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6ee95ab01b
Adds result_map, the engine-to-proto boundary, and wires the offline transcription RPC. The handler branches on the ROUTED task: for Asr the request's prompt is whisper-style decoding context and becomes a request option, for Alignment the same field IS the transcript to align and becomes the text input. Routing has already decided which. The result text is TaskResult.text_output verbatim and is never derived from the segments. audio.cpp carries transcript text in text_output and nowhere else, so deriving it returns an empty transcript for every producer that reports segments without word timing. transcript_assembly already enforces that; this commit's job is not to undo it at the proto boundary, and result_map_ctest pins it there. read_audio_file now takes the sample rate the caller needs. Both file-fed speech handlers ask for 16 kHz mono, for two reasons: silero_vad and sortformer_diar refuse anything else outright, which turned an ordinary 44.1 kHz upload into INTERNAL, and nemotron_asr emits word timestamps in its own 16 kHz feature domain whatever the input was, so only a 16 kHz buffer makes the emitted nanoseconds right. Zero keeps the file's native rate and channels, which is what source separation will need. LoadedModel::check_can_serve answers a capability refusal before the lane is taken and before the input file is read. Routing is a pure read of the immutable capabilities, so a model that cannot serve an RPC no longer waits out somebody else's run to say so. VAD and Diarize use it too. Assisted-by: Claude:claude-opus-5 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
37 lines
1.7 KiB
C++
37 lines
1.7 KiB
C++
#pragma once
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// Converts engine::runtime results into LocalAI proto messages. All of the
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// non-trivial shaping lives in transcript_assembly, which is stdlib-only and
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// unit tested; this unit is the thin engine-typed boundary around it.
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#include "backend.pb.h"
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#include "engine/framework/runtime/session.h"
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namespace audiocpp_backend {
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// Fills text, language, duration, segments and per-segment words.
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//
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// THE RULE: the top-level text is TaskResult.text_output verbatim. It is never
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// derived from segments or words. audio.cpp carries transcript text in
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// text_output and nowhere else: speech_segments, speaker_turns and
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// word_timestamps carry spans and labels and no text at all. Deriving the
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// transcript from them therefore returns an EMPTY text for every producer that
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// reports segments without word timing, which real VibeVoice diarized ASR does.
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// An earlier attempt at this backend shipped exactly that bug. assemble_transcript
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// enforces the rule and is heavily tested; this unit's job is not to re-derive
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// it but to not undo it at the proto boundary.
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//
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// `sample_rate` is the rate the result's spans are expressed in, which is the
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// rate of the AudioBuffer that was handed to the session, NOT the rate of the
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// file the caller uploaded. Those differ whenever read_audio_file resampled,
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// which is why the handler passes the buffer's rate rather than the file's.
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//
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// Segments are replaced, not appended to, so a message filled twice does not
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// accumulate.
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void fill_transcript_result(const engine::runtime::TaskResult &result,
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int sample_rate, float duration_seconds,
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backend::TranscriptResult *out);
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} // namespace audiocpp_backend
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