backend(audio-cpp): bound the lane a live stream can hold

AudioTranscriptionLive holds the model's inference lane for the whole stream,
which is correct (the streaming session is stateful and a concurrent run would
interleave two callers' audio) and newly dangerous. Every other RPC holds the
lane across compute, or across a write to a slow reader, and both of those
terminate on their own. A live stream instead blocks in a client-driven read,
and a peer that goes silent WITHOUT closing the stream never terminates
anything: the lane stays taken and every other request against that model queues
behind a client that stopped speaking.

live_watchdog is a one-shot idle timer that ends the stream when no frame has
arrived inside a window. It is standard library only, so it is unit tested
without an engine. gRPC's synchronous Read has no timeout and cannot be given
one, so the only way to unblock it is ServerContext::TryCancel, which decides
the wire status itself: the client sees CANCELLED rather than the
DEADLINE_EXCEEDED the handler returns, the reason is logged, and the lane coming
back is the point. When it fires the read loop throws rather than reporting
end-of-input, so the driver does not go on to finalize a decode nobody is
waiting for.

It is armed only after the lane is taken and disarmed as soon as the read side
closes, and both ends matter. Arming earlier would cover acquire(), which
legitimately blocks while another live stream runs, so a queued caller would be
cancelled for waiting its turn. Disarming later would cover our own decode,
where a window overrun is not a peer going quiet and cancelling would throw away
the transcript the client is waiting for.

The window is the new live_idle_timeout_ms option, 30 s by default, 0 meaning no
limit. core/http/endpoints/openai/realtime.go drives a 300 ms ticker and feeds
every tick that produced new audio while a turn is open, so 30 s of silence is a
hundred ticks that delivered nothing. It is also longer than any pause a speaker
takes mid-utterance, which is the case that must never be cut off, and
backend.proto lets one stream span many utterances, so a client that pauses
longer between them raises the option rather than discovering it.

Two smaller corrections in the same handler:

  - check_can_serve now runs BEFORE the sample rate check.
    pkg/grpc/grpcerrors/errors.go degrades to the file path on UNIMPLEMENTED and
    on nothing else, so a live-incapable model asked at a wrong rate was
    answering INVALID_ARGUMENT and costing the caller its fallback.
  - a negative sample rate is refused instead of silently becoming 16000. Zero
    still means 16000, which is what the proto documents; -1 is malformed rather
    than absent and gets the same refusal every other bad rate gets.

And one thing recorded rather than changed, at the handler: "live" here means
incremental INPUT, not low latency, and with the pinned families it does not yet
mean incremental OUTPUT either. nemotron_asr's process_audio_chunk only appends
to its buffer, so its whole decode and every delta happen inside finalize(),
after the client closes its send side. The policy-window buffering is inert for
that family and matters only for vibevoice_asr and higgs_audio_stt.

Verified on the wire with live_idle_timeout_ms:3000. A silent client acked at
371 ms and was cancelled at 3.371 s; a second live stream opened one second
later received its ack 2.37 s in, i.e. at the instant the first was cancelled,
and then transcribed successfully on the same cached session. Without the
watchdog it would still be waiting. Re-ran the live transcription (ready first,
59 incremental deltas, concat equal to the final text, word timestamps in
nanoseconds, eou and eob false), the citrinet refusal at both a right and a
wrong rate (UNIMPLEMENTED either way now), and Task 12's AudioTranscriptionStream
on nemotron_asr, which is unchanged.

Mutation testing the watchdog found a weakness in its own test: the destructor
test slept past the window inside the watched scope, so a destructor that
DETACHED the thread instead of joining it passed unnoticed. The test now uses a
window longer than the scope, which kills that mutant, and says why.

Assisted-by: Claude:claude-opus-5 [Claude Code]
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

backend/cpp/audio-cpp/grpc-server.cpp

@@ -17,6 +17,7 @@
 #include "capability_routing.h"
 #include "generation_request.h"
 #include "inference_lane.h"
+#include "live_watchdog.h"
 #include "loaded_model.h"
 #include "model_options.h"
 #include "result_map.h"
@@ -466,6 +467,18 @@ void refuse_cloning_without_a_clip(const audiocpp_backend::LoadedModel &model,
         "named preset)");
 }
 
