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""" |
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This script supports to load dataset from huggingface and sends it to the server |
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for decoding, in parallel. |
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|
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Usage: |
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# For offline Spark-TTS-0.5B |
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# huggingface dataset |
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num_task=2 |
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python3 client_grpc.py \ |
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--server-addr localhost \ |
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--model-name spark_tts \ |
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--num-tasks $num_task \ |
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--huggingface-dataset yuekai/seed_tts \ |
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--split-name wenetspeech4tts \ |
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--log-dir ./log_concurrent_tasks_${num_task} |
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""" |
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|
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import argparse |
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import asyncio |
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import json |
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|
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import os |
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import time |
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import types |
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from pathlib import Path |
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|
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import numpy as np |
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import soundfile as sf |
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import tritonclient |
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import tritonclient.grpc.aio as grpcclient |
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from tritonclient.utils import np_to_triton_dtype |
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def write_triton_stats(stats, summary_file): |
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with open(summary_file, "w") as summary_f: |
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model_stats = stats["model_stats"] |
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|
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summary_f.write( |
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"The log is parsing from triton_client.get_inference_statistics(), to better human readability. \n" |
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) |
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summary_f.write("To learn more about the log, please refer to: \n") |
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summary_f.write( |
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"1. https://github.com/triton-inference-server/server/blob/main/docs/user_guide/metrics.md \n" |
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) |
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summary_f.write( |
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"2. https://github.com/triton-inference-server/server/issues/5374 \n\n" |
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) |
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summary_f.write( |
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"To better improve throughput, we always would like let requests wait in the queue for a while, and then execute them with a larger batch size. \n" |
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) |
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summary_f.write( |
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"However, there is a trade-off between the increased queue time and the increased batch size. \n" |
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) |
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summary_f.write( |
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"You may change 'max_queue_delay_microseconds' and 'preferred_batch_size' in the model configuration file to achieve this. \n" |
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) |
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summary_f.write( |
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"See https://github.com/triton-inference-server/server/blob/main/docs/user_guide/model_configuration.md#delayed-batching for more details. \n\n" |
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) |
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for model_state in model_stats: |
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if "last_inference" not in model_state: |
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continue |
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summary_f.write(f"model name is {model_state['name']} \n") |
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model_inference_stats = model_state["inference_stats"] |
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total_queue_time_s = int(model_inference_stats["queue"]["ns"]) / 1e9 |
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total_infer_time_s = int(model_inference_stats["compute_infer"]["ns"]) / 1e9 |
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total_input_time_s = int(model_inference_stats["compute_input"]["ns"]) / 1e9 |
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total_output_time_s = ( |
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int(model_inference_stats["compute_output"]["ns"]) / 1e9 |
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) |
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summary_f.write( |
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f"queue time {total_queue_time_s:<5.2f} s, compute infer time {total_infer_time_s:<5.2f} s, compute input time {total_input_time_s:<5.2f} s, compute output time {total_output_time_s:<5.2f} s \n" |
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) |
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model_batch_stats = model_state["batch_stats"] |
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for batch in model_batch_stats: |
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batch_size = int(batch["batch_size"]) |
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compute_input = batch["compute_input"] |
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compute_output = batch["compute_output"] |
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compute_infer = batch["compute_infer"] |
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batch_count = int(compute_infer["count"]) |
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assert ( |
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compute_infer["count"] |
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== compute_output["count"] |
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== compute_input["count"] |
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) |
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compute_infer_time_ms = int(compute_infer["ns"]) / 1e6 |
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compute_input_time_ms = int(compute_input["ns"]) / 1e6 |
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compute_output_time_ms = int(compute_output["ns"]) / 1e6 |
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summary_f.write( |
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f"execuate inference with batch_size {batch_size:<2} total {batch_count:<5} times, total_infer_time {compute_infer_time_ms:<9.2f} ms, avg_infer_time {compute_infer_time_ms:<9.2f}/{batch_count:<5}={compute_infer_time_ms/batch_count:.2f} ms, avg_infer_time_per_sample {compute_infer_time_ms:<9.2f}/{batch_count:<5}/{batch_size}={compute_infer_time_ms/batch_count/batch_size:.2f} ms \n" |
