mirror of
https://github.com/mudler/LocalAI.git
synced 2026-06-12 18:58:49 -04:00
* feat(realtime): pipeline streaming + disable_thinking config
Add a nested pipeline.streaming.{llm,tts,transcription} block plus
pipeline.disable_thinking, with StreamLLM/StreamTTS/StreamTranscription/
ThinkingDisabled helpers. Pointer-bools so unset keeps the unary path;
existing configs are unaffected. Wiring into the realtime handler follows.
Assisted-by: Claude:claude-opus-4-8 go vet
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(realtime): sentence segmenter for streamed LLM->TTS pipelining
streamSegmenter accumulates streamed LLM tokens and emits complete
sentence/clause segments (terminator+whitespace, or newline) so TTS can
synthesize each segment as it completes instead of waiting for the whole
reply. Pure helper; the streaming handler wiring consumes it next.
Assisted-by: Claude:claude-opus-4-8 go vet
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(realtime): streaming TTS/transcription methods on Model interface
Add TTSStream and TranscribeStream to the realtime Model interface and
implement them on wrappedModel (delegating to backend.ModelTTSStream /
ModelTranscriptionStream) and transcriptOnlyModel. ttsStream adapts the
backend's WAV-framed stream (44-byte header carrying the sample rate, then
PCM) into raw PCM + sample rate for the realtime transports. Handler wiring
that consumes these (flag-gated) follows.
Assisted-by: Claude:claude-opus-4-8 go vet
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(realtime): emitSpeech with flag-gated streaming TTS
emitSpeech synthesizes a piece of text and forwards audio to the client,
streaming one output_audio.delta per backend PCM chunk when the pipeline
sets streaming.tts, or one delta for the whole utterance otherwise. WebRTC
gets raw PCM (it resamples internally); WebSocket gets base64 PCM at the
session rate. It emits no transcript/audio-done events so a streamed reply
can be split into multiple spoken segments sharing one response.
Adds fakeModel/fakeTransport test doubles for the realtime Model/Transport
interfaces, driving streaming assertions deterministically.
Assisted-by: Claude:claude-opus-4-8 go vet
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(realtime): route response audio through emitSpeech (streaming TTS)
Replace the inline unary TTS block in the response handler with emitSpeech,
which streams a response.output_audio.delta per backend PCM chunk when
pipeline.streaming.tts is set and otherwise preserves the single-delta unary
behaviour. emitSpeech returns the accumulated base64 audio, stored on the
conversation item as before. Transcript and audio-done events stay in the
handler so later per-segment streaming can reuse emitSpeech.
Assisted-by: Claude:claude-opus-4-8 go vet
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(realtime): streaming transcription text deltas
Add emitTranscription and route commitUtterance through it. With
pipeline.streaming.transcription set it streams each transcript fragment as
a conversation.item.input_audio_transcription.delta via TranscribeStream
then a completed event; otherwise it preserves the single completed-event
unary behaviour. Returns the final transcript for response generation.
Assisted-by: Claude:claude-opus-4-8 go vet
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(realtime): pipeline disable_thinking maps to enable_thinking off
applyPipelineThinking forces the LLM's ReasoningConfig.DisableReasoning when
pipeline.disable_thinking is set, which gRPCPredictOpts turns into the
enable_thinking=false backend metadata. Applied at newModel construction on
the per-session LLM config copy, so it doesn't leak to other model users and
needs no realtime-specific request plumbing.
Assisted-by: Claude:claude-opus-4-8 go vet
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(realtime): speechStreamer for token-streamed LLM->TTS
emitSpeech now returns raw PCM (caller base64-encodes) so streamed segments
accumulate correctly. speechStreamer consumes streamed LLM tokens: it strips
reasoning via the streaming ReasoningExtractor, emits a transcript delta per
content fragment, and sentence-pipes content into emitSpeech so each sentence
is synthesized as soon as it's ready. Handler wiring (plain-content turns)
follows.
Assisted-by: Claude:claude-opus-4-8 go vet
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(realtime): wire streamLLMResponse for token-streamed replies
triggerResponseAtTurn takes a streamed path when pipeline.streaming.llm is
set, the turn has no tools, and audio is requested: streamLLMResponse
announces the assistant item, drives the LLM token callback through a
speechStreamer (reasoning-stripped transcript deltas + sentence-piped TTS),
and emits the terminal events. Tool turns and non-streaming pipelines keep
the existing buffered path unchanged, so this is strictly opt-in.
Assisted-by: Claude:claude-opus-4-8 go vet
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* docs(realtime): document pipeline streaming + disable_thinking
Assisted-by: Claude:claude-opus-4-8
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* fix(realtime): register pipeline streaming/thinking config fields
TestAllFieldsHaveRegistryEntries (core/config/meta) requires every config
field to have a meta registry entry. The four new pipeline fields
(disable_thinking, streaming.{llm,tts,transcription}) had none, failing
tests-linux/tests-apple. Add toggle entries for them.
Also handle the os.Remove return in realtime_speech_test.go to satisfy
errcheck (golangci-lint).
Assisted-by: Claude:claude-opus-4-8 go test, golangci-lint
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* fix(realtime): always strip reasoning from spoken output
disable_thinking maps to ReasoningConfig.DisableReasoning=true on the LLM
config, which the backend reads as enable_thinking=false. But the realtime
handler reads that SAME config to drive reasoning extraction, and there
DisableReasoning=true means "skip stripping". PredictConfig() returns this
LLM config, so both the streamed (speechStreamer) and buffered realtime
paths stopped stripping <think>…</think> exactly when disable_thinking was
on — leaking raw reasoning to the client whenever the model ignored the
enable_thinking hint (e.g. lfm2.5).
Add spokenReasoningConfig() which clears DisableReasoning for extraction
(keeping custom tokens/tag pairs) and route both realtime paths through it.
Spoken output now always strips reasoning, independent of the backend
suppression hint.
Assisted-by: Claude:claude-opus-4-8 go test, golangci-lint
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* fix(realtime): clean TTS temp path before read (gosec G304)
emitSpeech reads the WAV file the TTS backend wrote. The read moved here
from realtime.go, so code-scanning flagged it as a new G304 alert even
though the path is backend-controlled (a temp file), not user input.
Wrap it in filepath.Clean — a real path normalization that also clears
the alert, keeping with the repo's no-#nosec convention.
Assisted-by: Claude:claude-opus-4-8 gosec, golangci-lint
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactor(realtime): buffer whole message for TTS, drop sentence segmenter
Per review (richiejp): the sentence segmenter pipelined unary TTS by
splitting on ASCII .!?/newline, which does nothing for languages without
those boundaries (CJK/Thai) — there it already degraded to buffering the
whole message anyway.
Replace it with a uniform model: stream the LLM transcript live, buffer the
full message, then synthesize it once. emitSpeech already streams the audio
chunks when the backend implements TTSStream and falls back to a single
unary delta otherwise, so this is real streaming TTS where supported and a
clean whole-message synthesis elsewhere — no per-sentence emulation, no
language assumptions. speechStreamer becomes transcriptStreamer (transcript
deltas only); the whole-message synthesis moves into streamLLMResponse.
Assisted-by: Claude:claude-opus-4-8 go test, golangci-lint
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(realtime): stream tool-call turns via tokenizer-template autoparser
Per review (richiejp): tool-call deltas exist, so streaming should work with
tools too. It does — for models that use their tokenizer template. The C++
autoparser then clears reply.Message and delivers content + tool calls via
ChatDeltas, so the streamed transcript carries only spoken content (no
tool-call JSON leak) and the tool calls are parsed from the final response.
- Drop the len(tools)==0 gate; stream when no tools OR use_tokenizer_template
(grammar-based function calling still buffers, since its call is emitted as
JSON in the token stream and would leak into the transcript).
- streamLLMResponse takes tools/toolChoice/toolTurn, reads ChatDelta content
in the token callback, parses tool calls from the final ChatDeltas, and
creates the assistant content item lazily so a content-less tool turn emits
only the tool calls.
- Extract emitToolCallItems from the buffered path so both paths finalize tool
calls, response.done, and server-side assistant-tool follow-ups identically.
Assisted-by: Claude:claude-opus-4-8 go test, golangci-lint
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(realtime): script-aware clause chunking + streamed-reply fixes
Opt-in pipeline.streaming.clause_chunking splits the streamed LLM reply
into speakable clauses and synthesizes each as soon as it completes,
lowering time-to-first-audio instead of buffering the whole message. The
splitter is script-aware (rivo/uniseg, pure Go): UAX#29 sentence
segmentation handles CJK 。!? with no whitespace, CJK clause
punctuation (,、;:) and Thai/Lao spaces give finer cuts, and a UAX#14
line-break cap bounds an over-long punctuation-less run. Unlike the old
ASCII .!?/newline segmenter (dropped in 076dcdbe) it does not degrade to
whole-message buffering for CJK/Thai; scripts needing a dictionary
(Khmer/Burmese) stay buffered until a space or end-of-message. Clauses
are synthesized synchronously in the token callback (the LLM keeps
generating into the gRPC stream meanwhile), so audio still starts
mid-generation. Off by default — the whole-message path is unchanged.
Also fix the streamed-reply path and the Talk page:
- Don't swallow streamed autoparser content as reasoning: the
tokenizer-template path already delivers reasoning-free content via
ChatDeltas, so prefilling the thinking start token re-tagged it as an
unclosed reasoning block, leaving no spoken reply. Disable the prefill
on that path; closed tag pairs are still stripped (#9985).
- Generate collision-free realtime IDs (16 random bytes) instead of a
constant, so per-item bookkeeping (cancel, conversation.item.retrieve)
works.
- Key the Talk transcript by the server item_id and upsert entries.
Realtime events arrive over a WebRTC data channel — outside React's
event system — so React defers the setTranscript updaters while
synchronous ref writes in handler bodies run first; the old
index-tracking ref rendered a duplicate assistant bubble on
completion. Upserts by item_id are idempotent and order-independent.
- Drop the partial assistant bubble on a cancelled response (barge-in):
the server discards the interrupted item and sends response.done with
status "cancelled"; mirror that in the UI so the regenerated reply
isn't rendered as a second assistant message.
Assisted-by: Claude:claude-opus-4-8 [Claude Code]
Assisted-by: Claude:claude-fable-5 [Claude Code]
Signed-off-by: Richard Palethorpe <io@richiejp.com>
---------
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
Signed-off-by: Richard Palethorpe <io@richiejp.com>
Co-authored-by: Ettore Di Giacinto <mudler@localai.io>
Co-authored-by: Richard Palethorpe <io@richiejp.com>
1996 lines
65 KiB
Go
1996 lines
65 KiB
Go
package openai
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import (
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"context"
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"crypto/rand"
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"encoding/base64"
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"encoding/binary"
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"encoding/hex"
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"encoding/json"
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"fmt"
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"math"
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"os"
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"strconv"
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"sync"
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"time"
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"net/http"
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"github.com/go-audio/audio"
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"github.com/gorilla/websocket"
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"github.com/labstack/echo/v4"
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"github.com/mudler/LocalAI/core/application"
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"github.com/mudler/LocalAI/core/backend"
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"github.com/mudler/LocalAI/core/config"
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"github.com/mudler/LocalAI/core/http/auth"
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mcpTools "github.com/mudler/LocalAI/core/http/endpoints/mcp"
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"github.com/mudler/LocalAI/core/http/endpoints/openai/types"
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"github.com/mudler/LocalAI/core/schema"
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"github.com/mudler/LocalAI/core/templates"
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laudio "github.com/mudler/LocalAI/pkg/audio"
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"github.com/mudler/LocalAI/pkg/functions"
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"github.com/mudler/LocalAI/pkg/grpc"
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"github.com/mudler/LocalAI/pkg/grpc/proto"
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model "github.com/mudler/LocalAI/pkg/model"
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"github.com/mudler/LocalAI/pkg/reasoning"
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"github.com/mudler/LocalAI/pkg/sound"
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"github.com/mudler/LocalAI/pkg/utils"
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"github.com/mudler/xlog"
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)
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const (
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// XXX: Presently it seems all ASR/VAD backends use 16Khz. If a backend uses 24Khz then it will likely still work, but have reduced performance
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localSampleRate = 16000
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defaultRemoteSampleRate = 24000
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// Maximum audio buffer size in bytes (100MB) to prevent memory exhaustion
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maxAudioBufferSize = 100 * 1024 * 1024
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// Maximum WebSocket message size in bytes (10MB) to prevent DoS attacks
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maxWebSocketMessageSize = 10 * 1024 * 1024
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defaultInstructions = "You are a helpful voice assistant. " +
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"Your responses will be spoken aloud using text-to-speech, so keep them concise and conversational. " +
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"Do not use markdown formatting, bullet points, numbered lists, code blocks, or special characters. " +
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"Speak naturally as you would in a phone conversation. " +
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"Avoid parenthetical asides, URLs, and anything that cannot be clearly vocalized."
