LocalAI/core/backend/audio_transform.go

177 lines
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feat: add LocalVQE backend and audio transformations UI (#9640) feat(audio-transform): add LocalVQE backend, bidi gRPC RPC, Studio UI Introduce a generic "audio transform" capability for any audio-in / audio-out operation (echo cancellation, noise suppression, dereverberation, voice conversion, etc.) and ship LocalVQE as the first backend implementation. Backend protocol: - Two new gRPC RPCs in backend.proto: unary AudioTransform for batch and bidirectional AudioTransformStream for low-latency frame-by-frame use. This is the first bidi stream in the proto; per-frame unary at LocalVQE's 16 ms hop would be RTT-bound. Wire it through pkg/grpc/{client,server, embed,interface,base} with paired-channel ergonomics. LocalVQE backend (backend/go/localvqe/): - Go-Purego wrapper around upstream liblocalvqe.so. CMake builds the upstream shared lib + its libggml-cpu-*.so runtime variants directly — no MODULE wrapper needed because LocalVQE handles CPU feature selection internally via GGML_BACKEND_DL. - Sets GGML_NTHREADS from opts.Threads (or runtime.NumCPU()-1) — without it LocalVQE runs single-threaded at ~1× realtime instead of the documented ~9.6×. - Reference-length policy: zero-pad short refs, truncate long ones (the trailing portion can't have leaked into a mic that wasn't recording). - Ginkgo test suite (9 always-on specs + 2 model-gated). HTTP layer: - POST /audio/transformations (alias /audio/transform): multipart batch endpoint, accepts audio + optional reference + params[*]=v form fields. Persists inputs alongside the output in GeneratedContentDir/audio so the React UI history can replay past (audio, reference, output) triples. - GET /audio/transformations/stream: WebSocket bidi, 16 ms PCM frames (interleaved stereo mic+ref in, mono out). JSON session.update envelope for config; constants hoisted in core/schema/audio_transform.go. - ffmpeg-based input normalisation to 16 kHz mono s16 WAV via the existing utils.AudioToWav (with passthrough fast-path), so the user can upload any format / rate without seeing the model's strict 16 kHz constraint. - BackendTraceAudioTransform integration so /api/backend-traces and the Traces UI light up with audio_snippet base64 and timing. - Routes registered under routes/localai.go (LocalAI extension; OpenAI has no /audio/transformations endpoint), traced via TraceMiddleware. Auth + capability + importer: - FLAG_AUDIO_TRANSFORM (model_config.go), FeatureAudioTransform (default-on, in APIFeatures), three RouteFeatureRegistry rows. - localvqe added to knownPrefOnlyBackends with modality "audio-transform". - Gallery entry localvqe-v1-1.3m (sha256-pinned, hosted on huggingface.co/LocalAI-io/LocalVQE). React UI: - New /app/transform page surfaced via a dedicated "Enhance" sidebar section (sibling of Tools / Biometrics) — the page is enhancement, not generation, so it lives outside Studio. Two AudioInput components (Upload + Record tabs, drag-drop, mic capture). - Echo-test button: records mic while playing the loaded reference through the speakers — the mic naturally picks up speaker bleed, giving a real (mic, ref) pair for AEC testing without leaving the UI. - Reusable WaveformPlayer (canvas peaks + click-to-seek + audio controls) and useAudioPeaks hook (shared module-scoped AudioContext to avoid hitting browser context limits with three players on one page); migrated TTS, Sound, Traces audio blocks to use it. - Past runs saved in localStorage via useMediaHistory('audio-transform') — the history entry stores all three URLs so clicking re-renders the full triple, not just the output. Build + e2e: - 11 matrix entries removed from .github/workflows/backend.yml (CUDA, ROCm, SYCL, Metal, L4T): upstream supports only CPU + Vulkan, so we ship those two and let GPU-class hardware route through Vulkan in the gallery capabilities map. - tests-localvqe-grpc-transform job in test-extra.yml (gated on detect-changes.outputs.localvqe). - New audio_transform capability + 4 specs in tests/e2e-backends. - Playwright spec suite in core/http/react-ui/e2e/audio-transform.spec.js (8 specs covering tabs, file upload, multipart shape, history, errors). Docs: - New docs/content/features/audio-transform.md covering the (audio, reference) mental model, batch + WebSocket wire formats, LocalVQE param keys, and a YAML config example. Cross-links from text-to-audio and audio-to-text feature pages. Assisted-by: Claude:claude-opus-4-7 [Bash Read Edit Write Agent TaskCreate] Signed-off-by: Richard Palethorpe <io@richiejp.com>
2026-05-04 20:07:11 +00:00
package backend
import (
"context"
"fmt"
"io"
"maps"
"os"
"path/filepath"
"time"
"github.com/mudler/LocalAI/core/config"
"github.com/mudler/LocalAI/core/trace"
"github.com/mudler/LocalAI/pkg/grpc"
"github.com/mudler/LocalAI/pkg/grpc/proto"
"github.com/mudler/LocalAI/pkg/model"
"github.com/mudler/LocalAI/pkg/utils"
)
// AudioTransformOptions carries per-request tuning for the unary transform.
