import base64 import math import os import time from multiprocessing import Pool import gradio as gr import numpy as np import pytube import requests from processing_whisper import WhisperPrePostProcessor from transformers.models.whisper.tokenization_whisper import TO_LANGUAGE_CODE from transformers.pipelines.audio_utils import ffmpeg_read title = "Whisper JAX: The Fastest Whisper API ⚡️" description = """Whisper JAX is an optimised implementation of the [Whisper model](https://huggingface.co/openai/whisper-large-v2) by OpenAI. It runs on JAX with a TPU v4-8 in the backend. Compared to PyTorch on an A100 GPU, it is over [**70x faster**](https://github.com/sanchit-gandhi/whisper-jax#benchmarks), making it the fastest Whisper API available. Note that at peak times, you may find yourself in the queue for this demo. When you submit a request, your queue position will be shown in the top right-hand side of the demo pane. Once you reach the front of the queue, your audio file will be transcribed, with the progress displayed through a progress bar. To skip the queue, you may wish to create your own inference endpoint, details for which can be found in the [Whisper JAX repository](https://github.com/sanchit-gandhi/whisper-jax#creating-an-endpoint). """ article = "Whisper large-v2 model by OpenAI. Backend running JAX on a TPU v4-8 through the generous support of the [TRC](https://sites.research.google/trc/about/) programme. Whisper JAX [code](https://github.com/sanchit-gandhi/whisper-jax) and Gradio demo by 🤗 Hugging Face." API_URL = os.getenv("API_URL") API_URL_FROM_FEATURES = os.getenv("API_URL_FROM_FEATURES") language_names = sorted(TO_LANGUAGE_CODE.keys()) CHUNK_LENGTH_S = 30 BATCH_SIZE = 16 NUM_PROC = 16 FILE_LIMIT_MB = 1000 def query(payload): response = requests.post(API_URL, json=payload) return response.json(), response.status_code def inference(inputs, task=None, return_timestamps=False): payload = {"inputs": inputs, "task": task, "return_timestamps": return_timestamps} data, status_code = query(payload) if status_code == 200: text = data["text"] else: text = data["detail"] timestamps = data.get("chunks") if timestamps is not None: timestamps = [ f"[{format_timestamp(chunk['timestamp'][0])} -> {format_timestamp(chunk['timestamp'][1])}] {chunk['text']}" for chunk in timestamps ] text = "\n".join(str(feature) for feature in timestamps) return text def chunked_query(payload): response = requests.post(API_URL_FROM_FEATURES, json=payload) return response.json() def forward(batch, task=None, return_timestamps=False): feature_shape = batch["input_features"].shape batch["input_features"] = base64.b64encode(batch["input_features"].tobytes()).decode() outputs = chunked_query( {"batch": batch, "task": task, "return_timestamps": return_timestamps, "feature_shape": feature_shape} ) outputs["tokens"] = np.asarray(outputs["tokens"]) return outputs def identity(batch): return batch # Copied from https://github.com/openai/whisper/blob/c09a7ae299c4c34c5839a76380ae407e7d785914/whisper/utils.py#L50 def format_timestamp(seconds: float, always_include_hours: bool = False, decimal_marker: str = "."): if seconds is not None: milliseconds = round(seconds * 1000.0) hours = milliseconds // 3_600_000 milliseconds -= hours * 3_600_000 minutes = milliseconds // 60_000 milliseconds -= minutes * 60_000 seconds = milliseconds // 1_000 milliseconds -= seconds * 1_000 hours_marker = f"{hours:02d}:" if always_include_hours or hours > 0 else "" return f"{hours_marker}{minutes:02d}:{seconds:02d}{decimal_marker}{milliseconds:03d}" else: # we have a malformed timestamp so just return it as is return seconds if __name__ == "__main__": processor = WhisperPrePostProcessor.from_pretrained("openai/whisper-large-v2") stride_length_s = CHUNK_LENGTH_S / 6 chunk_len = round(CHUNK_LENGTH_S * processor.feature_extractor.sampling_rate) stride_left = stride_right = round(stride_length_s * processor.feature_extractor.sampling_rate) step = chunk_len - stride_left - stride_right pool = Pool(NUM_PROC) def tqdm_generate(inputs: dict, task: str, return_timestamps: bool, progress: gr.Progress): inputs_len = inputs["array"].shape[0] all_chunk_start_idx = np.arange(0, inputs_len, step) num_samples = len(all_chunk_start_idx) num_batches = math.ceil(num_samples / BATCH_SIZE) dummy_batches = list( range(num_batches) ) # Gradio progress bar not compatible with generator, see https://github.com/gradio-app/gradio/issues/3841 dataloader = processor.preprocess_batch(inputs, chunk_length_s=CHUNK_LENGTH_S, batch_size=BATCH_SIZE) progress(0, desc="Pre-processing audio file...") dataloader = pool.map(identity, dataloader) model_outputs = [] start_time = time.time() # iterate over our chunked audio samples for batch, _ in zip(dataloader, progress.tqdm(dummy_batches, desc="Transcribing...")): model_outputs.append(forward(batch, task=task, return_timestamps=return_timestamps)) runtime = time.time() - start_time post_processed = processor.postprocess(model_outputs, return_timestamps=return_timestamps) text = post_processed["text"] timestamps = post_processed.get("chunks") if timestamps is not None: timestamps = [ f"[{format_timestamp(chunk['timestamp'][0])} -> {format_timestamp(chunk['timestamp'][1])}] {chunk['text']}" for chunk in timestamps ] text = "\n".join(str(feature) for feature in timestamps) return text, runtime def transcribe_chunked_audio(inputs, task, return_timestamps, progress=gr.Progress()): progress(0, desc="Loading audio file...") file_size_mb = os.stat(inputs).st_size / (1024 * 1024) if file_size_mb > FILE_LIMIT_MB: raise gr.Error( f"File size exceeds file size limit. Got file of size {file_size_mb:.2f}MB for a limit of {FILE_LIMIT_MB}MB." ) with open(inputs, "rb") as f: inputs = f.read() inputs = ffmpeg_read(inputs, processor.feature_extractor.sampling_rate) inputs = {"array": inputs, "sampling_rate": processor.feature_extractor.sampling_rate} text, runtime = tqdm_generate(inputs, task=task, return_timestamps=return_timestamps, progress=progress) return text, runtime def _return_yt_html_embed(yt_url): video_id = yt_url.split("?v=")[-1] HTML_str = ( f'