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)
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// resolveOutputModalities returns the effective output modalities for a
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// response: response-level overrides session-level, and the OpenAI Realtime
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// spec default is ["audio"] when neither is set.
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func resolveOutputModalities(session, response []types.Modality) []types.Modality {
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if len(response) > 0 {
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return response
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}
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if len(session) > 0 {
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return session
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}
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return []types.Modality{types.ModalityAudio}
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}
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// modalitiesContainAudio reports whether the resolved modalities include audio
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// output.
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func modalitiesContainAudio(m []types.Modality) bool {
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for _, x := range m {
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if x == types.ModalityAudio {
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return true
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}
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}
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return false
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}
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// A model can be "emulated" that is: transcribe audio to text -> feed text to the LLM -> generate audio as result
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// If the model support instead audio-to-audio, we will use the specific gRPC calls instead
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// Session represents a single WebSocket connection and its state
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type Session struct {
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ID string
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TranscriptionOnly bool
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// The pipeline or any-to-any model name (full realtime mode)
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Model string
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// The voice may be a TTS model name or a parameter passed to a TTS model
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Voice string
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TurnDetection *types.TurnDetectionUnion // "server_vad", "semantic_vad" or "none"
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InputAudioTranscription *types.AudioTranscription
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Tools []types.ToolUnion
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ToolChoice *types.ToolChoiceUnion
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Conversations map[string]*Conversation
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InputAudioBuffer []byte
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AudioBufferLock sync.Mutex
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OpusFrames [][]byte
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OpusFramesLock sync.Mutex
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Instructions string
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DefaultConversationID string
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ModelInterface Model
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// The pipeline model config or the config for an any-to-any model
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ModelConfig *config.ModelConfig
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InputSampleRate int
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OutputSampleRate int
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MaxOutputTokens types.IntOrInf
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// OutputModalities mirrors the OpenAI Realtime spec field of the same
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// name. Empty means "use the spec default" (audio). ["text"] suppresses
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// TTS so the client receives only response.output_text.* events.
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OutputModalities []types.Modality
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// MaxHistoryItems caps the number of MessageItems passed to the LLM each
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// turn (0 = unlimited). Small models — especially the LFM2.5-Audio 1.5B
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// served via the liquid-audio backend — degrade quickly past a handful
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// of turns. Counted from the tail; FunctionCall + FunctionCallOutput
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// pairs are kept together so we never feed an orphaned tool result.
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MaxHistoryItems int
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// AssistantExecutor is non-nil when the session opted into the in-process
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// LocalAI Assistant tool surface. Tool calls whose name matches this
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// executor's catalog are run inproc and their output is fed back to the
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// model server-side; the client never sees a function_call_arguments
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// event for those. Mirrors the chat handler's metadata.localai_assistant
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// path.
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AssistantExecutor mcpTools.ToolExecutor
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// AssistantTools is the cached ToolUnion slice we injected at session
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// creation. Re-applied after every client session.update so a
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// client-driven tool refresh (e.g. toggling a client MCP server) doesn't
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// silently strip Manage Mode's tools.
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AssistantTools []types.ToolUnion
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// Response cancellation: protects activeResponseCancel/activeResponseDone
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responseMu sync.Mutex
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activeResponseCancel context.CancelFunc
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activeResponseDone chan struct{}
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}
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// cancelActiveResponse cancels any in-flight response and waits for its
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// goroutine to exit. This ensures we never have overlapping responses and
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// that interrupted responses are fully cleaned up before starting a new one.
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func (s *Session) cancelActiveResponse() {
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s.responseMu.Lock()
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cancel := s.activeResponseCancel
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done := s.activeResponseDone
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s.responseMu.Unlock()
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if cancel != nil {
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cancel()
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}
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if done != nil {
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<-done
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}
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}
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// startResponse cancels any active response and returns a new context for
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// the replacement response. The caller MUST close the returned done channel
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// when the response goroutine exits.
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func (s *Session) startResponse(parent context.Context) (context.Context, chan struct{}) {
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s.cancelActiveResponse()
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ctx, cancel := context.WithCancel(parent)
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done := make(chan struct{})
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s.responseMu.Lock()
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s.activeResponseCancel = cancel
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s.activeResponseDone = done
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s.responseMu.Unlock()
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return ctx, done
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}
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func (s *Session) FromClient(session *types.SessionUnion) {
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}
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func (s *Session) ToServer() types.SessionUnion {
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if s.TranscriptionOnly {
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return types.SessionUnion{
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Transcription: &types.TranscriptionSession{
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ID: s.ID,
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Object: "realtime.transcription_session",
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Audio: &types.TranscriptionSessionAudio{
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Input: &types.SessionAudioInput{
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Transcription: s.InputAudioTranscription,
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},
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},
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},
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}
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} else {
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return types.SessionUnion{
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Realtime: &types.RealtimeSession{
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ID: s.ID,
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Object: "realtime.session",
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Model: s.Model,
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Instructions: s.Instructions,
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Tools: s.Tools,
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ToolChoice: s.ToolChoice,
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MaxOutputTokens: s.MaxOutputTokens,
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OutputModalities: s.OutputModalities,
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Audio: &types.RealtimeSessionAudio{
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Input: &types.SessionAudioInput{
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TurnDetection: s.TurnDetection,
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Transcription: s.InputAudioTranscription,
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},
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Output: &types.SessionAudioOutput{
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Voice: types.Voice(s.Voice),
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},
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},
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},
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}
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}
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}
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// Conversation represents a conversation with a list of items
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type Conversation struct {
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ID string
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Items []*types.MessageItemUnion
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Lock sync.Mutex
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}
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func (c *Conversation) ToServer() types.Conversation {
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return types.Conversation{
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ID: c.ID,
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Object: "realtime.conversation",
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}
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}
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// Map to store sessions (in-memory)
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var sessions = make(map[string]*Session)
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var sessionLock sync.Mutex
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type Model interface {
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VAD(ctx context.Context, request *schema.VADRequest) (*schema.VADResponse, error)
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Transcribe(ctx context.Context, audio, language string, translate bool, diarize bool, prompt string) (*schema.TranscriptionResult, error)
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Predict(ctx context.Context, messages schema.Messages, images, videos, audios []string, tokenCallback func(string, backend.TokenUsage) bool, tools []types.ToolUnion, toolChoice *types.ToolChoiceUnion, logprobs *int, topLogprobs *int, logitBias map[string]float64) (func() (backend.LLMResponse, error), error)
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TTS(ctx context.Context, text, voice, language string) (string, *proto.Result, error)
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// TTSStream synthesizes speech incrementally, invoking onAudio with raw PCM
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// chunks (and the backend sample rate) as they are produced.
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TTSStream(ctx context.Context, text, voice, language string, onAudio func(pcm []byte, sampleRate int) error) error
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// TranscribeStream transcribes audio incrementally, invoking onDelta for each
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// transcript text fragment and returning the final aggregated result.
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TranscribeStream(ctx context.Context, audio, language string, translate, diarize bool, prompt string, onDelta func(text string)) (*schema.TranscriptionResult, error)
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PredictConfig() *config.ModelConfig
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}
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var upgrader = websocket.Upgrader{
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CheckOrigin: func(r *http.Request) bool {
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return true // Allow all origins
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},
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}
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// TODO: Implement ephemeral keys to allow these endpoints to be used
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func RealtimeSessions(application *application.Application) echo.HandlerFunc {
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return func(c echo.Context) error {
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return c.NoContent(501)
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}
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}
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func RealtimeTranscriptionSession(application *application.Application) echo.HandlerFunc {
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return func(c echo.Context) error {
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return c.NoContent(501)
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}
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}
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// RealtimeSessionOptions bundles per-session knobs decoded from the WS query
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// string (or the WebRTC handshake body). Mirrors what chat.go pulls off
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// `metadata.localai_assistant` — admin-only opt-in to the in-process
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// management tool surface.
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type RealtimeSessionOptions struct {
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LocalAIAssistant bool
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// AuthEnabled mirrors chat.go's requireAssistantAccess gate. We resolve
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// admin role at handshake time (where the echo.Context has the auth
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// cookie/Bearer) and drop the result here so runRealtimeSession can
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// decide without holding onto the request.
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IsAdmin bool
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}
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func Realtime(application *application.Application) echo.HandlerFunc {
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return func(c echo.Context) error {
|
|
ws, err := upgrader.Upgrade(c.Response(), c.Request(), nil)
|
|
if err != nil {
|
|
return err
|
|
}
|
|
defer ws.Close()
|
|
|
|
// Set maximum message size to prevent DoS attacks
|
|
ws.SetReadLimit(maxWebSocketMessageSize)
|
|
|
|
// Extract query parameters from Echo context before passing to websocket handler
|
|
model := c.QueryParam("model")
|
|
assistantFlag, _ := strconv.ParseBool(c.QueryParam("localai_assistant"))
|
|
opts := RealtimeSessionOptions{
|
|
LocalAIAssistant: assistantFlag,
|
|
IsAdmin: isCurrentUserAdmin(c, application),
|
|
}
|
|
|
|
registerRealtime(application, model, opts)(ws)
|
|
return nil
|
|
}
|
|
}
|
|
|
|
// isCurrentUserAdmin replicates the chat-side admin check at the realtime
|
|
// handshake. When auth is disabled, every caller is treated as admin (same
|
|
// as chat's requireAssistantAccess).
|
|
func isCurrentUserAdmin(c echo.Context, application *application.Application) bool {
|
|
if application == nil || application.ApplicationConfig() == nil || !application.ApplicationConfig().Auth.Enabled {
|
|
return true
|
|
}
|
|
user := auth.GetUser(c)
|
|
return user != nil && user.Role == auth.RoleAdmin
|
|
}
|
|
|
|
func registerRealtime(application *application.Application, model string, opts RealtimeSessionOptions) func(c *websocket.Conn) {
|
|
return func(conn *websocket.Conn) {
|
|
t := NewWebSocketTransport(conn)
|
|
evaluator := application.TemplatesEvaluator()
|
|
xlog.Debug("Realtime WebSocket connection established", "address", conn.RemoteAddr().String(), "model", model)
|
|
runRealtimeSession(application, t, model, evaluator, opts)
|
|
}
|
|
}
|
|
|
|
// defaultMaxHistoryItems picks a sensible default cap for the session.
