[Bugfix][Model] Fix CohereASR streaming audio-token estimate (unit + subsampling) - #53829
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@claude review |
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❌ @hungnnvidia, A reviewer with write access must run |
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Hello @hungnnvidia, Checked the arithmetic against get_seq_len and the reported numbers hold for durations that are exact multiples of the hop. Two things I would want settled first. The estimate uses ceil where get_seq_len floors, so it runs one token high on roughly 11% of durations, and the test derives its expected value from the same formula it is testing, so it cannot catch that. Everything else reads right to me, and the description already covers disclosure and the duplicate check. Someone with write access will need to kick off /ci run since this touches a serving path. |
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Thanks @aaron-seq, all good catches — I pushed an update:
Let me know if you'd like the subsampling_factor symmetry change too. This will need a maintainer to /ci run since it touches the serving path. |
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Thanks for the review @aaron-seq — I pushed an update addressing all points. This is reproduction when rerunning against the new code: |
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Changes look right to me. Flooring matches On The pre-commit failure is not yours. |
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@DarkLight1337 could you help me review this PR |
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Thanks @aaron-seq, appreciate the thorough review. Glad the floor/ Understood on the |
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@DarkLight1337 I saw you verified this PR, could you help me merge it |
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Cohere folks said they will take a look at this PR |
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@hungnnvidia - Thanks for the bugfix and adding a test! Left a small comment and LGTM otherwise
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The get_num_audio_tokens classmethod used for streaming transcription usage accounting divided by window_stride in seconds (~0.01) as if it were a sample hop, and skipped the encoder subsampling factor. This inflated prompt_tokens by ~5 orders of magnitude (1s audio -> 1.6M tokens). Convert window_stride to a sample hop and divide by subsampling_factor, mirroring the per-request get_seq_len path. Read sample_rate/window_stride from the preprocessor with the same defaults as get_hf_processor, and floor-divide to match get_seq_len (which floors) instead of rounding up. The regression test pins concrete expected token counts. Co-authored-by: Cursor Agent Signed-off-by: hungh <hungh@nvidia.com>
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/ci run |
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✅ Triggered Buildkite CI #86848 for commit |
…subsampling) (vllm-project#53829) Signed-off-by: hungh <hungh@nvidia.com> Co-authored-by: Cyrus Leung <tlleungac@connect.ust.hk>
Summary
CohereAsrForConditionalGeneration.get_num_audio_tokens(audio_duration_s, stt_config, model_config)— the duration-based estimate used to add audio tokens toprompt_tokensfor streaming transcription usage stats — had two bugs in one small classmethod:preprocessor["window_stride"](a stride in seconds, ~0.01) directly as the divisor ofaudio_duration_s * sample_rate(a sample count). The feature extractor's hop isint(window_stride * sample_rate)samples (seeget_hf_processor), so this divided by0.01instead of160.subsampling_factor, unlike the per-request pathCohereASRProcessingInfo.get_num_audio_tokens(get_seq_len(...)thenceil(.../ subsampling_factor)).Together these inflated the reported
prompt_tokensby ~5 orders of magnitude.Root cause
Reached from
vllm/entrypoints/speech_to_text/base/serving.pywhere the return value is added tonum_prompt_tokensfor streaming usage.Reproduction (before fix)
Calling the real classmethod with a realistic config (
window_stride=0.01,sample_rate=16000,subsampling_factor=8):After fix
Fix
Convert
window_strideto a sample hop and divide by the encoder subsampling factor, mirroring the per-requestget_seq_lenpath.Tests
Linters:
Not a duplicate
No open/closed PR touches
get_num_audio_tokensincohere_asr.py(the only open cohere_asr PR, #39259, is about alibrosaimport). Verified viagh pr list --search.AI assistance was used to prepare this change.