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https://github.com/modelscope/FunASR
synced 2025-09-15 14:48:36 +08:00
ffmpeg
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67d9781251
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@ -11,8 +11,8 @@ model = AutoModel(model="iic/speech_seaco_paraformer_large_asr_nat-zh-cn-16k-com
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vad_model_revision="v2.0.4",
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punc_model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch",
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punc_model_revision="v2.0.4",
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spk_model="iic/speech_campplus_sv_zh-cn_16k-common",
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spk_model_revision="v2.0.2"
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# spk_model="iic/speech_campplus_sv_zh-cn_16k-common",
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# spk_model_revision="v2.0.2"
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)
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res = model.generate(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav",
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@ -14,7 +14,23 @@ try:
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except:
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print("urllib is not installed, if you infer from url, please install it first.")
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import pdb
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import subprocess
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from subprocess import CalledProcessError, run
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def is_ffmpeg_installed():
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try:
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# 尝试运行ffmpeg命令并获取其版本信息
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output = subprocess.check_output(['ffmpeg', '-version'], stderr=subprocess.STDOUT)
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return 'ffmpeg version' in output.decode('utf-8')
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except (subprocess.CalledProcessError, FileNotFoundError):
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# 若运行ffmpeg命令失败,则认为ffmpeg未安装
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return False
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use_ffmpeg=False
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if is_ffmpeg_installed():
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use_ffmpeg
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else:
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print("Warning: ffmpeg is not installed. torchaudio is used to load audio")
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def load_audio_text_image_video(data_or_path_or_list, fs: int = 16000, audio_fs: int = 16000, data_type="sound", tokenizer=None, **kwargs):
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if isinstance(data_or_path_or_list, (list, tuple)):
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@ -34,9 +50,13 @@ def load_audio_text_image_video(data_or_path_or_list, fs: int = 16000, audio_fs:
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if isinstance(data_or_path_or_list, str) and os.path.exists(data_or_path_or_list): # local file
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if data_type is None or data_type == "sound":
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data_or_path_or_list, audio_fs = torchaudio.load(data_or_path_or_list)
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if kwargs.get("reduce_channels", True):
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data_or_path_or_list = data_or_path_or_list.mean(0)
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if use_ffmpeg:
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data_or_path_or_list = _load_audio_ffmpeg(data_or_path_or_list, sr=fs)
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data_or_path_or_list = torch.from_numpy(data_or_path_or_list).squeeze() # [n_samples,]
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else:
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data_or_path_or_list, audio_fs = torchaudio.load(data_or_path_or_list)
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if kwargs.get("reduce_channels", True):
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data_or_path_or_list = data_or_path_or_list.mean(0)
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elif data_type == "text" and tokenizer is not None:
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data_or_path_or_list = tokenizer.encode(data_or_path_or_list)
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elif data_type == "image": # undo
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@ -113,3 +133,41 @@ def extract_fbank(data, data_len = None, data_type: str="sound", frontend=None,
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data_len = torch.tensor([data_len])
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return data.to(torch.float32), data_len.to(torch.int32)
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def _load_audio_ffmpeg(file: str, sr: int = 16000):
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"""
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Open an audio file and read as mono waveform, resampling as necessary
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Parameters
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----------
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file: str
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The audio file to open
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sr: int
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The sample rate to resample the audio if necessary
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Returns
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-------
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A NumPy array containing the audio waveform, in float32 dtype.
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"""
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# This launches a subprocess to decode audio while down-mixing
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# and resampling as necessary. Requires the ffmpeg CLI in PATH.
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# fmt: off
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cmd = [
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"ffmpeg",
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"-nostdin",
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"-threads", "0",
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"-i", file,
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"-f", "s16le",
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"-ac", "1",
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"-acodec", "pcm_s16le",
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"-ar", str(sr),
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"-"
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]
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# fmt: on
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try:
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out = run(cmd, capture_output=True, check=True).stdout
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except CalledProcessError as e:
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raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e
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return np.frombuffer(out, np.int16).flatten().astype(np.float32) / 32768.0
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