|
|
// Small any-to-any audio models degrade quickly past a handful of turns;
|
|
// legacy pipelines composing larger LLMs keep the historical "unlimited"
|
|
// default and rely on the LLM's own context window.
|
|
func defaultMaxHistoryItems(cfg *config.ModelConfig) int {
|
|
if cfg != nil && cfg.HasUsecases(config.FLAG_REALTIME_AUDIO) {
|
|
return 6
|
|
}
|
|
return 0
|
|
}
|
|
|
|
// trimRealtimeItems returns the tail of items capped at maxItems (0 = no cap).
|
|
// Walks backwards keeping function_call + function_call_output pairs together
|
|
// so we never feed the LLM an orphaned tool result that references a call it
|
|
// can't see.
|
|
func trimRealtimeItems(items []*types.MessageItemUnion, maxItems int) []*types.MessageItemUnion {
|
|
if maxItems <= 0 || len(items) <= maxItems {
|
|
return items
|
|
}
|
|
// Find the cut point starting from len-maxItems and pull it left until
|
|
// we're not in the middle of a tool-call pair.
|
|
cut := len(items) - maxItems
|
|
for cut > 0 && items[cut] != nil && items[cut].FunctionCallOutput != nil {
|
|
cut--
|
|
}
|
|
return items[cut:]
|
|
}
|
|
|
|
// prepareRealtimeConfig validates a model config for use in a realtime session
|
|
// and fills in pipeline slots for self-contained any-to-any models. It returns
|
|
// an error code + message pair suitable for sendError; the bool indicates
|
|
// whether the caller should proceed. Extracted from runRealtimeSession so the
|
|
// gate logic can be exercised in unit tests without a full Application.
|
|
func prepareRealtimeConfig(cfg *config.ModelConfig) (errCode, errMsg string, ok bool) {
|
|
if cfg == nil {
|
|
return "invalid_model", "Model is not a pipeline model", false
|
|
}
|
|
|
|
// Self-contained any-to-any models (e.g. liquid-audio) own the whole
|
|
// loop in one engine — surface them by populating empty pipeline slots
|
|
// with the model's own name so newModel can resolve a config for each
|
|
// role. The user can still pin individual slots (e.g. Pipeline.VAD =
|
|
// silero-vad) and those wins.
|
|
if cfg.HasUsecases(config.FLAG_REALTIME_AUDIO) {
|
|
if cfg.Pipeline.VAD == "" {
|
|
cfg.Pipeline.VAD = cfg.Name
|
|
}
|
|
if cfg.Pipeline.Transcription == "" {
|
|
cfg.Pipeline.Transcription = cfg.Name
|
|
}
|
|
if cfg.Pipeline.LLM == "" {
|
|
cfg.Pipeline.LLM = cfg.Name
|
|
}
|
|
if cfg.Pipeline.TTS == "" {
|
|
cfg.Pipeline.TTS = cfg.Name
|
|
}
|
|
return "", "", true
|
|
}
|
|
|
|
if cfg.Pipeline.VAD == "" && cfg.Pipeline.Transcription == "" && cfg.Pipeline.TTS == "" && cfg.Pipeline.LLM == "" {
|
|
return "invalid_model", "Model is not a pipeline model", false
|
|
}
|
|
return "", "", true
|
|
}
|
|
|
|
// runRealtimeSession runs the main event loop for a realtime session.
|
|
// It is transport-agnostic and works with both WebSocket and WebRTC.
|
|
func runRealtimeSession(application *application.Application, t Transport, model string, evaluator *templates.Evaluator, opts RealtimeSessionOptions) {
|
|
cl := application.ModelConfigLoader()
|
|
cfg, err := cl.LoadModelConfigFileByNameDefaultOptions(model, application.ApplicationConfig())
|
|
if err != nil {
|
|
xlog.Error("failed to load model config", "error", err)
|
|
sendError(t, "model_load_error", "Failed to load model config", "", "")
|
|
return
|
|
}
|
|
|
|
if code, msg, ok := prepareRealtimeConfig(cfg); !ok {
|
|
xlog.Error("model is not a pipeline", "model", model)
|
|
sendError(t, code, msg, "", "")
|
|
return
|
|
}
|
|
|
|
// LocalAI Assistant opt-in: gate on admin (same rule as chat.go's
|
|
// requireAssistantAccess) and grab the process-wide holder's executor.
|
|
// We collect tools + system prompt here and merge them into the session
|
|
// below so they're live from the first response.create.
|
|
var assistantTools []types.ToolUnion
|
|
var assistantSystemPrompt string
|
|
var assistantExecutor mcpTools.ToolExecutor
|
|
if opts.LocalAIAssistant {
|
|
if !opts.IsAdmin {
|
|
sendError(t, "forbidden", "localai_assistant requires admin", "", "")
|
|
return
|
|
}
|
|
appCfg := application.ApplicationConfig()
|
|
if appCfg != nil && appCfg.DisableLocalAIAssistant {
|
|
sendError(t, "unavailable", "LocalAI Assistant is disabled on this server", "", "")
|
|
return
|
|
}
|
|
holder := application.LocalAIAssistant()
|
|
if holder == nil || !holder.HasTools() {
|
|
sendError(t, "unavailable", "LocalAI Assistant is not available on this server", "", "")
|
|
return
|
|
}
|
|
exec := holder.Executor()
|
|
fns, discErr := exec.DiscoverTools(context.Background())
|
|
if discErr != nil {
|
|
xlog.Error("realtime: failed to discover LocalAI Assistant tools", "error", discErr)
|
|
sendError(t, "tool_discovery_failed", "failed to discover assistant tools: "+discErr.Error(), "", "")
|
|
return
|
|
}
|
|
assistantExecutor = exec
|
|
assistantSystemPrompt = holder.SystemPrompt()
|
|
assistantTools = make([]types.ToolUnion, 0, len(fns))
|
|
for _, fn := range fns {
|
|
fnCopy := fn
|
|
assistantTools = append(assistantTools, types.ToolUnion{
|
|
Function: &types.ToolFunction{
|
|
Name: fnCopy.Name,
|
|
Description: fnCopy.Description,
|
|
Parameters: fnCopy.Parameters,
|
|
},
|
|
})
|
|
}
|
|
xlog.Debug("realtime: LocalAI Assistant tools injected", "count", len(fns))
|
|
}
|
|
|
|
sttModel := cfg.Pipeline.Transcription
|
|
|
|
// Compose the system prompt: prepend the assistant prompt when we have
|
|
// one (it teaches the model the safety rules and tool recipes), then the
|
|
// session's default voice instructions. Order matches chat.go's
|
|
// hasSystemMessage check — assistant prompt comes first.
|
|
instructions := defaultInstructions
|
|
if assistantSystemPrompt != "" {
|
|
instructions = assistantSystemPrompt + "\n\n" + defaultInstructions
|
|
}
|
|
|
|
sessionID := generateSessionID()
|
|
session := &Session{
|
|
ID: sessionID,
|
|
TranscriptionOnly: false,
|
|
Model: model,
|
|
Voice: cfg.TTSConfig.Voice,
|
|
Instructions: instructions,
|
|
ModelConfig: cfg,
|
|
Tools: assistantTools,
|
|
AssistantTools: assistantTools,
|
|
AssistantExecutor: assistantExecutor,
|
|
TurnDetection: &types.TurnDetectionUnion{
|
|
ServerVad: &types.ServerVad{
|
|
Threshold: 0.5,
|
|
PrefixPaddingMs: 300,
|
|
SilenceDurationMs: 500,
|
|
CreateResponse: true,
|
|
},
|
|
},
|
|
InputAudioTranscription: &types.AudioTranscription{
|
|
Model: sttModel,
|
|
},
|
|
Conversations: make(map[string]*Conversation),
|
|
InputSampleRate: defaultRemoteSampleRate,
|
|
OutputSampleRate: defaultRemoteSampleRate,
|
|
MaxHistoryItems: defaultMaxHistoryItems(cfg),
|
|
}
|
|
|
|
// Create a default conversation
|
|
conversationID := generateConversationID()
|
|
conversation := &Conversation{
|
|
ID: conversationID,
|
|
// TODO: We need to truncate the conversation items when a new item is added and we have run out of space. There are multiple places where items
|
|
// can be added so we could use a datastructure here that enforces truncation upon addition
|
|
Items: []*types.MessageItemUnion{},
|
|
}
|
|
session.Conversations[conversationID] = conversation
|
|
session.DefaultConversationID = conversationID
|
|
|
|
m, err := newModel(
|
|
&cfg.Pipeline,
|
|
application.ModelConfigLoader(),
|
|
application.ModelLoader(),
|
|
application.ApplicationConfig(),
|
|
evaluator,
|
|
buildRealtimeRoutingContext(application, sessionID),
|
|
)
|
|
if err != nil {
|
|
xlog.Error("failed to load model", "error", err)
|
|
sendError(t, "model_load_error", "Failed to load model", "", "")
|
|
return
|
|
}
|
|
session.ModelInterface = m
|
|
|
|
// Store the session and notify the transport (for WebRTC audio track handling)
|
|
sessionLock.Lock()
|
|
sessions[sessionID] = session
|
|
sessionLock.Unlock()
|
|
|
|
// For WebRTC, inbound audio arrives as Opus (48kHz) and is decoded+resampled
|
|
// to localSampleRate in handleIncomingAudioTrack. Set InputSampleRate to
|
|
// match so handleVAD doesn't needlessly double-resample.
|
|
if _, ok := t.(*WebRTCTransport); ok {
|
|
session.InputSampleRate = localSampleRate
|
|
}
|
|
|
|
if sn, ok := t.(interface{ SetSession(*Session) }); ok {
|
|
sn.SetSession(session)
|
|
}
|
|
|
|
sendEvent(t, types.SessionCreatedEvent{
|
|
ServerEventBase: types.ServerEventBase{
|
|
EventID: "event_TODO",
|
|
},
|
|
Session: session.ToServer(),
|
|
})
|
|
|
|
var (
|
|
msg []byte
|
|
wg sync.WaitGroup
|
|
done = make(chan struct{})
|
|
)
|
|
|
|
vadServerStarted := false
|
|
toggleVAD := func() {
|
|
if session.TurnDetection != nil && session.TurnDetection.ServerVad != nil && !vadServerStarted {
|
|
xlog.Debug("Starting VAD goroutine...")
|
|
done = make(chan struct{})
|
|
wg.Go(func() {
|
|
conversation := session.Conversations[session.DefaultConversationID]
|
|
handleVAD(session, conversation, t, done)
|
|
})
|
|
vadServerStarted = true
|
|
} else if (session.TurnDetection == nil || session.TurnDetection.ServerVad == nil) && vadServerStarted {
|
|
xlog.Debug("Stopping VAD goroutine...")
|
|
close(done)
|
|
vadServerStarted = false
|
|
}
|
|
}
|
|
|
|
// For WebRTC sessions, start the Opus decode loop before VAD so that
|
|
// decoded PCM is already flowing when VAD's first tick fires.