+// A live stream whose peer stopped sending without closing, cancelled by the
+// idle watchdog. Its own type rather than a flag, so the read loop can unwind
+// without the driver going on to finalize a decode nobody is waiting for.
+//
+// NOT handled by to_status: it is thrown and caught inside one handler, and
+// giving it a clause there keeps to_status a statement about exceptions that
+// cross unit boundaries.
+class LiveIdleTimeout : public std::runtime_error {
+public:
+    using std::runtime_error::runtime_error;
+};
+
 // Maps a thrown exception onto the gRPC status the client should see.
 GStatus to_status(const std::exception &err) {
     if (dynamic_cast<const audiocpp_backend::ConfigError *>(&err) != nullptr) {
@@ -1559,7 +1572,7 @@ public:
     // cuts people off mid-sentence. False is not a degradation here, it is the
     // truth: this backend does not know.
     GStatus AudioTranscriptionLive(
-        ServerContext *,
+        ServerContext *context,
         grpc::ServerReaderWriter<backend::TranscriptLiveResponse,
                                  backend::TranscriptLiveRequest> *stream) override {
         try {
@@ -1587,6 +1600,24 @@ public:
             }
             const backend::TranscriptLiveConfig config = incoming.config();
 
+            audiocpp_backend::RequestShape shape;
+            // No request signal chooses the task here: TranscriptLiveConfig has
+            // no prompt field, so has_prompt_text stays false and the only
+            // candidate is Asr. pinned_task is still copied, because without it
+            // the model's `task:` option is dead on this RPC.
+            shape.pinned_task = model->pinned_task();
+
+            // FIRST, before the lane and BEFORE the rate check below, like every
+            // other handler here. A family that cannot stream ASR is answering a
+            // question about itself, and that answer outranks any complaint
+            // about the request: pkg/grpc/grpcerrors/errors.go degrades to the
+            // file path on UNIMPLEMENTED and on nothing else, so a live-incapable
+            // model asked at a wrong rate must not answer INVALID_ARGUMENT and
+            // cost the caller its fallback. Unreachable from LocalAI today,
+            // which always sends 16000, and wrong on its own terms regardless.
+            model->check_can_serve(audiocpp_backend::Rpc::AudioTranscriptionLive,
+                                   shape);
+
             // 16 kHz mono or nothing, and this is a REFUSAL rather than a
             // resample for the reason spelled out at kSpeechSampleRate: the
             // streaming families build their spans in their own 16 kHz feature
@@ -1598,34 +1629,23 @@ public:
             // resampler cannot be applied to it incrementally, so the only
             // honest answers are this refusal or a wrong timestamp.
             //
+            // ZERO means 16000, which is what backend.proto documents. Every
+            // OTHER value is refused, including a negative one: -1 is malformed
+            // rather than absent, and quietly handing it the default would be
+            // this handler inventing a request the caller did not make.
+            //
             // Costs nothing today: core/backend/transcript_live.go hardcodes
-            // liveSampleRate = 16000, and backend.proto documents 0 as 16000 and
-            // says backends may reject other rates.
-            const int sample_rate =
-                config.sample_rate() > 0 ? config.sample_rate() : kSpeechSampleRate;
-            if (sample_rate != kSpeechSampleRate) {
+            // liveSampleRate = 16000.
+            if (config.sample_rate() != 0 &&
+                config.sample_rate() != kSpeechSampleRate) {
                 return GStatus(grpc::StatusCode::INVALID_ARGUMENT,
                                "audio-cpp: live transcription accepts " +
                                    std::to_string(kSpeechSampleRate) +
-                                   " Hz mono PCM only, got " +
-                                   std::to_string(sample_rate) +
+                                   " Hz mono PCM only (or 0 to mean it), got " +
+                                   std::to_string(config.sample_rate()) +
                                    " Hz; resample on the client side");
             }
-
-            audiocpp_backend::RequestShape shape;
-            // No request signal chooses the task here: TranscriptLiveConfig has
-            // no prompt field, so has_prompt_text stays false and the only
-            // candidate is Asr. pinned_task is still copied, because without it
-            // the model's `task:` option is dead on this RPC.
-            shape.pinned_task = model->pinned_task();
-
-            // Before the lane, like every other handler: a family that cannot
-            // stream ASR is answering a question about itself and must not
-            // queue behind somebody else's run to be told no. This is also what
-            // produces the UNIMPLEMENTED the client reads as "degrade to the
-            // file path", so it has to happen before the ack below.
-            model->check_can_serve(audiocpp_backend::Rpc::AudioTranscriptionLive,
-                                   shape);
+            const int sample_rate = kSpeechSampleRate;
 