type AudioTransformOptions struct {
// Params is forwarded verbatim to the backend (e.g. LocalVQE reads
// params["noise_gate"] / params["noise_gate_threshold_dbfs"]).
Params map[string]string
}
// AudioTransformOutputs are the on-disk paths of the persisted artifacts —
// the user-visible Dst plus copies of the inputs the backend actually saw.
// Inputs are persisted because the React UI history needs to display past
// runs, and rejecting them once the temp dir is cleaned up would defeat
// the point.
type AudioTransformOutputs struct {
Dst string
AudioPath string
ReferencePath string
}
// ModelAudioTransform runs the unary AudioTransform RPC and returns the
// generated output path plus the persisted input paths. `audioPath` is
// required; `referencePath` is optional (empty => backend zero-fills the
// reference channel).
func ModelAudioTransform(
feat(whisper): honor client cancellation via ggml abort_callback (#9710) * refactor(transcription): propagate request ctx through ModelTranscription* Replaces context.Background() with the HTTP request ctx so client disconnects start cancelling the gRPC call. No backend-side abort wiring yet — that comes in a later commit. Pure plumbing. Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(cli): pass ctx to backend.ModelTranscription Follow-up to e65d3e1f which threaded ctx through ModelTranscription but missed the CLI caller. CLI commands have no request-scoped ctx, so context.Background() is correct here. Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * refactor(audio): propagate request ctx into TTS, sound-gen, audio-transform Same ctx-plumbing pattern applied to the rest of the audio path. CLI callers use context.Background() since there is no request scope; HTTP callers use c.Request().Context(). Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * refactor(backend): propagate request ctx into biometric, detection, rerank, diarization paths Replaces remaining context.Background() sites in core/backend with the caller's ctx. After this commit, every core/backend/*.go entry point threads the request ctx end-to-end to the gRPC client. Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * refactor(grpc): plumb ctx through AIModel.AudioTranscription{,Stream} Adds context.Context as first parameter to the AIModel interface methods that wrap whisper-style transcription. Server-side gRPC handler now forwards the per-RPC ctx (server-streaming uses stream.Context()). Whisper, Voxtral, vibevoice-cpp, and sherpa-onnx accept the parameter; none uses it yet — the actual cancellation primitive lands in the next commit so this is pure plumbing. Assisted-by: Claude:claude-sonnet-4-6 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * feat(whisper): add abort_callback hook in the C++ bridge Installs a std::atomic<int> flag, wires it into whisper_full_params.abort_callback, and exposes a set_abort(int) C symbol so Go can flip the flag from a goroutine watching the request context. transcribe() now distinguishes abort (return 2) from real whisper_full failure (return 1). Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * feat(whisper): register set_abort symbol in the purego loader Adds the Go-side binding for the new C export so the next commit can call CppSetAbort(1) from a watcher goroutine on ctx.Done(). Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * feat(whisper): honor ctx cancellation and return codes.Canceled A watcher goroutine watches ctx.Done() during AudioTranscription and calls CppSetAbort(1) on cancel. whisper_full sees abort_callback return true at the next compute graph step, returns non-zero, and the bridge returns 2 -> AudioTranscription maps that to codes.Canceled. Adds an opt-in test (gated on WHISPER_MODEL_PATH / WHISPER_AUDIO_PATH) that asserts cancellation latency under 5s and proves the abort flag resets cleanly so the next transcription succeeds. Assisted-by: Claude:claude-sonnet-4-6 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(whisper): join the cancel watcher goroutine before returning Follow-up to 85edf9d2. The previous commit used `defer close(done)` and called the watcher "joined synchronously" — but close() only signals, it does not block until the goroutine exits. That left a window where a late CppSetAbort(1) from a cancelled call could land on the next call, after its C-side g_abort reset but before whisper_full() began polling the abort callback, corrupting the second transcription. Switch to a sync.WaitGroup join so wg.Wait() blocks until the watcher has actually returned from its select. Assisted-by: Claude:claude-sonnet-4-6 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(whisper): short-circuit pre-cancelled ctx in AudioTranscription If ctx is already Done() at entry, return codes.Canceled immediately instead of running the full transcription. The C-side g_abort reset happens at the start of transcribe() and would otherwise overwrite a watcher-set abort flag from an already-cancelled ctx, producing a spurious successful transcription on a request the client has already abandoned. Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(tests/distributed): update testLLM mock for new AudioTranscription signature Phase B (93c48e19) added context.Context to AIModel.AudioTranscription but missed the testLLM mock in tests/e2e/distributed. CI golangci-lint caught it: *testLLM did not implement grpc.AIModel because the method signature lacked the ctx parameter, which broke the distributed test suite compilation and cascaded through every backend-build job that runs `go build ./...`. Assisted-by: Claude:claude-opus-4-7 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * test(whisper): port cancellation test to Ginkgo/Gomega Project policy (.agents/coding-style.md, enforced by golangci-lint forbidigo) is that all Go tests must use Ginkgo v2 + Gomega — no stdlib testing patterns (t.Skip, t.Fatalf, etc.). Convert the cancellation test to a Describe/It block with Skip(...) for env gating and Expect/HaveOccurred for assertions. Same coverage: cancel mid-flight returns codes.Canceled within 5s and a follow-up transcription succeeds, proving the C-side g_abort flag resets cleanly. Assisted-by: Claude:claude-opus-4-7 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> --------- Signed-off-by: Ettore Di Giacinto <mudler@localai.io> Co-authored-by: Ettore Di Giacinto <mudler@localai.io>
2026-05-07 23:44:47 +00:00
ctx context.Context,
feat: add LocalVQE backend and audio transformations UI (#9640) feat(audio-transform): add LocalVQE backend, bidi gRPC RPC, Studio UI Introduce a generic "audio transform" capability for any audio-in / audio-out operation (echo cancellation, noise suppression, dereverberation, voice conversion, etc.) and ship LocalVQE as the first backend implementation. Backend protocol: - Two new gRPC RPCs in backend.proto: unary AudioTransform for batch and bidirectional AudioTransformStream for low-latency frame-by-frame use. This is the first bidi stream in the proto; per-frame unary at LocalVQE's 16 ms hop would be RTT-bound. Wire it through pkg/grpc/{client,server, embed,interface,base} with paired-channel ergonomics. LocalVQE backend (backend/go/localvqe/): - Go-Purego wrapper around upstream liblocalvqe.so. CMake builds the upstream shared lib + its libggml-cpu-*.so runtime variants directly — no MODULE wrapper needed because LocalVQE handles CPU feature selection internally via GGML_BACKEND_DL. - Sets GGML_NTHREADS from opts.Threads (or runtime.NumCPU()-1) — without it