v0.0.52

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.

v0.0.51

Fixed

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

v0.0.50

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.

v0.0.49

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.

v0.0.48

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.

v0.0.47

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.

v0.0.46

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.

v0.0.45

Changed

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

v0.0.44

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.

v0.0.43

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.