transport).

  • Added FrameProcessor.push_error() to easily send an ErrorFrame upstream.

Fixed

  • Fixed Azure services (TTS and image generation).

Other

  • Updated simple-chatbot, moondream-chatbot and translation-chatbot examples.

[0.0.9] - 2024-05-12

Changed

Many things have changed in this version. Many of the main ideas such as frames, processors, services and transports are still there but some things have changed a bit.

  • Frames describe the basic units for processing. For example, text, image or audio frames. Or control frames to indicate a user has started or stopped speaking.

  • FrameProcessors process frames (e.g. they convert a TextFrame to an ImageRawFrame) and push new frames downstream or upstream to their linked peers.

  • FrameProcessors can be linked together. The easiest wait is to use the Pipeline which is a container for processors. Linking processors allow frames to travel upstream or downstream easily.

  • Transports are a way to send or receive frames. There can be local transports (e.g. local audio or native apps), network transports (e.g. websocket) or service transports (e.g. https://daily.co).

  • Pipelines are just a processor container for other processors.

  • A PipelineTask know how to run a pipeline.

  • A PipelineRunner can run one or more tasks and it is also used, for example, to capture Ctrl-C from the user.

[0.0.8] - 2024-04-11

Added

  • Added FireworksLLMService.

  • Added InterimTranscriptionFrame and enable interim results in DailyTransport transcriptions.

Changed

  • FalImageGenService now uses new fal_client package.

Fixed

  • FalImageGenService: use asyncio.to_thread to not block main loop when generating images.

  • Allow TranscriptionFrame after an end frame (transcriptions can be delayed and received after UserStoppedSpeakingFrame).

[0.0.7] - 2024-04-10

Added

  • Add use_cpu argument to MoondreamService.

[0.0.6] - 2024-04-10

Added

  • Added FalImageGenService.InputParams.

  • Added URLImageFrame and UserImageFrame.

  • Added UserImageRequestFrame and allow requesting an image from a participant.

  • Added base VisionService and MoondreamService

Changed

  • Don't pass image_size to ImageGenService, images should have their own size.

  • ImageFrame now receives a tuple(width,height) to specify the size.

  • on_first_other_participant_joined now gets a participant argument.

Fixed

  • Check if camera, speaker and microphone are enabled before writing to them.

Performance

  • DailyTransport only subscribe to desired participant video track.

[0.0.5] - 2024-04-06

Changed

  • Use camera_bitrate and camera_framerate.

  • Increase camera_framerate to 30 by default.

Fixed

  • Fixed LocalTransport.read_audio_frames.

[0.0.4] - 2024-04-04

Added

  • Added project optional dependencies [silero,openai,...].

Changed

  • Moved thransports to its own directory.

  • Use OPENAI_API_KEY instead of OPENAI_CHATGPT_API_KEY.

Fixed

  • Don't write to microphone/speaker if not enabled.

Other

  • Added live translation example.

  • Fix foundational examples.

[0.0.3] - 2024-03-13

Other

  • Added storybot and chatbot examples.

[0.0.2] - 2024-03-12

Initial public release.