|
|
var decodeDone chan struct{}
|
|
if wt, ok := t.(*WebRTCTransport); ok {
|
|
decodeDone = make(chan struct{})
|
|
go decodeOpusLoop(session, wt.opusBackend, decodeDone)
|
|
}
|
|
|
|
toggleVAD()
|
|
|
|
for {
|
|
msg, err = t.ReadEvent()
|
|
if err != nil {
|
|
xlog.Error("read error", "error", err)
|
|
break
|
|
}
|
|
|
|
// Handle diagnostic events that aren't part of the OpenAI protocol
|
|
var rawType struct {
|
|
Type string `json:"type"`
|
|
}
|
|
if json.Unmarshal(msg, &rawType) == nil && rawType.Type == "test_tone" {
|
|
if _, ok := t.(*WebSocketTransport); ok {
|
|
sendError(t, "not_supported", "test_tone is only supported on WebRTC connections", "", "")
|
|
} else {
|
|
xlog.Debug("Generating test tone")
|
|
go sendTestTone(t)
|
|
}
|
|
continue
|
|
}
|
|
|
|
// Parse the incoming message
|
|
event, err := types.UnmarshalClientEvent(msg)
|
|
if err != nil {
|
|
xlog.Error("invalid json", "error", err)
|
|
sendError(t, "invalid_json", "Invalid JSON format", "", "")
|
|
continue
|
|
}
|
|
|
|
switch e := event.(type) {
|
|
case types.SessionUpdateEvent:
|
|
xlog.Debug("recv", "message", string(msg))
|
|
|
|
// Handle transcription session update
|
|
if e.Session.Transcription != nil {
|
|
if err := updateTransSession(
|
|
session,
|
|
&e.Session,
|
|
application.ModelConfigLoader(),
|
|
application.ModelLoader(),
|
|
application.ApplicationConfig(),
|
|
); err != nil {
|
|
xlog.Error("failed to update session", "error", err)
|
|
sendError(t, "session_update_error", "Failed to update session", "", "")
|
|
continue
|
|
}
|
|
|
|
toggleVAD()
|
|
|
|
sendEvent(t, types.SessionUpdatedEvent{
|
|
ServerEventBase: types.ServerEventBase{
|
|
EventID: "event_TODO",
|
|
},
|
|
Session: session.ToServer(),
|
|
})
|
|
}
|
|
|
|
// Handle realtime session update
|
|
if e.Session.Realtime != nil {
|
|
if err := updateSession(
|
|
session,
|
|
&e.Session,
|
|
application.ModelConfigLoader(),
|
|
application.ModelLoader(),
|
|
application.ApplicationConfig(),
|
|
evaluator,
|
|
buildRealtimeRoutingContext(application, session.ID),
|
|
); err != nil {
|
|
xlog.Error("failed to update session", "error", err)
|
|
sendError(t, "session_update_error", "Failed to update session", "", "")
|
|
continue
|
|
}
|
|
|
|
toggleVAD()
|
|
|
|
sendEvent(t, types.SessionUpdatedEvent{
|
|
ServerEventBase: types.ServerEventBase{
|
|
EventID: "event_TODO",
|
|
},
|
|
Session: session.ToServer(),
|
|
})
|
|
}
|
|
|
|
case types.InputAudioBufferAppendEvent:
|
|
// Handle 'input_audio_buffer.append'
|
|
if e.Audio == "" {
|
|
xlog.Error("Audio data is missing in 'input_audio_buffer.append'")
|
|
sendError(t, "missing_audio_data", "Audio data is missing", "", "")
|
|
continue
|
|
}
|
|
|
|
// Decode base64 audio data
|
|
decodedAudio, err := base64.StdEncoding.DecodeString(e.Audio)
|
|
if err != nil {
|
|
xlog.Error("failed to decode audio data", "error", err)
|
|
sendError(t, "invalid_audio_data", "Failed to decode audio data", "", "")
|
|
continue
|
|
}
|
|
|
|
// Check buffer size limits before appending
|
|
session.AudioBufferLock.Lock()
|
|
newSize := len(session.InputAudioBuffer) + len(decodedAudio)
|
|
if newSize > maxAudioBufferSize {
|
|
session.AudioBufferLock.Unlock()
|
|
xlog.Error("audio buffer size limit exceeded", "current_size", len(session.InputAudioBuffer), "incoming_size", len(decodedAudio), "limit", maxAudioBufferSize)
|
|
sendError(t, "buffer_size_exceeded", fmt.Sprintf("Audio buffer size limit exceeded (max %d bytes)", maxAudioBufferSize), "", "")
|
|
continue
|
|
}
|
|
|
|
// Append to InputAudioBuffer
|
|
session.InputAudioBuffer = append(session.InputAudioBuffer, decodedAudio...)
|
|
session.AudioBufferLock.Unlock()
|
|
|
|
case types.InputAudioBufferCommitEvent:
|
|
xlog.Debug("recv", "message", string(msg))
|
|
|
|
sessionLock.Lock()
|
|
isServerVAD := session.TurnDetection != nil && session.TurnDetection.ServerVad != nil
|
|
sessionLock.Unlock()
|
|
|
|
// TODO: At the least need to check locking and timer state in the VAD Go routine before allowing this
|
|
if isServerVAD {
|
|
sendNotImplemented(t, "input_audio_buffer.commit in conjunction with VAD")
|
|
continue
|
|
}
|
|
|
|
session.AudioBufferLock.Lock()
|
|
allAudio := make([]byte, len(session.InputAudioBuffer))
|
|
copy(allAudio, session.InputAudioBuffer)
|
|
session.InputAudioBuffer = nil
|
|
session.AudioBufferLock.Unlock()
|
|
|
|
sendEvent(t, types.InputAudioBufferCommittedEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ItemID: generateItemID(),
|
|
})
|
|
|
|
respCtx, respDone := session.startResponse(context.Background())
|
|
go func() {
|
|
defer close(respDone)
|
|
commitUtterance(respCtx, allAudio, session, conversation, t)
|
|
}()
|
|
|
|
case types.ConversationItemCreateEvent:
|
|
xlog.Debug("recv", "message", string(msg))
|
|
// Add the item to the conversation
|
|
item := e.Item
|
|
// Ensure IDs are present
|
|
if item.User != nil && item.User.ID == "" {
|
|
item.User.ID = generateItemID()
|
|
}
|
|
if item.Assistant != nil && item.Assistant.ID == "" {
|
|
item.Assistant.ID = generateItemID()
|
|
}
|
|
if item.System != nil && item.System.ID == "" {
|
|
item.System.ID = generateItemID()
|
|
}
|
|
if item.FunctionCall != nil && item.FunctionCall.ID == "" {
|
|
item.FunctionCall.ID = generateItemID()
|
|
}
|
|
if item.FunctionCallOutput != nil && item.FunctionCallOutput.ID == "" {
|
|
item.FunctionCallOutput.ID = generateItemID()
|
|
}
|
|
|
|
conversation.Lock.Lock()
|
|
conversation.Items = append(conversation.Items, &item)
|
|
conversation.Lock.Unlock()
|
|
|
|
sendEvent(t, types.ConversationItemAddedEvent{
|
|
ServerEventBase: types.ServerEventBase{
|
|
EventID: e.EventID,
|
|
},
|
|
PreviousItemID: e.PreviousItemID,
|
|
Item: item,
|
|
})
|
|
|
|
case types.ConversationItemDeleteEvent:
|
|
sendError(t, "not_implemented", "Deleting items not implemented", "", "event_TODO")
|
|
|
|
case types.ConversationItemRetrieveEvent:
|
|
xlog.Debug("recv", "message", string(msg))
|
|
|
|
if e.ItemID == "" {
|
|
sendError(t, "invalid_item_id", "Need item_id, but none specified", "", "event_TODO")
|
|
continue
|
|
}
|
|
|
|
conversation.Lock.Lock()
|
|
var retrievedItem types.MessageItemUnion
|
|
for _, item := range conversation.Items {
|
|
// We need to check ID in the union
|
|
var id string
|
|
if item.System != nil {
|
|
id = item.System.ID
|
|
} else if item.User != nil {
|
|
id = item.User.ID
|
|
} else if item.Assistant != nil {
|
|
id = item.Assistant.ID
|
|
} else if item.FunctionCall != nil {
|
|
id = item.FunctionCall.ID
|
|
} else if item.FunctionCallOutput != nil {
|
|
id = item.FunctionCallOutput.ID
|
|
}
|
|
|
|
if id == e.ItemID {
|
|
retrievedItem = *item
|
|
break
|
|
}
|
|
}
|
|
conversation.Lock.Unlock()
|
|
|
|
sendEvent(t, types.ConversationItemRetrievedEvent{
|
|
ServerEventBase: types.ServerEventBase{
|
|
EventID: "event_TODO",
|
|
},
|
|
Item: retrievedItem,
|
|
})
|
|
|
|
case types.ResponseCreateEvent:
|
|
xlog.Debug("recv", "message", string(msg))
|
|
|
|
// Handle optional items to add to context
|
|
if len(e.Response.Input) > 0 {
|
|
conversation.Lock.Lock()
|
|
for _, item := range e.Response.Input {
|
|
// Ensure IDs are present
|
|
if item.User != nil && item.User.ID == "" {
|
|
item.User.ID = generateItemID()
|
|
}
|
|
if item.Assistant != nil && item.Assistant.ID == "" {
|
|
item.Assistant.ID = generateItemID()
|
|
}
|
|
if item.System != nil && item.System.ID == "" {
|
|
item.System.ID = generateItemID()
|
|
}
|
|
if item.FunctionCall != nil && item.FunctionCall.ID == "" {
|
|
item.FunctionCall.ID = generateItemID()
|
|
}
|
|
if item.FunctionCallOutput != nil && item.FunctionCallOutput.ID == "" {
|
|
item.FunctionCallOutput.ID = generateItemID()
|
|
}
|
|
|
|
conversation.Items = append(conversation.Items, &item)
|
|
}
|
|
conversation.Lock.Unlock()
|
|
}
|
|
|
|
respCtx, respDone := session.startResponse(context.Background())
|
|
go func() {
|
|
defer close(respDone)
|
|
triggerResponse(respCtx, session, conversation, t, &e.Response)
|
|
}()
|
|
|
|
case types.ResponseCancelEvent:
|
|
xlog.Debug("recv", "message", string(msg))
|
|
session.cancelActiveResponse()
|
|
|
|
default:
|
|
xlog.Error("unknown message type")
|
|
// sendError(t, "unknown_message_type", fmt.Sprintf("Unknown message type: %s", incomingMsg.Type), "", "")
|
|
}
|
|
}
|
|
|
|
// Cancel any in-flight response before tearing down
|
|
session.cancelActiveResponse()
|
|
|
|
// Stop the Opus decode goroutine (if running)
|
|
if decodeDone != nil {
|
|
close(decodeDone)
|
|
}
|
|
|
|
// Signal any running VAD goroutine to exit.
|
|
if vadServerStarted {
|
|
close(done)
|
|
}
|
|
wg.Wait()
|
|
|
|
// Remove the session from the sessions map
|
|
sessionLock.Lock()
|
|
delete(sessions, sessionID)
|
|
sessionLock.Unlock()
|
|
}
|
|
|
|
// sendEvent sends a server event via the transport, logging any errors.
|
|
func sendEvent(t Transport, event types.ServerEvent) {
|
|
if err := t.SendEvent(event); err != nil {
|
|
xlog.Error("write error", "error", err)
|
|
}
|
|
}
|
|
|
|
// sendError sends an error event to the client.
|
|
func sendError(t Transport, code, message, param, eventID string) {
|
|
errorEvent := types.ErrorEvent{
|
|
ServerEventBase: types.ServerEventBase{
|
|
EventID: eventID,
|
|
},
|
|
Error: types.Error{
|
|
Type: "invalid_request_error",
|
|
Code: code,
|
|
Message: message,
|
|
Param: param,
|
|
EventID: eventID,
|
|
},
|
|
}
|
|
|
|
sendEvent(t, errorEvent)
|
|
}
|
|
|
|
func sendNotImplemented(t Transport, message string) {
|
|
sendError(t, "not_implemented", message, "", "event_TODO")
|
|
}
|
|
|
|
// sendTestTone generates a 1-second 440 Hz sine wave and sends it through
|
|
// the transport's audio path. This exercises the full Opus encode → RTP →
|
|
// browser decode pipeline without involving TTS.