LocalVQE runs single-threaded at ~1× realtime instead of the documented ~9.6×. - Reference-length policy: zero-pad short refs, truncate long ones (the trailing portion can't have leaked into a mic that wasn't recording). - Ginkgo test suite (9 always-on specs + 2 model-gated). HTTP layer: - POST /audio/transformations (alias /audio/transform): multipart batch endpoint, accepts audio + optional reference + params[*]=v form fields. Persists inputs alongside the output in GeneratedContentDir/audio so the React UI history can replay past (audio, reference, output) triples. - GET /audio/transformations/stream: WebSocket bidi, 16 ms PCM frames (interleaved stereo mic+ref in, mono out). JSON session.update envelope for config; constants hoisted in core/schema/audio_transform.go. - ffmpeg-based input normalisation to 16 kHz mono s16 WAV via the existing utils.AudioToWav (with passthrough fast-path), so the user can upload any format / rate without seeing the model's strict 16 kHz constraint. - BackendTraceAudioTransform integration so /api/backend-traces and the Traces UI light up with audio_snippet base64 and timing. - Routes registered under routes/localai.go (LocalAI extension; OpenAI has no /audio/transformations endpoint), traced via TraceMiddleware. Auth + capability + importer: - FLAG_AUDIO_TRANSFORM (model_config.go), FeatureAudioTransform (default-on, in APIFeatures), three RouteFeatureRegistry rows. - localvqe added to knownPrefOnlyBackends with modality "audio-transform". - Gallery entry localvqe-v1-1.3m (sha256-pinned, hosted on huggingface.co/LocalAI-io/LocalVQE). React UI: - New /app/transform page surfaced via a dedicated "Enhance" sidebar section (sibling of Tools / Biometrics) — the page is enhancement, not generation, so it lives outside Studio. Two AudioInput components (Upload + Record tabs, drag-drop, mic capture). - Echo-test button: records mic while playing the loaded reference through the speakers — the mic naturally picks up speaker bleed, giving a real (mic, ref) pair for AEC testing without leaving the UI. - Reusable WaveformPlayer (canvas peaks + click-to-seek + audio controls) and useAudioPeaks hook (shared module-scoped AudioContext to avoid hitting browser context limits with three players on one page); migrated TTS, Sound, Traces audio blocks to use it. - Past runs saved in localStorage via useMediaHistory('audio-transform') — the history entry stores all three URLs so clicking re-renders the full triple, not just the output. Build + e2e: - 11 matrix entries removed from .github/workflows/backend.yml (CUDA, ROCm, SYCL, Metal, L4T): upstream supports only CPU + Vulkan, so we ship those two and let GPU-class hardware route through Vulkan in the gallery capabilities map. - tests-localvqe-grpc-transform job in test-extra.yml (gated on detect-changes.outputs.localvqe). - New audio_transform capability + 4 specs in tests/e2e-backends. - Playwright spec suite in core/http/react-ui/e2e/audio-transform.spec.js (8 specs covering tabs, file upload, multipart shape, history, errors). Docs: - New docs/content/features/audio-transform.md covering the (audio, reference) mental model, batch + WebSocket wire formats, LocalVQE param keys, and a YAML config example. Cross-links from text-to-audio and audio-to-text feature pages. Assisted-by: Claude:claude-opus-4-7 [Bash Read Edit Write Agent TaskCreate] Signed-off-by: Richard Palethorpe <io@richiejp.com>
2026-05-04 20:07:11 +00:00
audioPath, referencePath string,
opts AudioTransformOptions,
loader *model.ModelLoader,
appConfig *config.ApplicationConfig,
modelConfig config.ModelConfig,
) (AudioTransformOutputs, *proto.AudioTransformResult, error) {
mopts := ModelOptions(modelConfig, appConfig)
transformModel, err := loader.Load(mopts...)
