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The piece is a practical guide to evaluating streaming speech recognition, where word error rate can shift sharply with chunk size, and to measuring latency alongside accuracy. Its performance results (FP16 and per-channel INT8 on Arm SME2, profiled with ExecuTorch on a Pixel 11) come from one device and one model, so readers should treat them as a method to replicate rather than a general benchmark. The main takeaway for ASR developers is that chunk cadence and precision must be chosen together against their own accuracy and latency targets.
Sursă care acoperăPyTorch

Optimizing Streaming Speech Recognition on Arm® Edge Devices with ExecuTorch

PyTorch10 min de cititArticolul și perspectiva sunt în engleză.

When you’re speaking to a device, you expect text to appear as you talk. That makes streaming automatic speech recognition (ASR) a different challenge from batch transcription, where the model... The post Optimizing Streaming Speech Recognition on Arm® Edge Devices with ExecuTorch appeared first on PyTorch.

When you’re speaking to a device, you expect text to appear as you talk. That makes streaming automatic speech recognition (ASR) a different challenge from batch transcription, where the model has the complete recording before it starts. Strong batch accuracy does not necessarily carry over into streaming. One cross-model ASR study illustrates this gap. Qwen3-ASR-1.7B achieved the best batch accuracy among the models tested, but its word error rate (WER) rose by 77% when evaluated with 2.4-second audio chunks. In that study’s cross-model streaming comparison (Table 5), NVIDIA® Nemotron™ Speech Streaming 0.6B had the lowest average streaming WER when operating over 560 ms audio chunks. The comparison makes a critical point: accuracy must be measured at the chunk cadence used in deployment. This case study takes the newer, multilingual Nemotron 3.5 from PyTorch to ExecuTorch and profiles its on-device execution on a Google Pixel 11. We ask how precision and chunk size change streaming quality, how much processing time the audio requires with one inference thread, where the workload spends its time, and which native kernels run. The goal is a practical way to evaluate accuracy…

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