Support spoken language identification with whisper (#694)
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172
python-api-examples/spoken-language-identification.py
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172
python-api-examples/spoken-language-identification.py
Executable file
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#!/usr/bin/env python3
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"""
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This script shows how to use Python APIs for spoken languge identification.
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It detects the language spoken in the given wave file.
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Usage:
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1. Download a whisper multilingual model. We use a tiny model below.
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Please refer to https://github.com/k2-fsa/sherpa-onnx/releases/tag/asr-models
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to download more models.
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wget https://github.com/k2-fsa/sherpa-onnx/releases/download/asr-models/sherpa-onnx-whisper-tiny.tar.bz2
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tar xvf sherpa-onnx-whisper-tiny.tar.bz2
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rm sherpa-onnx-whisper-tiny.tar.bz2
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We only use the int8.onnx models below.
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2. Download a test wave.
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You can find many wave files for different languages at
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https://hf-mirror.com/spaces/k2-fsa/spoken-language-identification/tree/main/test_wavs
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wget https://hf-mirror.com/spaces/k2-fsa/spoken-language-identification/resolve/main/test_wavs/de-german.wav
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python3 ./python-api-examples/spoken-language-identification.py
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--whisper-encoder=sherpa-onnx-whisper-tiny/tiny-encoder.int8.onnx \
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--whisper-decoder=sherpa-onnx-whisper-tiny/tiny-decoder.int8.onnx \
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--num-threads=1 \
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./de-german.wav
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"""
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import argparse
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import logging
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import time
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import wave
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from pathlib import Path
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from typing import Tuple
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import numpy as np
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import sherpa_onnx
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def get_args():
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parser = argparse.ArgumentParser(
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formatter_class=argparse.ArgumentDefaultsHelpFormatter
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)
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parser.add_argument(
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"--whisper-encoder",
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required=True,
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type=str,
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help="Path to a multilingual whisper encoder model",
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)
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parser.add_argument(
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"--whisper-decoder",
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required=True,
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type=str,
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help="Path to a multilingual whisper decoder model",
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)
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parser.add_argument(
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"--num-threads",
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type=int,
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default=1,
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help="Number of threads for neural network computation",
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)
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parser.add_argument(
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"--debug",
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type=bool,
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default=False,
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help="True to show debug messages",
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)
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parser.add_argument(
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"--provider",
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type=str,
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default="cpu",
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help="Valid values: cpu, cuda, coreml",
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)
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parser.add_argument(
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"sound_file",
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type=str,
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help="The input sound file to identify. It must be of WAVE"
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"format with a single channel, and each sample has 16-bit, "
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"i.e., int16_t. "
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"The sample rate of the file can be arbitrary and does not need to "
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"be 16 kHz",
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)
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return parser.parse_args()
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def assert_file_exists(filename: str):
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assert Path(filename).is_file(), (
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f"{filename} does not exist!\n"
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"Please refer to "
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"https://k2-fsa.github.io/sherpa/onnx/pretrained_models/whisper/index.html to download it"
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)
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def read_wave(wave_filename: str) -> Tuple[np.ndarray, int]:
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"""
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Args:
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wave_filename:
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Path to a wave file. It should be single channel and each sample should
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be 16-bit. Its sample rate does not need to be 16kHz.
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Returns:
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Return a tuple containing:
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- A 1-D array of dtype np.float32 containing the samples, which are
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normalized to the range [-1, 1].
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- sample rate of the wave file
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"""
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with wave.open(wave_filename) as f:
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assert f.getnchannels() == 1, f.getnchannels()
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assert f.getsampwidth() == 2, f.getsampwidth() # it is in bytes
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num_samples = f.getnframes()
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samples = f.readframes(num_samples)
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samples_int16 = np.frombuffer(samples, dtype=np.int16)
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samples_float32 = samples_int16.astype(np.float32)
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samples_float32 = samples_float32 / 32768
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return samples_float32, f.getframerate()
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def main():
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args = get_args()
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assert_file_exists(args.whisper_encoder)
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assert_file_exists(args.whisper_decoder)
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assert args.num_threads > 0, args.num_threads
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config = sherpa_onnx.SpokenLanguageIdentificationConfig(
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whisper=sherpa_onnx.SpokenLanguageIdentificationWhisperConfig(
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encoder=args.whisper_encoder,
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decoder=args.whisper_decoder,
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),
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num_threads=args.num_threads,
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debug=args.debug,
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provider=args.provider,
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)
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slid = sherpa_onnx.SpokenLanguageIdentification(config)
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samples, sample_rate = read_wave(args.sound_file)
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start_time = time.time()
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stream = slid.create_stream()
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stream.accept_waveform(sample_rate=sample_rate, waveform=samples)
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lang = slid.compute(stream)
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end_time = time.time()
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elapsed_seconds = end_time - start_time
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audio_duration = len(samples) / sample_rate
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real_time_factor = elapsed_seconds / audio_duration
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logging.info(f"File: {args.sound_file}")
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logging.info(f"Detected language: {lang}")
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logging.info(f"Elapsed seconds: {elapsed_seconds:.3f}")
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logging.info(f"Audio duration in seconds: {audio_duration:.3f}")
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logging.info(
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f"RTF: {elapsed_seconds:.3f}/{audio_duration:.3f} = {real_time_factor:.3f}"
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)
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if __name__ == "__main__":
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formatter = "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s"
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logging.basicConfig(format=formatter, level=logging.INFO)
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main()
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