if err != nil {
recordModelLoadFailure(appConfig, modelConfig.Name, modelConfig.Backend, err, nil)
return AudioTransformOutputs{}, nil, err
}
if transformModel == nil {
return AudioTransformOutputs{}, nil, fmt.Errorf("could not load audio-transform model %q", modelConfig.Model)
}
audioDir := filepath.Join(appConfig.GeneratedContentDir, "audio")
if err := os.MkdirAll(audioDir, 0750); err != nil {
return AudioTransformOutputs{}, nil, fmt.Errorf("failed creating audio directory: %s", err)
}
dst := filepath.Join(audioDir, utils.GenerateUniqueFileName(audioDir, "transform", ".wav"))
persistedAudio, err := persistAudioInput(audioPath, audioDir, "transform-input", ".wav")
if err != nil {
return AudioTransformOutputs{}, nil, fmt.Errorf("persist input audio: %w", err)
}
persistedRef := ""
if referencePath != "" {
persistedRef, err = persistAudioInput(referencePath, audioDir, "transform-ref", ".wav")
if err != nil {
return AudioTransformOutputs{}, nil, fmt.Errorf("persist reference: %w", err)
}
}
var startTime time.Time
if appConfig.EnableTracing {
trace.InitBackendTracingIfEnabled(appConfig.TracingMaxItems)
startTime = time.Now()
}
feat(whisper): honor client cancellation via ggml abort_callback (#9710) * refactor(transcription): propagate request ctx through ModelTranscription* Replaces context.Background() with the HTTP request ctx so client disconnects start cancelling the gRPC call. No backend-side abort wiring yet — that comes in a later commit. Pure plumbing. Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(cli): pass ctx to backend.ModelTranscription Follow-up to e65d3e1f which threaded ctx through ModelTranscription but missed the CLI caller. CLI commands have no request-scoped ctx, so context.Background() is correct here. Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * refactor(audio): propagate request ctx into TTS, sound-gen, audio-transform Same ctx-plumbing pattern applied to the rest of the audio path. CLI callers use context.Background() since there is no request scope; HTTP callers use c.Request().Context(). Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * refactor(backend): propagate request ctx into biometric, detection, rerank, diarization paths Replaces remaining context.Background() sites in core/backend with the caller's ctx. After this commit, every core/backend/*.go entry point threads the request ctx end-to-end to the gRPC client. Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * refactor(grpc): plumb ctx through AIModel.AudioTranscription{,Stream} Adds context.Context as first parameter to the AIModel interface methods that wrap whisper-style transcription. Server-side gRPC handler now forwards the per-RPC ctx (server-streaming uses stream.Context()). Whisper, Voxtral, vibevoice-cpp, and sherpa-onnx accept the parameter; none uses it yet — the actual cancellation primitive lands in the next commit so this is pure plumbing. Assisted-by: Claude:claude-sonnet-4-6 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * feat(whisper): add abort_callback hook in the C++ bridge Installs a std::atomic<int> flag, wires it into whisper_full_params.abort_callback, and exposes a set_abort(int) C symbol so Go can flip the flag from a goroutine watching the request context. transcribe() now distinguishes abort (return 2) from real whisper_full failure (return 1). Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * feat(whisper): register set_abort symbol in the purego loader Adds the Go-side binding for the new C export so the next commit can call CppSetAbort(1) from a watcher goroutine on ctx.Done(). Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * feat(whisper): honor ctx cancellation and return codes.Canceled A watcher goroutine watches ctx.Done() during AudioTranscription and calls CppSetAbort(1) on cancel. whisper_full sees abort_callback return true at the next compute graph step, returns non-zero, and the bridge returns 2 -> AudioTranscription maps that to codes.Canceled. Adds an opt-in test (gated on WHISPER_MODEL_PATH / WHISPER_AUDIO_PATH) that asserts cancellation latency under 5s and proves the abort flag resets cleanly so the next transcription succeeds. Assisted-by: Claude:claude-sonnet-4-6 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(whisper): join the cancel watcher goroutine before returning Follow-up to 85edf9d2. The previous commit used `defer close(done)` and called the watcher "joined synchronously" — but close() only signals, it does not block until the goroutine exits. That left a window where a late CppSetAbort(1) from a cancelled call could land on the next call, after its C-side g_abort