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) |
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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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|
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parser.add_argument( |
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"--server-addr", |
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type=str, |
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default="localhost", |
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help="Address of the server", |
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) |
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|
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parser.add_argument( |
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"--server-port", |
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type=int, |
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default=8001, |
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help="Grpc port of the triton server, default is 8001", |
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) |
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|
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parser.add_argument( |
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"--reference-audio", |
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type=str, |
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default=None, |
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help="Path to a single audio file. It can't be specified at the same time with --manifest-dir", |
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) |
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|
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parser.add_argument( |
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"--reference-text", |
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type=str, |
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default="", |
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help="", |
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) |
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|
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parser.add_argument( |
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"--target-text", |
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type=str, |
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default="", |
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help="", |
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) |
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|
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parser.add_argument( |
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"--huggingface-dataset", |
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type=str, |
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default="yuekai/seed_tts", |
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help="dataset name in huggingface dataset hub", |
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) |
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|
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parser.add_argument( |
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"--split-name", |
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type=str, |
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default="wenetspeech4tts", |
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choices=["wenetspeech4tts", "test_zh", "test_en", "test_hard"], |
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help="dataset split name, default is 'test'", |
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) |
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|
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parser.add_argument( |
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"--manifest-path", |
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type=str, |
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default=None, |
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help="Path to the manifest dir which includes wav.scp trans.txt files.", |
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) |
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|
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parser.add_argument( |
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"--model-name", |
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type=str, |
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default="f5_tts", |
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choices=[ |
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"f5_tts", "spark_tts" |
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], |
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help="triton model_repo module name to request: transducer for k2, attention_rescoring for wenet offline, streaming_wenet for wenet streaming, infer_pipeline for paraformer large offline", |
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) |
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|
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parser.add_argument( |
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"--num-tasks", |
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type=int, |
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default=1, |
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help="Number of concurrent tasks for sending", |
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) |
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|
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parser.add_argument( |
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"--log-interval", |
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type=int, |
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default=5, |
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help="Controls how frequently we print the log.", |
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) |
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|
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parser.add_argument( |
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"--compute-wer", |
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action="store_true", |
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default=False, |
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help="""True to compute WER. |
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""", |
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) |
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|
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parser.add_argument( |
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"--log-dir", |
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type=str, |
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required=False, |
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default="./tmp", |
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help="log directory", |
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) |
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|
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parser.add_argument( |
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"--batch-size", |
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type=int, |
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default=1, |
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help="Inference batch_size per request for offline mode.", |
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) |
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|
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return parser.parse_args() |
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|
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def load_audio(wav_path, target_sample_rate=16000): |
