granite/run_eval.py (181 lines of code) (raw):
import argparse
import os
import torch
from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq, models
import evaluate
from normalizer import data_utils
import time
from tqdm import tqdm
# ensure installed transformers supports granite_speech
assert hasattr(models, "granite_speech")
wer_metric = evaluate.load("wer")
torch.set_float32_matmul_precision('high')
def main(args):
processor = AutoProcessor.from_pretrained(args.model_id)
tokenizer = processor.tokenizer
model = AutoModelForSpeechSeq2Seq.from_pretrained(args.model_id).to(args.device)
# create text prompt
chat = [
{
"role": "system",
"content": "Knowledge Cutoff Date: April 2024.\nToday's Date: December 19, 2024.\nYou are Granite, developed by IBM. You are a helpful AI assistant",
},
{
"role": "user",
"content": "<|audio|>can you transcribe the speech into a written format?",
}
]
text = tokenizer.apply_chat_template(
chat, tokenize=False, add_generation_prompt=True
)
gen_kwargs = {"max_new_tokens": args.max_new_tokens, "num_beams": args.num_beams}
def benchmark(batch, min_new_tokens=None):
# Load audio inputs
audios = [audio["array"] for audio in batch["audio"]]
minibatch_size = len(audios)
texts=[text] * minibatch_size
# START TIMING
start_time = time.time()
with torch.autocast(model.device.type, enabled=True):
model_inputs = processor(
texts,
audios,
device=args.device, # Computation device; returned tensors are put on CPU
return_tensors="pt",
).to(args.device)
# Model Inference
model_outputs = model.generate(
**model_inputs,
bos_token_id=tokenizer.bos_token_id,
pad_token_id=tokenizer.pad_token_id,
eos_token_id=tokenizer.eos_token_id,
repetition_penalty=1.0,
**gen_kwargs,
min_new_tokens=min_new_tokens,
)
# Transformers includes the input IDs in the response.
num_input_tokens = model_inputs["input_ids"].shape[-1]
new_tokens = model_outputs[:, num_input_tokens:]
output_text = tokenizer.batch_decode(
new_tokens, add_special_tokens=False, skip_special_tokens=True
)
# END TIMING
runtime = time.time() - start_time
# normalize by minibatch size since we want the per-sample time
batch["transcription_time_s"] = minibatch_size * [runtime / minibatch_size]
# normalize transcriptions with English normalizer
batch["predictions"] = [data_utils.normalizer(pred) for pred in output_text]
batch["references"] = batch["norm_text"]
return batch
if args.warmup_steps is not None:
dataset = data_utils.load_data(args)
dataset = data_utils.prepare_data(dataset)
num_warmup_samples = args.warmup_steps * args.batch_size
if args.streaming:
warmup_dataset = dataset.take(num_warmup_samples)
else:
warmup_dataset = dataset.select(range(min(num_warmup_samples, len(dataset))))
warmup_dataset = iter(warmup_dataset.map(benchmark, batch_size=args.batch_size, batched=True, fn_kwargs={"min_new_tokens": args.max_new_tokens}))
for _ in tqdm(warmup_dataset, desc="Warming up..."):
continue
dataset = data_utils.load_data(args)
if args.max_eval_samples is not None and args.max_eval_samples > 0:
print(f"Subsampling dataset to first {args.max_eval_samples} samples!")
if args.streaming:
dataset = dataset.take(args.max_eval_samples)
else:
dataset = dataset.select(range(min(args.max_eval_samples, len(dataset))))
dataset = data_utils.prepare_data(dataset)
dataset = dataset.map(
benchmark, batch_size=args.batch_size, batched=True, remove_columns=["audio"],
)
all_results = {
"audio_length_s": [],
"transcription_time_s": [],
"predictions": [],
"references": [],
}
result_iter = iter(dataset)
for result in tqdm(result_iter, desc="Samples..."):
for key in all_results:
all_results[key].append(result[key])
# Write manifest results (WER and RTFX)
manifest_path = data_utils.write_manifest(
all_results["references"],
all_results["predictions"],
args.model_id,
args.dataset_path,
args.dataset,
args.split,
audio_length=all_results["audio_length_s"],
transcription_time=all_results["transcription_time_s"],
)
print("Results saved at path:", os.path.abspath(manifest_path))
wer = wer_metric.compute(
references=all_results["references"], predictions=all_results["predictions"]
)
wer = round(100 * wer, 2)
rtfx = round(sum(all_results["audio_length_s"]) / sum(all_results["transcription_time_s"]), 2)
print("WER:", wer, "%", "RTFx:", rtfx)
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument(
"--model_id",
type=str,
required=True,
help="Model identifier. Should be loadable with 🤗 Transformers",
)
parser.add_argument(
"--dataset_path",
type=str,
default="esb/datasets",
help="Dataset path. By default, it is `esb/datasets`",
)
parser.add_argument(
"--dataset",
type=str,
required=True,
help="Dataset name. *E.g.* `'librispeech_asr` for the LibriSpeech ASR dataset, or `'common_voice'` for Common Voice. The full list of dataset names "
"can be found at `https://huggingface.co/datasets/esb/datasets`",
)
parser.add_argument(
"--split",
type=str,
default="test",
help="Split of the dataset. *E.g.* `'validation`' for the dev split, or `'test'` for the test split.",
)
parser.add_argument(
"--device",
type=int,
default=-1,
help="The device to run the pipeline on. -1 for CPU (default), 0 for the first GPU and so on.",
)
parser.add_argument(
"--batch_size",
type=int,
default=16,
help="Number of samples to go through each streamed batch.",
)
parser.add_argument(
"--num_beams",
type=int,
default=1,
help="Number of beams for beam search.",
)
parser.add_argument(
"--max_eval_samples",
type=int,
default=None,
help="Number of samples to be evaluated. Put a lower number e.g. 64 for testing this script.",
)
parser.add_argument(
"--no-streaming",
dest="streaming",
action="store_false",
help="Choose whether you'd like to download the entire dataset or stream it during the evaluation.",
)
parser.add_argument(
"--max_new_tokens",
type=int,
default=None,
help="Maximum number of tokens to generate (for auto-regressive models).",
)
parser.add_argument(
"--warmup_steps",
type=int,
default=2,
help="Number of warm-up steps to run before launching the timed runs.",
)
args = parser.parse_args()
parser.set_defaults(streaming=False)
main(args)