reset but before whisper_full() began polling the abort callback, corrupting the second transcription. Switch to a sync.WaitGroup join so wg.Wait() blocks until the watcher has actually returned from its select. Assisted-by: Claude:claude-sonnet-4-6 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(whisper): short-circuit pre-cancelled ctx in AudioTranscription If ctx is already Done() at entry, return codes.Canceled immediately instead of running the full transcription. The C-side g_abort reset happens at the start of transcribe() and would otherwise overwrite a watcher-set abort flag from an already-cancelled ctx, producing a spurious successful transcription on a request the client has already abandoned. Assisted-by: Claude:claude-haiku-4-5 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(tests/distributed): update testLLM mock for new AudioTranscription signature Phase B (93c48e19) added context.Context to AIModel.AudioTranscription but missed the testLLM mock in tests/e2e/distributed. CI golangci-lint caught it: *testLLM did not implement grpc.AIModel because the method signature lacked the ctx parameter, which broke the distributed test suite compilation and cascaded through every backend-build job that runs `go build ./...`. Assisted-by: Claude:claude-opus-4-7 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * test(whisper): port cancellation test to Ginkgo/Gomega Project policy (.agents/coding-style.md, enforced by golangci-lint forbidigo) is that all Go tests must use Ginkgo v2 + Gomega — no stdlib testing patterns (t.Skip, t.Fatalf, etc.). Convert the cancellation test to a Describe/It block with Skip(...) for env gating and Expect/HaveOccurred for assertions. Same coverage: cancel mid-flight returns codes.Canceled within 5s and a follow-up transcription succeeds, proving the C-side g_abort flag resets cleanly. Assisted-by: Claude:claude-opus-4-7 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> --------- Signed-off-by: Ettore Di Giacinto <mudler@localai.io> Co-authored-by: Ettore Di Giacinto <mudler@localai.io>
2026-05-07 23:44:47 +00:00
res, err := transformModel.AudioTransform(ctx, &proto.AudioTransformRequest{
feat: add LocalVQE backend and audio transformations UI (#9640) feat(audio-transform): add LocalVQE backend, bidi gRPC RPC, Studio UI Introduce a generic "audio transform" capability for any audio-in / audio-out operation (echo cancellation, noise suppression, dereverberation, voice conversion, etc.) and ship LocalVQE as the first backend implementation. Backend protocol: - Two new gRPC RPCs in backend.proto: unary AudioTransform for batch and bidirectional AudioTransformStream for low-latency frame-by-frame use. This is the first bidi stream in the proto; per-frame unary at LocalVQE's 16 ms hop would be RTT-bound. Wire it through pkg/grpc/{client,server, embed,interface,base} with paired-channel ergonomics. LocalVQE backend (backend/go/localvqe/): - Go-Purego wrapper around upstream liblocalvqe.so. CMake builds the upstream shared lib + its libggml-cpu-*.so runtime variants directly — no MODULE wrapper needed because LocalVQE handles CPU feature selection internally via GGML_BACKEND_DL. - Sets GGML_NTHREADS from opts.Threads (or runtime.NumCPU()-1) — without it LocalVQE runs single-threaded at ~1× realtime instead of the documented ~9.6×. - Reference-length policy: zero-pad short refs, truncate long ones (the trailing portion can't have leaked into a mic that wasn't recording). - Ginkgo test suite (9 always-on specs + 2 model-gated). HTTP layer: - POST /audio/transformations (alias /audio/transform): multipart batch endpoint, accepts audio + optional reference + params[*]=v form fields. Persists inputs alongside the output in GeneratedContentDir/audio so the React UI history can replay past (audio, reference, output) triples. - GET /audio/transformations/stream: WebSocket bidi, 16 ms PCM frames (interleaved stereo mic+ref in, mono out). JSON session.update envelope for config; constants hoisted in core/schema/audio_transform.go. - ffmpeg-based input normalisation to 16 kHz mono s16 WAV via the existing utils.AudioToWav (with passthrough fast-path), so the user can upload any format / rate without seeing the model's strict 16 kHz constraint. - BackendTraceAudioTransform integration so /api/backend-traces and the Traces UI light up with audio_snippet base64 and timing. - Routes registered under routes/localai.go (LocalAI extension; OpenAI has no /audio/transformations endpoint), traced via TraceMiddleware. Auth + capability + importer: - FLAG_AUDIO_TRANSFORM (model_config.go), FeatureAudioTransform (default-on, in