torch2.0, remove compile for now, round to times to 3 decimal

Update pyannote and torch version

README.md

@@ -61,23 +61,23 @@ This repository refines the timestamps of openAI's Whisper model via forced alig
 
 
 <h2 align="left" id="setup">Setup ⚙️</h2>
-Tested for PyTorch 0.11, Python 3.8 (use other versions at your own risk!)
+Tested for PyTorch 2.0, Python 3.10 (use other versions at your own risk!)
 
 GPU execution requires the NVIDIA libraries cuBLAS 11.x and cuDNN 8.x to be installed on the system. Please refer to the [CTranslate2 documentation](https://opennmt.net/CTranslate2/installation.html).
 
 
-### 1. Create Python3.8 environment
+### 1. Create Python3.10 environment
 
-`conda create --name whisperx python=3.8`
+`conda create --name whisperx python=3.10`
 
 `conda activate whisperx`
 
 
-### 2. Install PyTorch 0.11.0, e.g. for Linux and Windows:
+### 2. Install PyTorch2.0, e.g. for Linux and Windows CUDA11.7:
 
-`pip install torch==1.11.0+cu113 torchvision==0.12.0+cu113 torchaudio==0.11.0 --extra-index-url https://download.pytorch.org/whl/cu113`
+`pip3 install torch torchvision torchaudio`
 
-See other methods [here.](https://pytorch.org/get-started/previous-versions/#wheel-4)
+See other methods [here.](https://pytorch.org/get-started/locally/)
 
 ### 3. Install this repo
 

requirements.txt

@@ -1,6 +1,5 @@
-torch==1.11.0
+torch==2.0.0
-torchaudio==0.11.0
+torchaudio==2.0.1
-pyannote.audio
 faster-whisper
 transformers
 ffmpeg-python==0.2.0

setup.py

@@ -6,7 +6,7 @@ from setuptools import setup, find_packages
 setup(
     name="whisperx",
     py_modules=["whisperx"],
-    version="3.0.0",
+    version="3.0.2",
     description="Time-Accurate Automatic Speech Recognition using Whisper.",
     readme="README.md",
     python_requires=">=3.8",
@@ -19,7 +19,7 @@ setup(
         for r in pkg_resources.parse_requirements(
             open(os.path.join(os.path.dirname(__file__), "requirements.txt"))
         )
-    ],
+    ] + ["pyannote.audio @ git+https://github.com/pyannote/pyannote-audio@11b56a137a578db9335efc00298f6ec1932e6317"],
     entry_points = {
         'console_scripts': ['whisperx=whisperx.transcribe:cli'],
     },

whisperx/alignment.py

@@ -38,6 +38,7 @@ DEFAULT_ALIGN_MODELS_HF = {
     "fa": "jonatasgrosman/wav2vec2-large-xlsr-53-persian",
     "el": "jonatasgrosman/wav2vec2-large-xlsr-53-greek",
     "tr": "mpoyraz/wav2vec2-xls-r-300m-cv7-turkish",
+    "he": "imvladikon/wav2vec2-xls-r-300m-hebrew",
 }
 
 
@@ -231,8 +232,13 @@ def align(
 
             emission = emissions[0].cpu().detach()
 
-            trellis = get_trellis(emission, tokens)
+            blank_id = 0
-            path = backtrack(trellis, emission, tokens)
+            for char, code in model_dictionary.items():
+                if char == '[pad]' or char == '<pad>':
+                    blank_id = code
+
+            trellis = get_trellis(emission, tokens, blank_id)
+            path = backtrack(trellis, emission, tokens, blank_id)
             if path is None:
                 print(f'Failed to align segment ("{segment["text"]}"): backtrack failed, resorting to original...')
                 break
@@ -262,8 +268,8 @@ def align(
                 start, end, score = None, None, None
                 if cdx in clean_cdx:
                     char_seg = char_segments[clean_cdx.index(cdx)]
-                    start = char_seg.start * ratio + t1
+                    start = round(char_seg.start * ratio + t1, 3)
-                    end = char_seg.end * ratio + t1
+                    end = round(char_seg.end * ratio + t1, 3)
                     score = char_seg.score        
 
                 char_segments_arr["char"].append(char)
@@ -439,8 +445,8 @@ def align(
         word_list.append(
             {
                 "word": curr_text.rstrip(),
-                "start": word_start,
+                "start": wseg.iloc[wdx]['start'],
-                "end": word_end,
+                "end": wseg.iloc[wdx]['end'],
             }
         )
 

whisperx/diarize.py

@@ -1,14 +1,19 @@
 import numpy as np
 import pandas as pd
 from pyannote.audio import Pipeline
+from typing import Optional, Union
+import torch
 
 class DiarizationPipeline:
     def __init__(
         self,
         model_name="pyannote/speaker-diarization@2.1",
         use_auth_token=None,
+        device: Optional[Union[str, torch.device]] = "cpu",
     ):
-        self.model = Pipeline.from_pretrained(model_name, use_auth_token=use_auth_token)
+        if isinstance(device, str):
+            device = torch.device(device)
+        self.model = Pipeline.from_pretrained(model_name, use_auth_token=use_auth_token).to(device)
 
     def __call__(self, audio, min_speakers=None, max_speakers=None):
         segments = self.model(audio, min_speakers=min_speakers, max_speakers=max_speakers)

whisperx/transcribe.py

@@ -72,7 +72,6 @@ def cli():
 
     parser.add_argument("--hf_token", type=str, default=None, help="Hugging Face Access Token to access PyAnnote gated models")
     # parser.add_argument("--model_flush", action="store_true", help="Flush memory from each model after use, reduces GPU requirement but slower processing >1 audio file.")
-    parser.add_argument("--tmp_dir", default=None, help="Temporary directory to write audio file if input if not .wav format (only for VAD).")
     # fmt: on
 
     args = parser.parse_args().__dict__
@@ -86,10 +85,6 @@ def cli():
     # model_flush: bool = args.pop("model_flush")
     os.makedirs(output_dir, exist_ok=True)
 
-    tmp_dir: str = args.pop("tmp_dir")
-    if tmp_dir is not None:
-        os.makedirs(tmp_dir, exist_ok=True)
-
     align_model: str = args.pop("align_model")
     interpolate_method: str = args.pop("interpolate_method")
     no_align: bool = args.pop("no_align")
@@ -193,6 +188,7 @@ def cli():
         if hf_token is None:
             print("Warning, no --hf_token used, needs to be saved in environment variable, otherwise will throw error loading diarization model...")
         tmp_results = results
+        print(">>Performing diarization...")
         results = []
         diarize_model = DiarizationPipeline(use_auth_token=hf_token)
         for result, input_audio_path in tmp_results: