Adding exampl.py
Browse filesSigned-off-by: taejinp <[email protected]>
- example.py +72 -0
example.py
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# Copyright (c) 2025, NVIDIA CORPORATION. All rights reserved.
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#
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# Licensed under the NVIDIA Open Model License (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# www.nvidia.com/en-us/agreements/enterprise-software/nvidia-open-model-license/
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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"""
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Load one of the NeMo speaker diarization models:
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[Streaming Sortformer Diarizer v2](https://huggingface.co/nvidia/diar_streaming_sortformer_4spk-v2.1),
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[Streaming Sortformer Diarizer v2.1](https://huggingface.co/nvidia/diar_streaming_sortformer_4spk-v2.1)
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"""
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```python
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from nemo.collections.asr.models import SortformerEncLabelModel, ASRModel
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import torch
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# A speaker diarization model is needed for tracking the speech activity of each speaker.
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diar_model = SortformerEncLabelModel.from_pretrained("nvidia/diar_streaming_sortformer_4spk-v2.1").eval().to(torch.device("cuda"))
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asr_model = ASRModel.from_pretrained("nvidia/multitalker-parakeet-streaming-0.6b-v1.nemo").eval().to(torch.device("cuda"))
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# Use the pre-defined dataclass template `MultitalkerTranscriptionConfig` from `multitalker_transcript_config.py`.
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# Configure the diarization model using streaming parameters:
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from multitalker_transcript_config import MultitalkerTranscriptionConfig
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from omegaconf import OmegaConf
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cfg = OmegaConf.structured(MultitalkerTranscriptionConfig())
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cfg.audio_file = "/path/to/your/audio.wav"
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cfg.output_path = "/path/to/output_transcription.json"
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diar_model = MultitalkerTranscriptionConfig.init_diar_model(cfg, diar_model)
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# Load your audio file into a streaming audio buffer to simulate a real-time audio session.
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from nemo.collections.asr.parts.utils.streaming_utils import CacheAwareStreamingAudioBuffer
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samples = [{'audio_filepath': cfg.audio_file}]
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streaming_buffer = CacheAwareStreamingAudioBuffer(
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model=asr_model,
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online_normalization=cfg.online_normalization,
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pad_and_drop_preencoded=cfg.pad_and_drop_preencoded,
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)
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streaming_buffer.append_audio_file(audio_filepath=cfg.audio_file, stream_id=-1)
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streaming_buffer_iter = iter(streaming_buffer)
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# Use the helper class `SpeakerTaggedASR`, which handles all ASR and diarization cache data for streaming.
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from nemo.collections.asr.parts.utils.multispk_transcribe_utils import SpeakerTaggedASR
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multispk_asr_streamer = SpeakerTaggedASR(cfg, asr_model, diar_model)
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for step_num, (chunk_audio, chunk_lengths) in enumerate(streaming_buffer_iter):
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drop_extra_pre_encoded = (
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0
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if step_num == 0 and not cfg.pad_and_drop_preencoded
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else asr_model.encoder.streaming_cfg.drop_extra_pre_encoded
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)
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with torch.inference_mode():
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with torch.amp.autocast(diar_model.device.type, enabled=True):
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with torch.no_grad():
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multispk_asr_streamer.perform_parallel_streaming_stt_spk(
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step_num=step_num,
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chunk_audio=chunk_audio,
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chunk_lengths=chunk_lengths,
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is_buffer_empty=streaming_buffer.is_buffer_empty(),
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drop_extra_pre_encoded=drop_extra_pre_encoded,
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)
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# Generate the speaker-tagged transcript and print it.
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multispk_asr_streamer.generate_seglst_dicts_from_parallel_streaming(samples=samples)
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print(multispk_asr_streamer.instance_manager.seglst_dict_list)
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