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assert target_sample_rate == 16000, "hard coding in server" |
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if isinstance(wav_path, dict): |
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waveform = wav_path["array"] |
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sample_rate = wav_path["sampling_rate"] |
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else: |
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waveform, sample_rate = sf.read(wav_path) |
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if sample_rate != target_sample_rate: |
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from scipy.signal import resample |
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num_samples = int(len(waveform) * (target_sample_rate / sample_rate)) |
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waveform = resample(waveform, num_samples) |
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return waveform, target_sample_rate |
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|
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async def send( |
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manifest_item_list: list, |
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name: str, |
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triton_client: tritonclient.grpc.aio.InferenceServerClient, |
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protocol_client: types.ModuleType, |
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log_interval: int, |
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model_name: str, |
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padding_duration: int = None, |
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audio_save_dir: str = "./", |
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): |
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total_duration = 0.0 |
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results = [] |
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latency_data = [] |
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task_id = int(name[5:]) |
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|
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print(f"manifest_item_list: {manifest_item_list}") |
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for i, item in enumerate(manifest_item_list): |
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if i % log_interval == 0: |
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print(f"{name}: {i}/{len(manifest_item_list)}") |
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waveform, sample_rate = load_audio(item["audio_filepath"], target_sample_rate=16000) |
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duration = len(waveform) / sample_rate |
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lengths = np.array([[len(waveform)]], dtype=np.int32) |
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reference_text, target_text = item["reference_text"], item["target_text"] |
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estimated_target_duration = duration / len(reference_text) * len(target_text) |
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|
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if padding_duration: |
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|
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samples = np.zeros( |
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( |
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1, |
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padding_duration |
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* sample_rate |
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* ((int(duration) // padding_duration) + 1), |
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), |
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dtype=np.float32, |
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) |
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|
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samples[0, : len(waveform)] = waveform |
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else: |
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samples = waveform |
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|
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samples = samples.reshape(1, -1).astype(np.float32) |
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|
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inputs = [ |
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protocol_client.InferInput( |
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"reference_wav", samples.shape, np_to_triton_dtype(samples.dtype) |
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), |
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protocol_client.InferInput( |
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"reference_wav_len", lengths.shape, np_to_triton_dtype(lengths.dtype) |
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), |
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protocol_client.InferInput("reference_text", [1, 1], "BYTES"), |
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protocol_client.InferInput("target_text", [1, 1], "BYTES") |
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] |
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inputs[0].set_data_from_numpy(samples) |
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inputs[1].set_data_from_numpy(lengths) |
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|
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input_data_numpy = np.array([reference_text], dtype=object) |
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input_data_numpy = input_data_numpy.reshape((1, 1)) |
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inputs[2].set_data_from_numpy(input_data_numpy) |
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input_data_numpy = np.array([target_text], dtype=object) |
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input_data_numpy = input_data_numpy.reshape((1, 1)) |
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inputs[3].set_data_from_numpy(input_data_numpy) |
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|
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outputs = [protocol_client.InferRequestedOutput("waveform")] |
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|
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sequence_id = 100000000 + i + task_id * 10 |
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start = time.time() |
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response = await triton_client.infer( |
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model_name, inputs, request_id=str(sequence_id), outputs=outputs |
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) |
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audio = response.as_numpy("waveform").reshape(-1) |
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end = time.time() - start |
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|
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audio_save_path = os.path.join( |
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audio_save_dir, f"{item['target_audio_path']}.wav" |
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) |
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sf.write(audio_save_path, audio, 16000, "PCM_16") |
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|
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latency_data.append((end, estimated_target_duration)) |
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total_duration += estimated_target_duration |
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|
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return total_duration, latency_data |
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|
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def load_manifests(manifest_path): |