             // THE LANE IS HELD FOR THE WHOLE STREAM, which is longer than any
             // other handler holds it: as long as the user keeps talking. The
@@ -1718,9 +1738,43 @@ public:
                 }
             };
 
+            // ARMED ONLY NOW, which is after the lane was taken and before the
+            // first audio frame is read, and DISARMED when the read side closes.
+            // Both ends matter:
+            //
+            //   - not earlier, because acquire() above can legitimately block
+            //     for as long as another live stream is running, and cancelling
+            //     a caller for waiting its turn would be this backend punishing
+            //     its own queue. The read of the CONFIG is likewise unwatched:
+            //     it holds no lane, so a client that stalls there wedges
+            //     nothing.
+            //   - not later, because everything past the read loop is OUR
+            //     compute. A decode that outruns the window is not a peer that
+            //     went quiet, and cancelling there would throw away the
+            //     transcript the client is waiting for.
+            //
+            // The status the client actually receives is CANCELLED, not the
+            // DEADLINE_EXCEEDED returned below: TryCancel is the only way to
+            // unblock a synchronous Read, and it decides the wire status itself.
+            // The point of this is not the status, it is that the lane comes
+            // back.
+            audiocpp_backend::IdleWatchdog idle(
+                std::chrono::milliseconds(model->live_idle_timeout_ms()),
+                [context, &model] {
+                    // Logged, because from the client's side this looks like an
+                    // unexplained cancellation and there is nowhere else to say
+                    // why.
+                    std::cerr << "audio-cpp: live stream idle for "
+                              << model->live_idle_timeout_ms()
+                              << " ms with the send side still open; cancelling "
+                                 "it to release the model lane\n";
+                    context->TryCancel();
+                });
+
             std::int64_t consumed_frames = 0;
             const auto next_frames = [&](std::vector<float> &out) {
                 while (stream->Read(&incoming)) {
+                    idle.touch();
                     if (incoming.has_config()) {
                         // backend.proto says a second Config resets the decode
                         // session. audio.cpp cannot do that truthfully: a reset
@@ -1752,6 +1806,21 @@ public:
                     consumed_frames += static_cast<std::int64_t>(pcm.size());
                     return true;
                 }
+                // The read side is closed, one way or the other. Stop the timer
+                // BEFORE saying so, since what follows is finalize().
+                idle.disarm();
+                if (idle.fired()) {
+                    // Read returned false because we cancelled the RPC, not
+                    // because the client closed. Thrown rather than returned so
+                    // the driver does not go on to finalize: there is nobody
+                    // left to send a transcript to, and the whole point was to
+                    // stop occupying the lane.
+                    throw LiveIdleTimeout(
+                        "audio-cpp: no audio frame arrived within " +
+                        std::to_string(model->live_idle_timeout_ms()) +
+                        " ms and the client did not close the stream; cancelled "
+                        "to release the model");
+                }
                 return false;
             };
 
@@ -1762,6 +1831,19 @@ public:
             // also buffers the wire's frames up to the family's own preferred
             // window, which is one second for nemotron_asr and nothing like the
             // 512-sample frames a client's audio callback produces.
+            //
+            // "LIVE" HERE MEANS INCREMENTAL INPUT, NOT LOW LATENCY, and with the
+            // families in the pinned checkout it does not yet mean incremental
+            // OUTPUT either. nemotron_asr's process_audio_chunk only appends to
+            // its buffer (src/models/nemotron_asr/session.cpp:424-436); the whole
+            // decode, and therefore every delta, happens inside finalize(). So
+            // against nemotron_asr every delta arrives AFTER the client closes
+            // its send side, and the driver's policy-window buffering changes
+            // nothing measurable. It is not inert for vibevoice_asr or
+            // higgs_audio_stt, which decode per chunk. Nobody should benchmark
+            // time-to-first-delta against nemotron_asr and conclude this is
+            // broken: it is the family, and the fix is a family that streams its
+            // decode.
             const auto result = audiocpp_backend::run_streaming_live(
                 session, task, next_frames, emit, lane);
 
@@ -1781,6 +1863,11 @@ public:
                 stream->Write(final_response);
             }
             return GStatus::OK;
+        } catch (const LiveIdleTimeout &err) {
+            // For the server's own record, and for a client that somehow reads
+            // it: TryCancel has already decided the wire status is CANCELLED.
+            // The lane was released by unwinding to here, which is the point.
+            return GStatus(grpc::StatusCode::DEADLINE_EXCEEDED, err.what());
         } catch (const std::exception &err) {
             return to_status(err);
         }