APIFeatures), three RouteFeatureRegistry rows. - localvqe added to knownPrefOnlyBackends with modality "audio-transform". - Gallery entry localvqe-v1-1.3m (sha256-pinned, hosted on huggingface.co/LocalAI-io/LocalVQE). React UI: - New /app/transform page surfaced via a dedicated "Enhance" sidebar section (sibling of Tools / Biometrics) — the page is enhancement, not generation, so it lives outside Studio. Two AudioInput components (Upload + Record tabs, drag-drop, mic capture). - Echo-test button: records mic while playing the loaded reference through the speakers — the mic naturally picks up speaker bleed, giving a real (mic, ref) pair for AEC testing without leaving the UI. - Reusable WaveformPlayer (canvas peaks + click-to-seek + audio controls) and useAudioPeaks hook (shared module-scoped AudioContext to avoid hitting browser context limits with three players on one page); migrated TTS, Sound, Traces audio blocks to use it. - Past runs saved in localStorage via useMediaHistory('audio-transform') — the history entry stores all three URLs so clicking re-renders the full triple, not just the output. Build + e2e: - 11 matrix entries removed from .github/workflows/backend.yml (CUDA, ROCm, SYCL, Metal, L4T): upstream supports only CPU + Vulkan, so we ship those two and let GPU-class hardware route through Vulkan in the gallery capabilities map. - tests-localvqe-grpc-transform job in test-extra.yml (gated on detect-changes.outputs.localvqe). - New audio_transform capability + 4 specs in tests/e2e-backends. - Playwright spec suite in core/http/react-ui/e2e/audio-transform.spec.js (8 specs covering tabs, file upload, multipart shape, history, errors). Docs: - New docs/content/features/audio-transform.md covering the (audio, reference) mental model, batch + WebSocket wire formats, LocalVQE param keys, and a YAML config example. Cross-links from text-to-audio and audio-to-text feature pages. Assisted-by: Claude:claude-opus-4-7 [Bash Read Edit Write Agent TaskCreate] Signed-off-by: Richard Palethorpe <io@richiejp.com>
2026-05-04 20:07:11 +00:00
AudioPath: audioPath,
ReferencePath: referencePath,
Dst: dst,
Params: opts.Params,
})
if appConfig.EnableTracing {
errStr := ""
if err != nil {
errStr = err.Error()
}
data := map[string]any{
"audio_path": audioPath,
"reference_path": referencePath,
"dst": dst,
"params": opts.Params,
}
if err == nil && res != nil {
data["sample_rate"] = res.SampleRate
data["samples"] = res.Samples
data["reference_provided"] = res.ReferenceProvided
if snippet := trace.AudioSnippet(dst); snippet != nil {
maps.Copy(data, snippet)
}
}
trace.RecordBackendTrace(trace.BackendTrace{
Timestamp: startTime,
Duration: time.Since(startTime),
Type: trace.BackendTraceAudioTransform,
ModelName: modelConfig.Name,
Backend: modelConfig.Backend,
Summary: trace.TruncateString(filepath.Base(audioPath), 200),
Error: errStr,
Data: data,
})
}
if err != nil {
return AudioTransformOutputs{}, nil, err
}
return AudioTransformOutputs{
Dst: dst,
AudioPath: persistedAudio,
ReferencePath: persistedRef,
}, res, nil
}
// ModelAudioTransformStream opens the bidirectional AudioTransformStream RPC
// and returns the underlying stream client. The caller is responsible for
// sending the initial Config message, subsequent Frame messages, and for
// calling CloseSend when input is done. The returned stream's Recv reports
// EOF when the backend has finished emitting frames.
func ModelAudioTransformStream(
ctx context.Context,
loader *model.ModelLoader,
appConfig *config.ApplicationConfig,
modelConfig config.ModelConfig,
) (grpc.AudioTransformStreamClient, error) {
mopts := ModelOptions(modelConfig, appConfig)
transformModel, err := loader.Load(mopts...)
if err != nil {
recordModelLoadFailure(appConfig, modelConfig.Name, modelConfig.Backend, err, nil)
return nil, err
}
if transformModel == nil {
return nil, fmt.Errorf("could not load audio-transform model %q", modelConfig.Model)
}
return transformModel.AudioTransformStream(ctx)
}
// persistAudioInput copies a transient input file (typically a multipart
// upload that lives in an os.TempDir slated for cleanup) into the long-lived
// GeneratedContentDir under a unique name, so the React UI can replay it
// from history.
func persistAudioInput(srcPath, dir, prefix, ext string) (string, error) {
src, err := os.Open(srcPath)
if err != nil {
return "", err
}
defer func() { _ = src.Close() }()
dst := filepath.Join(dir, utils.GenerateUniqueFileName(dir, prefix, ext))
out, err := os.Create(dst)
if err != nil {
return "", err
}
defer func() { _ = out.Close() }()
if _, err := io.Copy(out, src); err != nil {
return "", err
}
return dst, nil
}