|
|
func sendTestTone(t Transport) {
|
|
const (
|
|
freq = 440.0
|
|
sampleRate = 24000
|
|
duration = 1 // seconds
|
|
amplitude = 16000
|
|
numSamples = sampleRate * duration
|
|
)
|
|
|
|
pcm := make([]byte, numSamples*2) // 16-bit samples = 2 bytes each
|
|
for i := range numSamples {
|
|
sample := int16(amplitude * math.Sin(2*math.Pi*freq*float64(i)/sampleRate))
|
|
binary.LittleEndian.PutUint16(pcm[i*2:], uint16(sample))
|
|
}
|
|
|
|
xlog.Debug("Sending test tone", "samples", numSamples, "sample_rate", sampleRate, "freq", freq)
|
|
if err := t.SendAudio(context.Background(), pcm, sampleRate); err != nil {
|
|
xlog.Error("test tone send failed", "error", err)
|
|
}
|
|
}
|
|
|
|
func updateTransSession(session *Session, update *types.SessionUnion, cl *config.ModelConfigLoader, ml *model.ModelLoader, appConfig *config.ApplicationConfig) error {
|
|
sessionLock.Lock()
|
|
defer sessionLock.Unlock()
|
|
|
|
// In transcription session update, we look at Transcription field
|
|
if update.Transcription == nil || update.Transcription.Audio == nil || update.Transcription.Audio.Input == nil {
|
|
return nil
|
|
}
|
|
|
|
trUpd := update.Transcription.Audio.Input.Transcription
|
|
trCur := session.InputAudioTranscription
|
|
|
|
session.TranscriptionOnly = true
|
|
|
|
if trUpd != nil && trUpd.Model != "" && trUpd.Model != trCur.Model {
|
|
cfg, err := cl.LoadModelConfigFileByNameDefaultOptions(trUpd.Model, appConfig)
|
|
if err != nil {
|
|
return err
|
|
}
|
|
if cfg == nil || (cfg.Pipeline.VAD == "" || cfg.Pipeline.Transcription == "") {
|
|
return fmt.Errorf("model is not a valid pipeline model: %s", trUpd.Model)
|
|
}
|
|
|
|
m, cfg, err := newTranscriptionOnlyModel(&cfg.Pipeline, cl, ml, appConfig)
|
|
if err != nil {
|
|
return err
|
|
}
|
|
|
|
session.ModelInterface = m
|
|
session.ModelConfig = cfg
|
|
}
|
|
|
|
if trUpd != nil {
|
|
trCur.Language = trUpd.Language
|
|
trCur.Prompt = trUpd.Prompt
|
|
}
|
|
|
|
if update.Transcription.Audio.Input.TurnDetectionSet {
|
|
session.TurnDetection = update.Transcription.Audio.Input.TurnDetection
|
|
}
|
|
|
|
if update.Transcription.Audio.Input.Format != nil && update.Transcription.Audio.Input.Format.PCM != nil {
|
|
if update.Transcription.Audio.Input.Format.PCM.Rate > 0 {
|
|
session.InputSampleRate = update.Transcription.Audio.Input.Format.PCM.Rate
|
|
}
|
|
}
|
|
|
|
return nil
|
|
}
|
|
|
|
func updateSession(session *Session, update *types.SessionUnion, cl *config.ModelConfigLoader, ml *model.ModelLoader, appConfig *config.ApplicationConfig, evaluator *templates.Evaluator, routing *RealtimeRoutingContext) error {
|
|
sessionLock.Lock()
|
|
defer sessionLock.Unlock()
|
|
|
|
if update.Realtime == nil {
|
|
return nil
|
|
}
|
|
|
|
session.TranscriptionOnly = false
|
|
rt := update.Realtime
|
|
|
|
if rt.Model != "" {
|
|
cfg, err := cl.LoadModelConfigFileByNameDefaultOptions(rt.Model, appConfig)
|
|
if err != nil {
|
|
return err
|
|
}
|
|
if cfg == nil || (cfg.Pipeline.VAD == "" || cfg.Pipeline.Transcription == "" || cfg.Pipeline.TTS == "" || cfg.Pipeline.LLM == "") {
|
|
return fmt.Errorf("model is not a valid pipeline model: %s", rt.Model)
|
|
}
|
|
|
|
if session.InputAudioTranscription == nil {
|
|
session.InputAudioTranscription = &types.AudioTranscription{}
|
|
}
|
|
session.InputAudioTranscription.Model = cfg.Pipeline.Transcription
|
|
session.Voice = cfg.TTSConfig.Voice
|
|
session.Model = rt.Model
|
|
session.ModelConfig = cfg
|
|
}
|
|
|
|
if rt.Audio != nil && rt.Audio.Output != nil && rt.Audio.Output.Voice != "" {
|
|
session.Voice = string(rt.Audio.Output.Voice)
|
|
}
|
|
|
|
if rt.Audio != nil && rt.Audio.Input != nil && rt.Audio.Input.Transcription != nil {
|
|
session.InputAudioTranscription = rt.Audio.Input.Transcription
|
|
session.ModelConfig.Pipeline.Transcription = rt.Audio.Input.Transcription.Model
|
|
}
|
|
|
|
if rt.Model != "" || (rt.Audio != nil && rt.Audio.Output != nil && rt.Audio.Output.Voice != "") || (rt.Audio != nil && rt.Audio.Input != nil && rt.Audio.Input.Transcription != nil) {
|
|
m, err := newModel(&session.ModelConfig.Pipeline, cl, ml, appConfig, evaluator, routing)
|
|
if err != nil {
|
|
return err
|
|
}
|
|
session.ModelInterface = m
|
|
}
|
|
|
|
if rt.Audio != nil && rt.Audio.Input != nil && rt.Audio.Input.TurnDetectionSet {
|
|
session.TurnDetection = rt.Audio.Input.TurnDetection
|
|
}
|
|
|
|
if rt.Audio != nil && rt.Audio.Input != nil && rt.Audio.Input.Format != nil && rt.Audio.Input.Format.PCM != nil {
|
|
if rt.Audio.Input.Format.PCM.Rate > 0 {
|
|
session.InputSampleRate = rt.Audio.Input.Format.PCM.Rate
|
|
}
|
|
}
|
|
|
|
if rt.Audio != nil && rt.Audio.Output != nil && rt.Audio.Output.Format != nil && rt.Audio.Output.Format.PCM != nil {
|
|
if rt.Audio.Output.Format.PCM.Rate > 0 {
|
|
session.OutputSampleRate = rt.Audio.Output.Format.PCM.Rate
|
|
}
|
|
}
|
|
|
|
if rt.Instructions != "" {
|
|
session.Instructions = rt.Instructions
|
|
}
|
|
|
|
if rt.Tools != nil {
|
|
// Manage Mode tools survive a client-driven session.update — the
|
|
// alternative is silently dropping them whenever the user toggles
|
|
// a client MCP server, which would break the modality mid-session.
|
|
// Names from rt.Tools win on collision (the client is explicit;
|
|
// we preserve, we don't override).
|
|
merged := append([]types.ToolUnion(nil), rt.Tools...)
|
|
seen := make(map[string]struct{}, len(merged))
|
|
for _, t := range merged {
|
|
if t.Function != nil {
|
|
seen[t.Function.Name] = struct{}{}
|
|
}
|
|
}
|
|
for _, t := range session.AssistantTools {
|
|
if t.Function == nil {
|
|
continue
|
|
}
|
|
if _, ok := seen[t.Function.Name]; ok {
|
|
continue
|
|
}
|
|
merged = append(merged, t)
|
|
}
|
|
session.Tools = merged
|
|
}
|
|
if rt.ToolChoice != nil {
|
|
session.ToolChoice = rt.ToolChoice
|
|
}
|
|
|
|
if rt.MaxOutputTokens != 0 {
|
|
session.MaxOutputTokens = rt.MaxOutputTokens
|
|
}
|
|
|
|
if len(rt.OutputModalities) > 0 {
|
|
session.OutputModalities = rt.OutputModalities
|
|
}
|
|
|
|
return nil
|
|
}
|
|
|
|
// decodeOpusLoop runs a ticker that drains buffered raw Opus frames from the
|
|
// session, decodes them in a single batched gRPC call, and appends the
|
|
// resulting PCM to InputAudioBuffer. This gives ~3 gRPC calls/sec instead of
|
|
// 50 (one per RTP packet) and keeps decode diagnostics once-per-batch.
|
|
func decodeOpusLoop(session *Session, opusBackend grpc.Backend, done chan struct{}) {
|
|
ticker := time.NewTicker(300 * time.Millisecond)
|
|
defer ticker.Stop()
|
|
for {
|
|
select {
|
|
case <-ticker.C:
|
|
session.OpusFramesLock.Lock()
|
|
frames := session.OpusFrames
|
|
session.OpusFrames = nil
|
|
session.OpusFramesLock.Unlock()
|
|
if len(frames) == 0 {
|
|
continue
|
|
}
|
|
|
|
result, err := opusBackend.AudioDecode(context.Background(), &proto.AudioDecodeRequest{
|
|
Frames: frames,
|
|
Options: map[string]string{
|
|
"session_id": session.ID,
|
|
},
|
|
})
|
|
if err != nil {
|
|
xlog.Warn("opus decode batch error", "error", err, "frames", len(frames))
|
|
continue
|
|
}
|
|
|
|
samples := sound.BytesToInt16sLE(result.PcmData)
|
|
|
|
xlog.Debug("opus decode batch",
|
|
"frames", len(frames),
|
|
"decoded_samples", len(samples),
|
|
"sample_rate", result.SampleRate,
|
|
)
|
|
|
|
// Resample from 48kHz to session input rate (16kHz) if needed
|
|
if result.SampleRate != int32(session.InputSampleRate) {
|
|
samples = sound.ResampleInt16(samples, int(result.SampleRate), session.InputSampleRate)
|
|
}
|
|
|
|
pcmBytes := sound.Int16toBytesLE(samples)
|
|
session.AudioBufferLock.Lock()
|
|
newSize := len(session.InputAudioBuffer) + len(pcmBytes)
|
|
if newSize <= maxAudioBufferSize {
|
|
session.InputAudioBuffer = append(session.InputAudioBuffer, pcmBytes...)
