speaker-diarization-3.1 using 0% gpu

[MEPO29]

I'm using Google Colab and it is utilizing 0% gpu. Sometimes it uses 100% for a second and then it goes back to 0%. The audio is about 1.5 hours long. Is this normal behaviour? Keep in mind CPU is at 100% almost all of the time.

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[petros94]

Same here

[MohammedAlhajji]

I somehow fixed this by using

pip uninstall pyannote.audio
conda install -c conda-forge pyannote.core
pip install pyannote.audio

on my M3

[KennethTrinh]

this worked for me after uninstalling onnxruntime and onnxruntime-gpu pip install optimum[onnxruntime-gpu] See this link: https://huggingface.co/docs/optimum/main/en/onnxruntime/usage_guides/gpu#installation

[arnavmehta7]

@KennethTrinh I don't understand how that'll work, because new version isn't dependent on onnx.

[pourmand1376]

@KennethTrinh I tested your solution and It didn't work for me.

[KennethTrinh]

oops I was running in a different environment than you guys (was in ec2 instance) - apologies! I tried to reproduce in a colab notebook with a plain old T4 gpu:

Non-blocking code to poll the gpu for usage and memory (I don't have cloud shell since I'm poor!) - run this first if you don't have cloud shell

import threading
import subprocess
import time

def run_nvidia_smi():
    while True:
        try:
            output = subprocess.check_output(
                ['nvidia-smi', '--query-gpu=timestamp,utilization.gpu,utilization.memory', '--format=csv']
            )

            output_str = output.decode('utf-8').strip()
            with open('output.log', 'a') as f:
                f.write(output_str + '\n')
        except subprocess.CalledProcessError as e:
            print(f"Error running nvidia-smi: {e}")
        time.sleep(1)

thread = threading.Thread(target=run_nvidia_smi)
thread.start()

Code to run the diarization - don't forget to define your TOKEN beforehand

!pip install -q --upgrade pyannote.audio
!pip install -q transformers==4.35.2
!pip install -q datasets

import os
import torch
import soundfile as sf
import json
import transformers

from pyannote.audio import Pipeline
from datasets import load_dataset

DIARIZATION_MODEL = "pyannote/speaker-diarization-3.1"
device = torch.device("cuda:0" if torch.cuda.is_available() else "cpu")

concatenated_librispeech = load_dataset("sanchit-gandhi/concatenated_librispeech", split="train", streaming=True)
sample = next(iter(concatenated_librispeech))

diarization_pipeline = Pipeline.from_pretrained(
        DIARIZATION_MODEL, 
        use_auth_token = TOKEN,
    ).to(device)

input_tensor = torch.from_numpy(sample['audio']['array']).float().unsqueeze(0) # (channel, time)
output = diarization_pipeline({"waveform": input_tensor, "sample_rate": sample['audio']['sampling_rate']})
output

My logs show that the gpu was indeed used (albeit very little, but my audio was (347360,) samples , so may change with your audio). The key difference is that I'm passing in a torch tensor of shape (channel, time) , but you guys are just passing the .wav file?

timestamp, utilization.gpu [%], utilization.memory [%]
2023/11/27 19:34:24.171, 1 %, 0 %
timestamp, utilization.gpu [%], utilization.memory [%]
2023/11/27 19:34:25.188, 6 %, 1 %
timestamp, utilization.gpu [%], utilization.memory [%]
2023/11/27 19:34:26.208, 5 %, 0 %

[jaffee]

I found that when passing a filename to the speaker diarization pipeline, I got very poor performance. Upon profiling I discovered this was due to many many calls to get_torchaudio_info and torchaudio._backend.ffmpeg._load_audio_fileobj. This indicated to me that the file was being reprocessed many times unnecessarily (this was all coming from the "crop" method). I noticed that there were very different codepaths if the incoming file object already had a "waveform" computed, so I did the following:

import torchaudio

waveform, sample_rate = torchaudio.load("segment_0.wav")
audio_file = {"waveform": waveform, "sample_rate": sample_rate}

and then I passed audio_file to my pipeline. This took my runtime from 5m to 13s.

I suspect it would be straightforward to change the code to perform this step internally initially and save the user the trouble... and also probably simplify a lot of the downstream code as it could just assume it always had a waveform.

[rmeissn]

https://github.com/pyannote/pyannote-audio/issues/1557#issuecomment-1922466847 (comment above) solved it also for me, using an eGPU and cuda.

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