|
with open(manifest_path, "r") as f: |
|
manifest_list = [] |
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for line in f: |
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assert len(line.strip().split("|")) == 4 |
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utt, prompt_text, prompt_wav, gt_text = line.strip().split("|") |
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utt = Path(utt).stem |
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|
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if not os.path.isabs(prompt_wav): |
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prompt_wav = os.path.join(os.path.dirname(manifest_path), prompt_wav) |
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manifest_list.append( |
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{ |
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"audio_filepath": prompt_wav, |
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"reference_text": prompt_text, |
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"target_text": gt_text, |
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"target_audio_path": utt |
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} |
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) |
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return manifest_list |
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|
|
|
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def split_data(data, k): |
|
n = len(data) |
|
if n < k: |
|
print( |
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f"Warning: the length of the input list ({n}) is less than k ({k}). Setting k to {n}." |
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) |
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k = n |
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|
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quotient = n // k |
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remainder = n % k |
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|
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result = [] |
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start = 0 |
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for i in range(k): |
|
if i < remainder: |
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end = start + quotient + 1 |
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else: |
|
end = start + quotient |
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|
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result.append(data[start:end]) |
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start = end |
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|
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return result |
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|
|
|
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async def main(): |
|
args = get_args() |
|
url = f"{args.server_addr}:{args.server_port}" |
|
|
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triton_client = grpcclient.InferenceServerClient(url=url, verbose=False) |
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protocol_client = grpcclient |
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|
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if args.reference_audio: |
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args.num_tasks = 1 |
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args.log_interval = 1 |
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manifest_item_list = [ |
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{ |
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"reference_text": args.reference_text, |
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"target_text": args.target_text, |
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"audio_filepath": args.reference_audio, |
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"target_audio_path": "test", |
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} |
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] |
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elif args.huggingface_dataset: |
|
import datasets |
|
|
|
dataset = datasets.load_dataset( |
|
args.huggingface_dataset, |
|
split=args.split_name, |
|
trust_remote_code=True, |
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) |
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manifest_item_list = [] |
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for i in range(len(dataset)): |
|
manifest_item_list.append( |
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{ |
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"audio_filepath": dataset[i]["prompt_audio"], |
|
"reference_text": dataset[i]["prompt_text"], |
|
"target_audio_path": dataset[i]["id"], |
|
"target_text": dataset[i]["target_text"], |
|
} |
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) |
|
else: |
|
manifest_item_list = load_manifests(args.manifest_path) |
|
|
|
args.num_tasks = min(args.num_tasks, len(manifest_item_list)) |
|
manifest_item_list = split_data(manifest_item_list, args.num_tasks) |
|
|
|
os.makedirs(args.log_dir, exist_ok=True) |
|
tasks = [] |
|
start_time = time.time() |
|
for i in range(args.num_tasks): |
|
task = asyncio.create_task( |
|
send( |
|
manifest_item_list[i], |
|
name=f"task-{i}", |
|
triton_client=triton_client, |
|
protocol_client=protocol_client, |
|
log_interval=args.log_interval, |
|
model_name=args.model_name, |
|
audio_save_dir=args.log_dir, |
|
padding_duration=None, |
|
) |
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) |
|
tasks.append(task) |
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|
|
ans_list = await asyncio.gather(*tasks) |
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|
|
end_time = time.time() |
|
elapsed = end_time - start_time |
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|
|
|
total_duration = 0.0 |
|
latency_data = [] |
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for ans in ans_list: |
|
total_duration += ans[0] |
|
latency_data += ans[1] |
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|
|
rtf = elapsed / total_duration |
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|
|
s = f"RTF: {rtf:.4f}\n" |
|
s += f"total_duration: {total_duration:.3f} seconds\n" |
|
s += f"({total_duration/3600:.2f} hours)\n" |
|
s += f"processing time: {elapsed:.3f} seconds " f"({elapsed/3600:.2f} hours)\n" |
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|
|
latency_list = [chunk_end for (chunk_end, chunk_duration) in latency_data] |
|
latency_ms = sum(latency_list) / float(len(latency_list)) * 1000.0 |
|
latency_variance = np.var(latency_list, dtype=np.float64) * 1000.0 |
|
s += f"latency_variance: {latency_variance:.2f}\n" |
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s += f"latency_50_percentile_ms: {np.percentile(latency_list, 50) * 1000.0:.2f}\n" |
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s += f"latency_90_percentile_ms: {np.percentile(latency_list, 90) * 1000.0:.2f}\n" |
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s += f"latency_95_percentile_ms: {np.percentile(latency_list, 95) * 1000.0:.2f}\n" |
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s += f"latency_99_percentile_ms: {np.percentile(latency_list, 99) * 1000.0:.2f}\n" |
|
s += f"average_latency_ms: {latency_ms:.2f}\n" |
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|
|
print(s) |
|
if args.manifest_path: |
|
name = Path(args.manifest_path).stem |
|
elif args.split_name: |
|
name = args.split_name |
|
with open(f"{args.log_dir}/rtf-{name}.txt", "w") as f: |
|
f.write(s) |
|
|
|
stats = await triton_client.get_inference_statistics(model_name="", as_json=True) |
|
write_triton_stats(stats, f"{args.log_dir}/stats_summary-{name}.txt") |
|
|
|
metadata = await triton_client.get_model_config(model_name=args.model_name, as_json=True) |
|
with open(f"{args.log_dir}/model_config-{name}.json", "w") as f: |
|
json.dump(metadata, f, indent=4) |
|
if __name__ == "__main__": |
|
asyncio.run(main()) |
|
|