|
|
}
|
|
session.AudioBufferLock.Unlock()
|
|
case <-done:
|
|
return
|
|
}
|
|
}
|
|
}
|
|
|
|
// handleVAD is a goroutine that listens for audio data from the client,
|
|
// runs VAD on the audio data, and commits utterances to the conversation
|
|
func handleVAD(session *Session, conv *Conversation, t Transport, done chan struct{}) {
|
|
vadContext, cancel := context.WithCancel(context.Background())
|
|
go func() {
|
|
<-done
|
|
cancel()
|
|
}()
|
|
|
|
silenceThreshold := 0.5 // Default 500ms
|
|
if session.TurnDetection != nil && session.TurnDetection.ServerVad != nil {
|
|
silenceThreshold = float64(session.TurnDetection.ServerVad.SilenceDurationMs) / 1000
|
|
}
|
|
|
|
speechStarted := false
|
|
startTime := time.Now()
|
|
|
|
ticker := time.NewTicker(300 * time.Millisecond)
|
|
defer ticker.Stop()
|
|
|
|
for {
|
|
select {
|
|
case <-done:
|
|
return
|
|
case <-ticker.C:
|
|
session.AudioBufferLock.Lock()
|
|
allAudio := make([]byte, len(session.InputAudioBuffer))
|
|
copy(allAudio, session.InputAudioBuffer)
|
|
session.AudioBufferLock.Unlock()
|
|
|
|
aints := sound.BytesToInt16sLE(allAudio)
|
|
if len(aints) == 0 || len(aints) < int(silenceThreshold*float64(session.InputSampleRate)) {
|
|
continue
|
|
}
|
|
|
|
// Resample from InputSampleRate to 16kHz
|
|
aints = sound.ResampleInt16(aints, session.InputSampleRate, localSampleRate)
|
|
|
|
segments, err := runVAD(vadContext, session, aints)
|
|
if err != nil {
|
|
if err.Error() == "unexpected speech end" {
|
|
xlog.Debug("VAD cancelled")
|
|
continue
|
|
}
|
|
xlog.Error("failed to process audio", "error", err)
|
|
sendError(t, "processing_error", "Failed to process audio: "+err.Error(), "", "")
|
|
continue
|
|
}
|
|
|
|
audioLength := float64(len(aints)) / localSampleRate
|
|
|
|
// TODO: When resetting the buffer we should retain a small postfix
|
|
if len(segments) == 0 && audioLength > silenceThreshold {
|
|
session.AudioBufferLock.Lock()
|
|
session.InputAudioBuffer = nil
|
|
session.AudioBufferLock.Unlock()
|
|
|
|
continue
|
|
} else if len(segments) == 0 {
|
|
continue
|
|
}
|
|
|
|
if !speechStarted {
|
|
// Barge-in: cancel any in-flight response so we stop
|
|
// sending audio and don't keep the interrupted reply in history.
|
|
session.cancelActiveResponse()
|
|
|
|
sendEvent(t, types.InputAudioBufferSpeechStartedEvent{
|
|
ServerEventBase: types.ServerEventBase{
|
|
EventID: "event_TODO",
|
|
},
|
|
AudioStartMs: time.Since(startTime).Milliseconds(),
|
|
})
|
|
speechStarted = true
|
|
}
|
|
|
|
// Segment still in progress when audio ended
|
|
segEndTime := segments[len(segments)-1].End
|
|
if segEndTime == 0 {
|
|
continue
|
|
}
|
|
|
|
if float32(audioLength)-segEndTime > float32(silenceThreshold) {
|
|
xlog.Debug("Detected end of speech segment")
|
|
session.AudioBufferLock.Lock()
|
|
session.InputAudioBuffer = nil
|
|
session.AudioBufferLock.Unlock()
|
|
|
|
sendEvent(t, types.InputAudioBufferSpeechStoppedEvent{
|
|
ServerEventBase: types.ServerEventBase{
|
|
EventID: "event_TODO",
|
|
},
|
|
AudioEndMs: time.Since(startTime).Milliseconds(),
|
|
})
|
|
speechStarted = false
|
|
|
|
sendEvent(t, types.InputAudioBufferCommittedEvent{
|
|
ServerEventBase: types.ServerEventBase{
|
|
EventID: "event_TODO",
|
|
},
|
|
ItemID: generateItemID(),
|
|
PreviousItemID: "TODO",
|
|
})
|
|
|
|
abytes := sound.Int16toBytesLE(aints)
|
|
// TODO: Remove prefix silence that is is over TurnDetectionParams.PrefixPaddingMs
|
|
respCtx, respDone := session.startResponse(vadContext)
|
|
go func() {
|
|
defer close(respDone)
|
|
commitUtterance(respCtx, abytes, session, conv, t)
|
|
}()
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
func commitUtterance(ctx context.Context, utt []byte, session *Session, conv *Conversation, t Transport) {
|
|
if len(utt) == 0 {
|
|
return
|
|
}
|
|
|
|
f, err := os.CreateTemp("", "realtime-audio-chunk-*.wav")
|
|
if err != nil {
|
|
xlog.Error("failed to create temp file", "error", err)
|
|
return
|
|
}
|
|
defer f.Close()
|
|
defer os.Remove(f.Name())
|
|
xlog.Debug("Writing to file", "file", f.Name())
|
|
|
|
hdr := laudio.NewWAVHeader(uint32(len(utt)))
|
|
if err := hdr.Write(f); err != nil {
|
|
xlog.Error("Failed to write WAV header", "error", err)
|
|
return
|
|
}
|
|
|
|
if _, err := f.Write(utt); err != nil {
|
|
xlog.Error("Failed to write audio data", "error", err)
|
|
return
|
|
}
|
|
|
|
f.Sync()
|
|
|
|
// TODO: If we have a real any-to-any model then transcription is optional
|
|
var transcript string
|
|
if session.InputAudioTranscription != nil {
|
|
// emitTranscription streams transcript deltas when
|
|
// pipeline.streaming.transcription is set, otherwise emits a single
|
|
// completed event; either way it returns the final transcript text.
|
|
var err error
|
|
transcript, err = emitTranscription(ctx, t, session, generateItemID(), f.Name())
|
|
if err != nil {
|
|
sendError(t, "transcription_failed", err.Error(), "", "event_TODO")
|
|
return
|
|
}
|
|
} else {
|
|
sendNotImplemented(t, "any-to-any models")
|
|
return
|
|
}
|
|
|
|
if !session.TranscriptionOnly {
|
|
generateResponse(ctx, session, utt, transcript, conv, t)
|
|
}
|
|
}
|
|
|
|
func runVAD(ctx context.Context, session *Session, adata []int16) ([]schema.VADSegment, error) {
|
|
soundIntBuffer := &audio.IntBuffer{
|
|
Format: &audio.Format{SampleRate: localSampleRate, NumChannels: 1},
|
|
SourceBitDepth: 16,
|
|
Data: sound.ConvertInt16ToInt(adata),
|
|
}
|
|
|
|
float32Data := soundIntBuffer.AsFloat32Buffer().Data
|
|
|
|
resp, err := session.ModelInterface.VAD(ctx, &schema.VADRequest{
|
|
Audio: float32Data,
|
|
})
|
|
if err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
// If resp.Segments is empty => no speech
|
|
return resp.Segments, nil
|
|
}
|
|
|
|
// Function to generate a response based on the conversation
|
|
func generateResponse(ctx context.Context, session *Session, utt []byte, transcript string, conv *Conversation, t Transport) {
|
|
xlog.Debug("Generating realtime response...")
|
|
|
|
// Create user message item
|
|
item := types.MessageItemUnion{
|
|
User: &types.MessageItemUser{
|
|
ID: generateItemID(),
|
|
Status: types.ItemStatusCompleted,
|
|
Content: []types.MessageContentInput{
|
|
{
|
|
Type: types.MessageContentTypeInputAudio,
|
|
Audio: base64.StdEncoding.EncodeToString(utt),
|
|
Transcript: transcript,
|
|
},
|
|
},
|
|
},
|
|
}
|
|
conv.Lock.Lock()
|
|
conv.Items = append(conv.Items, &item)
|
|
conv.Lock.Unlock()
|
|
|
|
sendEvent(t, types.ConversationItemAddedEvent{
|
|
Item: item,
|
|
})
|
|
|
|
triggerResponse(ctx, session, conv, t, nil)
|
|
}
|
|
|
|
// maxAssistantToolTurns caps the server-side agentic loop. Mirrors the
|
|
// chat-page maxToolTurns:10 from useChat.js — the model gets up to this
|
|
// many consecutive tool round-trips before we return control to the user
|
|
// without another response cycle.
|
|
const maxAssistantToolTurns = 10
|
|
|
|
func triggerResponse(ctx context.Context, session *Session, conv *Conversation, t Transport, overrides *types.ResponseCreateParams) {
|
|
triggerResponseAtTurn(ctx, session, conv, t, overrides, 0)
|
|
}
|
|
|
|
func triggerResponseAtTurn(ctx context.Context, session *Session, conv *Conversation, t Transport, overrides *types.ResponseCreateParams, toolTurn int) {
|
|
config := session.ModelInterface.PredictConfig()
|
|
|
|
// Default values
|
|
tools := session.Tools
|
|
toolChoice := session.ToolChoice
|
|
instructions := session.Instructions
|
|
maxOutputTokens := session.MaxOutputTokens
|
|
// Overrides
|
|
if overrides != nil {
|
|
if overrides.Tools != nil {
|
|
tools = overrides.Tools
|
|
}
|
|
if overrides.ToolChoice != nil {
|
|
toolChoice = overrides.ToolChoice
|
|
}
|
|
if overrides.Instructions != "" {
|
|
instructions = overrides.Instructions
|
|
}
|
|
if overrides.MaxOutputTokens != 0 {
|
|
maxOutputTokens = overrides.MaxOutputTokens
|
|
}
|
|
}
|
|
|
|
// Apply MaxOutputTokens to model config if specified
|
|
// Save original value to restore after prediction
|
|
var originalMaxTokens *int
|
|
if config != nil {
|
|
originalMaxTokens = config.Maxtokens
|
|
if maxOutputTokens != 0 && !maxOutputTokens.IsInf() {
|
|
tokenValue := int(maxOutputTokens)
|
|
config.Maxtokens = &tokenValue
|
|
xlog.Debug("Applied max_output_tokens to config", "value", tokenValue)
|
|
}
|
|
}
|
|
// Defer restoration of original value
|
|
defer func() {
|
|
if config != nil {
|
|
config.Maxtokens = originalMaxTokens
|
|
}
|
|
}()
|
|
|
|
var conversationHistory schema.Messages
|
|
conversationHistory = append(conversationHistory, schema.Message{
|
|
Role: string(types.MessageRoleSystem),
|
|
StringContent: instructions,
|
|
Content: instructions,
|
|
})
|
|
|
|
imgIndex := 0
|
|
conv.Lock.Lock()
|
|
items := trimRealtimeItems(conv.Items, session.MaxHistoryItems)
|
|
for _, item := range items {
|
|
if item.User != nil {
|
|
msg := schema.Message{
|
|
Role: string(types.MessageRoleUser),
|
|
}
|
|
textContent := ""
|
|
nrOfImgsInMessage := 0
|
|
for _, content := range item.User.Content {
|
|
switch content.Type {
|
|
case types.MessageContentTypeInputText:
|
|
textContent += content.Text
|
|
case types.MessageContentTypeInputAudio:
|
|
textContent += content.Transcript
|
|
case types.MessageContentTypeInputImage:
|
|
img, err := utils.GetContentURIAsBase64(content.ImageURL)
|
|
if err != nil {
|
|
xlog.Warn("Failed to process image", "error", err)
|
|
continue
|
|
}
|
|
msg.StringImages = append(msg.StringImages, img)
|
|
imgIndex++
|
|
nrOfImgsInMessage++
|
|
}
|
|
}
|
|
if nrOfImgsInMessage > 0 && !config.TemplateConfig.UseTokenizerTemplate {
|
|
templated, err := templates.TemplateMultiModal(config.TemplateConfig.Multimodal, templates.MultiModalOptions{
|
|
TotalImages: imgIndex,
|
|
ImagesInMessage: nrOfImgsInMessage,
|
|
}, textContent)
|
|
if err != nil {
|
|
xlog.Warn("Failed to apply multimodal template", "error", err)
|
|
templated = textContent
|
|
}
|
|
msg.StringContent = templated
|
|
msg.Content = templated
|
|
} else {
|
|
msg.StringContent = textContent
|
|
msg.Content = textContent
|
|
}
|
|
conversationHistory = append(conversationHistory, msg)
|
|
} else if item.Assistant != nil {
|
|
for _, content := range item.Assistant.Content {
|
|
switch content.Type {
|
|
case types.MessageContentTypeOutputText:
|
|
conversationHistory = append(conversationHistory, schema.Message{
|
|
Role: string(types.MessageRoleAssistant),
|
|
StringContent: content.Text,
|
|
Content: content.Text,
|
|
})
|
|
case types.MessageContentTypeOutputAudio:
|
|
conversationHistory = append(conversationHistory, schema.Message{
|
|
Role: string(types.MessageRoleAssistant),
|
|
StringContent: content.Transcript,
|
|
Content: content.Transcript,
|
|
StringAudios: []string{content.Audio},
|
|
})
|
|
}
|
|
}
|
|
} else if item.System != nil {
|
|
for _, content := range item.System.Content {
|
|
conversationHistory = append(conversationHistory, schema.Message{
|
|
Role: string(types.MessageRoleSystem),
|
|
StringContent: content.Text,
|
|
Content: content.Text,
|
|
})
|
|
}
|
|
} else if item.FunctionCall != nil {
|
|
conversationHistory = append(conversationHistory, schema.Message{
|
|
Role: string(types.MessageRoleAssistant),
|
|
ToolCalls: []schema.ToolCall{
|
|
{
|
|
ID: item.FunctionCall.CallID,
|
|
Type: "function",
|
|
FunctionCall: schema.FunctionCall{
|
|
Name: item.FunctionCall.Name,
|
|
Arguments: item.FunctionCall.Arguments,
|
|
},
|
|
},
|
|
},
|
|
})
|
|
} else if item.FunctionCallOutput != nil {
|
|
conversationHistory = append(conversationHistory, schema.Message{
|
|
Role: "tool",
|
|
Name: item.FunctionCallOutput.CallID,
|
|
Content: item.FunctionCallOutput.Output,
|
|
StringContent: item.FunctionCallOutput.Output,
|
|
})
|
|
}
|
|
}
|
|
conv.Lock.Unlock()
|
|
|
|
var images []string
|
|
for _, m := range conversationHistory {
|
|
images = append(images, m.StringImages...)
