* common: add memory buffer overload of read_audio_data whisper-server /inference without --convert passed the uploaded file bytes to read_audio_data as a filename, so ma_decoder_init_file tried to open a path starting with "RIFF" and failed. every request returned HTTP 400 "Invalid request" on builds without WHISPER_FFMPEG, which is the default. factor the PCM extraction into a shared helper and add an overload that decodes straight from a memory buffer via ma_decoder_init_memory, which the function already used for the stdin path. server now calls it with the upload content. the filename overload behavior is unchanged. |
||
|---|---|---|
| .. | ||
| CMakeLists.txt | ||
| README.md | ||
| bench.js | ||
| httplib.h | ||
| server.cpp | ||
README.md
whisper.cpp/examples/server
Simple http server. WAV Files are passed to the inference model via http requests.
https://github.com/ggerganov/whisper.cpp/assets/1991296/e983ee53-8741-4eb5-9048-afe5e4594b8f
Usage
./build/bin/whisper-server -h
usage: ./build/bin/whisper-server [options]
options:
-h, --help [default] show this help message and exit
-t N, --threads N [4 ] number of threads to use during computation
-p N, --processors N [1 ] number of processors to use during computation
-ot N, --offset-t N [0 ] time offset in milliseconds
-on N, --offset-n N [0 ] segment index offset
-d N, --duration N [0 ] duration of audio to process in milliseconds
-mc N, --max-context N [-1 ] maximum number of text context tokens to store
-ml N, --max-len N [0 ] maximum segment length in characters
-sow, --split-on-word [false ] split on word rather than on token
-bo N, --best-of N [2 ] number of best candidates to keep
-bs N, --beam-size N [-1 ] beam size for beam search
-ac N, --audio-ctx N [0 ] audio context size (0 - all)
-wt N, --word-thold N [0.01 ] word timestamp probability threshold
-et N, --entropy-thold N [2.40 ] entropy threshold for decoder fail
-lpt N, --logprob-thold N [-1.00 ] log probability threshold for decoder fail
-debug, --debug-mode [false ] enable debug mode (eg. dump log_mel)
-tr, --translate [false ] translate from source language to english
-di, --diarize [false ] stereo audio diarization
-tdrz, --tinydiarize [false ] enable tinydiarize (requires a tdrz model)
-nf, --no-fallback [false ] do not use temperature fallback while decoding
-ps, --print-special [false ] print special tokens
-pc, --print-colors [false ] print colors
-pr, --print-realtime [false ] print output in realtime
-pp, --print-progress [false ] print progress
-nt, --no-timestamps [false ] do not print timestamps
-l LANG, --language LANG [en ] spoken language ('auto' for auto-detect)
-dl, --detect-language [false ] exit after automatically detecting language
--prompt PROMPT [ ] initial prompt
--carry-initial-prompt [false ] always prepend initial prompt
-m FNAME, --model FNAME [models/ggml-base.en.bin] model path
-oved D, --ov-e-device DNAME [CPU ] the OpenVINO device used for encode inference
-dtw MODEL --dtw MODEL [ ] compute token-level timestamps
--host HOST, [127.0.0.1] Hostname/ip-adress for the server
--port PORT, [8080 ] Port number for the server
--public PATH, [examples/server/public] Path to the public folder
--request-path PATH, [ ] Request path for all requests
--inference-path PATH, [/inference] Inference path for all requests
--convert, [false ] Convert audio to WAV, requires ffmpeg on the server
-sns, --suppress-nst [false ] suppress non-speech tokens
-nth N, --no-speech-thold N [0.60 ] no speech threshold
-nc, --no-context [false ] do not use previous audio context
-ng, --no-gpu [false ] do not use gpu
-fa, --flash-attn [false ] flash attention
Voice Activity Detection (VAD) options:
--vad [false ] enable Voice Activity Detection (VAD)
-vm FNAME, --vad-model FNAME [ ] VAD model path
-vt N, --vad-threshold N [0.50 ] VAD threshold for speech recognition
-vspd N, --vad-min-speech-duration-ms N [250 ] VAD min speech duration (0.0-1.0)
-vsd N, --vad-min-silence-duration-ms N [100 ] VAD min silence duration (to split segments)
-vmsd N, --vad-max-speech-duration-s N [FLT_MAX] VAD max speech duration (auto-split longer)
-vp N, --vad-speech-pad-ms N [30 ] VAD speech padding (extend segments)
-vo N, --vad-samples-overlap N [0.10 ] VAD samples overlap (seconds between segments)
[!WARNING] Do not run the server example with administrative privileges and ensure it's operated in a sandbox environment, especially since it involves risky operations like accepting user file uploads and using ffmpeg for format conversions. Always validate and sanitize inputs to guard against potential security threats.
request examples
/inference
curl 127.0.0.1:8080/inference \
-H "Content-Type: multipart/form-data" \
-F file="@<file-path>" \
-F temperature="0.0" \
-F temperature_inc="0.2" \
-F prompt="<prompt>" \
-F carry_initial_prompt="true" \
-F response_format="json"
/load
curl 127.0.0.1:8080/load \
-H "Content-Type: multipart/form-data" \
-F model="<path-to-model-file>"
Load testing with k6
Note: Install k6 before running the benchmark script.
You can benchmark the Whisper server using the provided bench.js script with k6. This script sends concurrent multipart requests to the /inference endpoint and is fully configurable via environment variables.
Example usage:
k6 run bench.js \
--env FILE_PATH=/absolute/path/to/samples/jfk.wav \
--env BASE_URL=http://127.0.0.1:8080 \
--env ENDPOINT=/inference \
--env CONCURRENCY=4 \
--env TEMPERATURE=0.0 \
--env TEMPERATURE_INC=0.2 \
--env RESPONSE_FORMAT=json
Environment variables:
FILE_PATH: Path to the audio file to send (must be absolute or relative to the k6 working directory)BASE_URL: Server base URL (default:http://127.0.0.1:8080)ENDPOINT: API endpoint (default:/inference)CONCURRENCY: Number of concurrent requests (default: 4)TEMPERATURE: Decoding temperature (default: 0.0)TEMPERATURE_INC: Temperature increment (default: 0.2)RESPONSE_FORMAT: Response format (default:json)
Note:
- The server must be running and accessible at the specified
BASE_URLandENDPOINT. - The script is located in the same directory as this README:
bench.js.