# Changelog 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). ## [Unreleased] ### Added - Added a new RTVI message called `disconnect-bot`, which when handled pushes an `EndFrame` to trigger the pipeline to stop. ## [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 `TextFrame`s (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 - Pipecat now uses Ruff as its formatter (https://github.com/astral-sh/ruff). ## [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. - `AudioRawFrame`s 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 `FrameProcessor`s 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 - Added Daily transport `start_dialout()` to be able to make phone or SIP calls. See https://reference-python.daily.co/api_reference.html#daily.CallClient.start_dialout - Added Daily transport support for dial-in use cases. - Added Daily transport events: `on_dialout_connected`, `on_dialout_stopped`, `on_dialout_error` and `on_dialout_warning`. See https://reference-python.daily.co/api_reference.html#daily.EventHandler ## [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](https://github.com/csteinmetz1/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. - `Frame`s 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. - `FrameProcessor`s process frames (e.g. they convert a `TextFrame` to an `ImageRawFrame`) and push new frames downstream or upstream to their linked peers. - `FrameProcessor`s 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. - `Transport`s 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). - `Pipeline`s 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.