|
|
}
|
|
|
|
responseID := generateUniqueID()
|
|
sendEvent(t, types.ResponseCreatedEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
Response: types.Response{
|
|
ID: responseID,
|
|
Object: "realtime.response",
|
|
Status: types.ResponseStatusInProgress,
|
|
},
|
|
})
|
|
|
|
// Streamed LLM path: when the pipeline opts into LLM streaming, stream the
|
|
// transcript to the client as it is generated and synthesize the buffered
|
|
// message once. Tool turns are supported only when the model uses its
|
|
// tokenizer template: the C++ autoparser then delivers content and tool
|
|
// calls via ChatDeltas (clearing the text stream), so the spoken transcript
|
|
// never leaks tool-call tokens. Grammar-based function calling emits the
|
|
// call as JSON in the token stream, so those turns keep the buffered path.
|
|
if config != nil && session.ModelConfig != nil && session.ModelConfig.Pipeline.StreamLLM() {
|
|
canStream := len(tools) == 0 || config.TemplateConfig.UseTokenizerTemplate
|
|
var respMods []types.Modality
|
|
if overrides != nil {
|
|
respMods = overrides.OutputModalities
|
|
}
|
|
if canStream && modalitiesContainAudio(resolveOutputModalities(session.OutputModalities, respMods)) {
|
|
if streamLLMResponse(ctx, session, conv, t, responseID, conversationHistory, images, config, tools, toolChoice, toolTurn) {
|
|
return
|
|
}
|
|
}
|
|
}
|
|
|
|
predFunc, err := session.ModelInterface.Predict(ctx, conversationHistory, images, nil, nil, nil, tools, toolChoice, nil, nil, nil)
|
|
if err != nil {
|
|
sendError(t, "inference_failed", fmt.Sprintf("backend error: %v", err), "", "") // item.Assistant.ID is unknown here
|
|
return
|
|
}
|
|
|
|
pred, err := predFunc()
|
|
if err != nil {
|
|
sendError(t, "prediction_failed", fmt.Sprintf("backend error: %v", err), "", "")
|
|
return
|
|
}
|
|
|
|
// Check for cancellation after LLM inference (barge-in may have fired)
|
|
if ctx.Err() != nil {
|
|
xlog.Debug("Response cancelled after LLM inference (barge-in)")
|
|
sendEvent(t, types.ResponseDoneEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
Response: types.Response{
|
|
ID: responseID,
|
|
Object: "realtime.response",
|
|
Status: types.ResponseStatusCancelled,
|
|
},
|
|
})
|
|
return
|
|
}
|
|
|
|
xlog.Debug("Function config for parsing", "function_name_key", config.FunctionsConfig.FunctionNameKey, "function_arguments_key", config.FunctionsConfig.FunctionArgumentsKey)
|
|
xlog.Debug("LLM raw response", "text", pred.Response, "response_length", len(pred.Response), "usage", pred.Usage)
|
|
|
|
// Safely dereference pointer fields for logging
|
|
maxTokens := "nil"
|
|
if config.Maxtokens != nil {
|
|
maxTokens = fmt.Sprintf("%d", *config.Maxtokens)
|
|
}
|
|
contextSize := "nil"
|
|
if config.ContextSize != nil {
|
|
contextSize = fmt.Sprintf("%d", *config.ContextSize)
|
|
}
|
|
xlog.Debug("Model parameters", "max_tokens", maxTokens, "context_size", contextSize, "stopwords", config.StopWords)
|
|
|
|
rawResponse := pred.Response
|
|
if config.TemplateConfig.ReplyPrefix != "" {
|
|
rawResponse = config.TemplateConfig.ReplyPrefix + rawResponse
|
|
}
|
|
|
|
// Detect thinking start token from template for reasoning extraction
|
|
var template string
|
|
if config.TemplateConfig.UseTokenizerTemplate {
|
|
template = config.GetModelTemplate()
|
|
} else {
|
|
template = config.TemplateConfig.Chat
|
|
}
|
|
thinkingStartToken := reasoning.DetectThinkingStartToken(template, &config.ReasoningConfig)
|
|
|
|
// When the C++ autoparser emitted ChatDeltas with actionable data,
|
|
// prefer them — the backend clears Reply.Message in that path and
|
|
// delivers parsed content/reasoning/tool-calls via the delta stream
|
|
// (see pkg/functions/chat_deltas.go, mirrored from chat.go's non-SSE
|
|
// handling). Without this, Response is empty and realtime would
|
|
// synthesize silence for replies that actually produced tokens.
|
|
var reasoningText, responseWithoutReasoning, textContent, cleanedResponse string
|
|
var toolCalls []functions.FuncCallResults
|
|
deltaToolCalls := functions.ToolCallsFromChatDeltas(pred.ChatDeltas)
|
|
deltaContent := functions.ContentFromChatDeltas(pred.ChatDeltas)
|
|
deltaReasoning := functions.ReasoningFromChatDeltas(pred.ChatDeltas)
|
|
if len(deltaToolCalls) > 0 || deltaContent != "" {
|
|
xlog.Debug("[ChatDeltas] realtime: using C++ autoparser deltas",
|
|
"tool_calls", len(deltaToolCalls),
|
|
"content_len", len(deltaContent),
|
|
"reasoning_len", len(deltaReasoning))
|
|
// Issue #9985: when the autoparser only delivered content (no
|
|
// reasoning_content), it may be running in the "pure content"
|
|
// PEG fallback (non-jinja path) which leaves <think>…</think>
|
|
// embedded in the content. Run Go-side extraction defensively.
|
|
// ExtractReasoningWithConfig is a no-op when no tag pair matches,
|
|
// so it's safe to apply unconditionally in the no-reasoning branch.
|
|
if deltaReasoning == "" && deltaContent != "" {
|
|
deltaReasoning, deltaContent = reasoning.ExtractReasoningComplete(deltaContent, thinkingStartToken, spokenReasoningConfig(config.ReasoningConfig))
|
|
}
|
|
reasoningText = deltaReasoning
|
|
responseWithoutReasoning = deltaContent
|
|
textContent = deltaContent
|
|
cleanedResponse = deltaContent
|
|
toolCalls = deltaToolCalls
|
|
} else {
|
|
reasoningText, responseWithoutReasoning = reasoning.ExtractReasoningComplete(rawResponse, thinkingStartToken, spokenReasoningConfig(config.ReasoningConfig))
|
|
textContent = functions.ParseTextContent(responseWithoutReasoning, config.FunctionsConfig)
|
|
cleanedResponse = functions.CleanupLLMResult(responseWithoutReasoning, config.FunctionsConfig)
|
|
toolCalls = functions.ParseFunctionCall(cleanedResponse, config.FunctionsConfig)
|
|
}
|
|
xlog.Debug("LLM Response", "reasoning", reasoningText, "response_without_reasoning", responseWithoutReasoning)
|
|
|
|
xlog.Debug("Function call parsing", "textContent", textContent, "cleanedResponse", cleanedResponse, "toolCallsCount", len(toolCalls))
|
|
|
|
noActionName := "answer"
|
|
if config.FunctionsConfig.NoActionFunctionName != "" {
|
|
noActionName = config.FunctionsConfig.NoActionFunctionName
|
|
}
|
|
isNoAction := len(toolCalls) > 0 && toolCalls[0].Name == noActionName
|
|
|
|
var finalSpeech string
|
|
var finalToolCalls []functions.FuncCallResults
|
|
|
|
if isNoAction {
|
|
arg := toolCalls[0].Arguments
|
|
arguments := map[string]any{}
|
|
if err := json.Unmarshal([]byte(arg), &arguments); err == nil {
|
|
if m, exists := arguments["message"]; exists {
|
|
if message, ok := m.(string); ok {
|
|
finalSpeech = message
|
|
} else {
|
|
xlog.Warn("NoAction function message field is not a string", "type", fmt.Sprintf("%T", m))
|
|
}
|
|
} else {
|
|
xlog.Warn("NoAction function missing 'message' field in arguments")
|
|
}
|
|
} else {
|
|
xlog.Warn("Failed to unmarshal NoAction function arguments", "error", err, "arguments", arg)
|
|
}
|
|
if finalSpeech == "" {
|
|
// Fallback if parsing failed
|
|
xlog.Warn("NoAction function did not produce speech, using cleaned response as fallback")
|
|
finalSpeech = cleanedResponse
|
|
}
|
|
} else {
|
|
finalToolCalls = toolCalls
|
|
xlog.Debug("Setting finalToolCalls", "count", len(finalToolCalls))
|
|
if len(toolCalls) > 0 {
|
|
finalSpeech = textContent
|
|
} else {
|
|
finalSpeech = cleanedResponse
|
|
}
|
|
}
|
|
|
|
if finalSpeech != "" {
|
|
// Create the assistant item now that we have content
|
|
item := types.MessageItemUnion{
|
|
Assistant: &types.MessageItemAssistant{
|
|
ID: generateItemID(),
|
|
Status: types.ItemStatusInProgress,
|
|
Content: []types.MessageContentOutput{
|
|
{
|
|
Type: types.MessageContentTypeOutputAudio,
|
|
Transcript: finalSpeech,
|
|
},
|
|
},
|
|
},
|
|
}
|
|
|
|
conv.Lock.Lock()
|
|
conv.Items = append(conv.Items, &item)
|
|
conv.Lock.Unlock()
|
|
|
|
sendEvent(t, types.ResponseOutputItemAddedEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
OutputIndex: 0,
|
|
Item: item,
|
|
})
|
|
|
|
sendEvent(t, types.ResponseContentPartAddedEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
ItemID: item.Assistant.ID,
|
|
OutputIndex: 0,
|
|
ContentIndex: 0,
|
|
Part: item.Assistant.Content[0],
|
|
})
|
|
|
|
// removeItemFromConv removes the last occurrence of an item with
|
|
// the given assistant ID from conversation history.
|
|
removeItemFromConv := func(assistantID string) {
|
|
conv.Lock.Lock()
|
|
for i := len(conv.Items) - 1; i >= 0; i-- {
|
|
if conv.Items[i].Assistant != nil && conv.Items[i].Assistant.ID == assistantID {
|
|
conv.Items = append(conv.Items[:i], conv.Items[i+1:]...)
|
|
break
|
|
}
|
|
}
|
|
conv.Lock.Unlock()
|
|
}
|
|
|
|
// sendCancelledResponse emits the cancelled status and cleans up the
|
|
// assistant item so the interrupted reply is not in chat history.
|
|
sendCancelledResponse := func() {
|
|
removeItemFromConv(item.Assistant.ID)
|
|
sendEvent(t, types.ResponseDoneEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
Response: types.Response{
|
|
ID: responseID,
|
|
Object: "realtime.response",
|
|
Status: types.ResponseStatusCancelled,
|
|
},
|
|
})
|
|
}
|
|
|
|
var audioString string
|
|
_, isWebRTC := t.(*WebRTCTransport)
|
|
var respMods []types.Modality
|
|
if overrides != nil {
|
|
respMods = overrides.OutputModalities
|
|
}
|
|
modalities := resolveOutputModalities(session.OutputModalities, respMods)
|
|
if modalitiesContainAudio(modalities) {
|
|
// Check for cancellation before TTS
|
|
if ctx.Err() != nil {
|
|
xlog.Debug("Response cancelled before TTS (barge-in)")
|
|
sendCancelledResponse()
|
|
return
|
|
}
|
|
|
|
// Transcript of the spoken reply (the audio's text).
|
|
sendEvent(t, types.ResponseOutputAudioTranscriptDeltaEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
ItemID: item.Assistant.ID,
|
|
OutputIndex: 0,
|
|
ContentIndex: 0,
|
|
Delta: finalSpeech,
|
|
})
|
|
sendEvent(t, types.ResponseOutputAudioTranscriptDoneEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
ItemID: item.Assistant.ID,
|
|
OutputIndex: 0,
|
|
ContentIndex: 0,
|
|
Transcript: finalSpeech,
|
|
})
|
|
|
|
// Synthesize and send the audio. With pipeline.streaming.tts enabled
|
|
// emitSpeech forwards a response.output_audio.delta per backend PCM
|
|
// chunk as it's produced; otherwise it sends the whole utterance as a
|
|
// single delta. The returned PCM is stored (base64) on the item below.
|
|
pcmAudio, err := emitSpeech(ctx, t, session, responseID, item.Assistant.ID, finalSpeech)
|
|
if err != nil {
|
|
if ctx.Err() != nil {
|
|
xlog.Debug("TTS cancelled (barge-in)")
|
|
sendCancelledResponse()
|
|
return
|
|
}
|
|
xlog.Error("TTS failed", "error", err)
|
|
sendError(t, "tts_error", fmt.Sprintf("TTS generation failed: %v", err), "", item.Assistant.ID)
|
|
return
|
|
}
|
|
if !isWebRTC {
|
|
audioString = base64.StdEncoding.EncodeToString(pcmAudio)
|
|
}
|
|
|
|
if !isWebRTC {
|
|
sendEvent(t, types.ResponseOutputAudioDoneEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
ItemID: item.Assistant.ID,
|
|
OutputIndex: 0,
|
|
ContentIndex: 0,
|
|
})
|
|
}
|
|
} else {
|
|
// Text-only mode: skip TTS, emit only the text events.
|
|
sendEvent(t, types.ResponseOutputTextDeltaEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
ItemID: item.Assistant.ID,
|
|
OutputIndex: 0,
|
|
ContentIndex: 0,
|
|
Delta: finalSpeech,
|
|
})
|
|
sendEvent(t, types.ResponseOutputTextDoneEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
ItemID: item.Assistant.ID,
|
|
OutputIndex: 0,
|
|
ContentIndex: 0,
|
|
Text: finalSpeech,
|
|
})
|
|
}
|
|
|
|
sendEvent(t, types.ResponseContentPartDoneEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
ItemID: item.Assistant.ID,
|
|
OutputIndex: 0,
|
|
ContentIndex: 0,
|
|
Part: item.Assistant.Content[0],
|
|
})
|
|
|
|
conv.Lock.Lock()
|
|
item.Assistant.Status = types.ItemStatusCompleted
|
|
if !isWebRTC {
|
|
item.Assistant.Content[0].Audio = audioString
|
|
}
|
|
conv.Lock.Unlock()
|
|
|
|
sendEvent(t, types.ResponseOutputItemDoneEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
OutputIndex: 0,
|
|
Item: item,
|
|
})
|
|
}
|
|
|
|
// Emit the parsed tool calls, the terminal response.done, and (for
|
|
// server-side assistant tools) the follow-up response. Shared with the
|
|
// streamed path so both finalize tool calls identically.
|
|
emitToolCallItems(ctx, session, conv, t, responseID, finalToolCalls, finalSpeech != "", toolTurn)
|
|
}
|
|
|
|
// emitToolCallItems emits the realtime function_call items for the parsed tool
|
|
// calls, the terminal response.done, and — for server-side LocalAI Assistant
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|
// tools — re-triggers a follow-up response so the model can speak the result.
|
|
// hasContent shifts the tool-call output index past the assistant content item
|
|
// when the same turn also produced spoken/text content. Two tool paths:
|
|
// - LocalAI Assistant tools (session.AssistantExecutor.IsTool) run server-side;
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|
// we append both the call and its output to conv.Items and re-trigger. The
|
|
// client only sees observability events.
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|
// - All other tools follow the standard OpenAI flow: emit
|
|
// function_call_arguments.done and wait for the client to send
|
|
// conversation.item.create back.
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|
func emitToolCallItems(ctx context.Context, session *Session, conv *Conversation, t Transport, responseID string, toolCalls []functions.FuncCallResults, hasContent bool, toolTurn int) {
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|
xlog.Debug("About to handle tool calls", "finalToolCallsCount", len(toolCalls))
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|
executedAssistantTool := false
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|
for i, tc := range toolCalls {
|
|
toolCallID := generateItemID()
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|
callID := "call_" + generateUniqueID() // OpenAI uses call_xyz
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|
|
|
// Create FunctionCall Item
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|
fcItem := types.MessageItemUnion{
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|
FunctionCall: &types.MessageItemFunctionCall{
|
|
ID: toolCallID,
|
|
CallID: callID,
|
|
Name: tc.Name,
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|
Arguments: tc.Arguments,
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|
Status: types.ItemStatusCompleted,
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|
},
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|
}
|
|
|
|
conv.Lock.Lock()
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|
conv.Items = append(conv.Items, &fcItem)
|
|
conv.Lock.Unlock()
|
|
|
|
outputIndex := i
|
|
if hasContent {
|
|
outputIndex++
|
|
}
|
|
|
|
sendEvent(t, types.ResponseOutputItemAddedEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
OutputIndex: outputIndex,
|
|
Item: fcItem,
|
|
})
|
|
|
|
serverSide := session.AssistantExecutor != nil && session.AssistantExecutor.IsTool(tc.Name)
|
|
if serverSide {
|
|
output, execErr := session.AssistantExecutor.ExecuteTool(ctx, tc.Name, tc.Arguments)
|
|
if execErr != nil {
|
|
output = "Error: " + execErr.Error()
|
|
xlog.Error("realtime: assistant tool execution failed", "tool", tc.Name, "error", execErr)
|
|
}
|
|
foItem := types.MessageItemUnion{
|
|
FunctionCallOutput: &types.MessageItemFunctionCallOutput{
|
|
ID: generateItemID(),
|
|
CallID: callID,
|
|
Output: output,
|
|
Status: types.ItemStatusCompleted,
|
|
},
|
|
}
|
|
conv.Lock.Lock()
|
|
conv.Items = append(conv.Items, &foItem)
|
|
conv.Lock.Unlock()
|
|
// Close the call out and emit the output as its own paired
|
|
// added/done — the OpenAI spec pairs every item-done with a
|
|
// preceding item-added, so we re-pair here for the output.
|
|
// The UI renders the transcript entry on item.done for both
|
|
// shapes (FunctionCall + FunctionCallOutput).
|
|
sendEvent(t, types.ResponseOutputItemDoneEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
OutputIndex: outputIndex,
|
|
Item: fcItem,
|
|
})
|
|
sendEvent(t, types.ResponseOutputItemAddedEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
OutputIndex: outputIndex,
|
|
Item: foItem,
|
|
})
|
|
sendEvent(t, types.ResponseOutputItemDoneEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
OutputIndex: outputIndex,
|
|
Item: foItem,
|
|
})
|
|
executedAssistantTool = true
|
|
continue
|
|
}
|
|
|
|
sendEvent(t, types.ResponseFunctionCallArgumentsDeltaEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
ItemID: toolCallID,
|
|
OutputIndex: outputIndex,
|
|
CallID: callID,
|
|
Delta: tc.Arguments,
|
|
})
|
|
|
|
sendEvent(t, types.ResponseFunctionCallArgumentsDoneEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
ItemID: toolCallID,
|
|
OutputIndex: outputIndex,
|
|
CallID: callID,
|
|
Arguments: tc.Arguments,
|
|
Name: tc.Name,
|
|
})
|
|
|
|
sendEvent(t, types.ResponseOutputItemDoneEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
ResponseID: responseID,
|
|
OutputIndex: outputIndex,
|
|
Item: fcItem,
|
|
})
|
|
}
|
|
|
|
sendEvent(t, types.ResponseDoneEvent{
|
|
ServerEventBase: types.ServerEventBase{},
|
|
Response: types.Response{
|
|
ID: responseID,
|
|
Object: "realtime.response",
|
|
Status: types.ResponseStatusCompleted,
|
|
},
|
|
})
|
|
|
|
// If we executed any assistant tools inproc, run another response cycle
|
|
// so the model can speak the result. Mirrors the chat-side agentic loop
|
|
// but driven server-side rather than by client round-trip. Bounded so a
|
|
// degenerate "model keeps calling tools" doesn't blow the stack.
|
|
if executedAssistantTool {
|
|
if toolTurn+1 >= maxAssistantToolTurns {
|
|
xlog.Warn("realtime: assistant tool-turn limit reached, stopping the agentic loop",
|
|
"limit", maxAssistantToolTurns, "model", session.Model)
|
|
return
|
|
}
|
|
triggerResponseAtTurn(ctx, session, conv, t, nil, toolTurn+1)
|
|
}
|
|
}
|
|
|
|
// Helper functions to generate unique IDs
|
|
func generateSessionID() string {
|
|
// Generate a unique session ID
|
|
// Implement as needed
|
|
return "sess_" + generateUniqueID()
|
|
}
|
|
|
|
func generateConversationID() string {
|
|
// Generate a unique conversation ID
|
|
// Implement as needed
|
|
return "conv_" + generateUniqueID()
|
|
}
|
|
|
|
func generateItemID() string {
|
|
// Generate a unique item ID
|
|
// Implement as needed
|
|
return "item_" + generateUniqueID()
|
|
}
|
|
|
|
func generateUniqueID() string {
|
|
// 16 random bytes, hex-encoded. Must be collision-free: session, item,
|
|
// response and call IDs build on this, and the conversation tracks/removes
|
|
// items by ID (e.g. cancel() in realtime_stream.go, conversation.item.retrieve).
|
|
// A constant would make every ID alias and corrupt that bookkeeping.
|
|
var b [16]byte
|
|
_, _ = rand.Read(b[:])
|
|
return hex.EncodeToString(b[:])
|
|
}
|