New PR for Faster Whisper: Batching Support, Speed Boosts, and Qualit…

…y Enhancements (#856)

Batching Support, Speed Boosts, and Quality Enhancements

---------

Co-authored-by: Hargun Mujral <[email protected]>
Co-authored-by: MahmoudAshraf97 <[email protected]>

MANIFEST.in

@@ -1,3 +1,4 @@
 include faster_whisper/assets/silero_vad.onnx
 include requirements.txt
 include requirements.conversion.txt
+include faster_whisper/assets/pyannote_vad_model.bin

README.md

@@ -69,7 +69,6 @@ segments, info = model.transcribe("audio.mp3", beam_size=5, language="en")
 
 * Python 3.8 or greater
 
-Unlike openai-whisper, FFmpeg does **not** need to be installed on the system. The audio is decoded with the Python library [PyAV](https://github.com/PyAV-Org/PyAV) which bundles the FFmpeg libraries in its package.
 
 ### GPU
 
@@ -166,6 +165,35 @@ for segment in segments:
 segments, _ = model.transcribe("audio.mp3")
 segments = list(segments)  # The transcription will actually run here.
 ```
+
+### multi-segment language detection
+
+To directly use the model for improved language detection, the following code snippet can be used:
+
+```python
+from faster_whisper import WhisperModel
+model = WhisperModel("medium", device="cuda", compute_type="float16")
+language_info = model.detect_language_multi_segment("audio.mp3")
+```
+
+### Batched faster-whisper
+
+
+The batched version of faster-whisper is inspired by [whisper-x](https://github.com/m-bain/whisperX) licensed under the BSD-2 Clause license and integrates its VAD model to this library. We modify this implementation and also replaced the feature extraction with a faster torch-based implementation. Batched version improves the speed upto 10-12x compared to openAI implementation and 3-4x compared to the sequential faster_whisper version. It works by transcribing semantically meaningful audio chunks as batches leading to faster inference. 
+
+The following code snippet illustrates how to run inference with batched version on an example audio file. Please also refer to the test scripts of batched faster whisper.
+
+```python
+from faster_whisper import WhisperModel, BatchedInferencePipeline
+
+model = WhisperModel("medium", device="cuda", compute_type="float16")
+batched_model = BatchedInferencePipeline(model=model)
+segments, info = batched_model.transcribe("audio.mp3", batch_size=16)
+
+for segment in segments:
+    print("[%.2fs -> %.2fs] %s" % (segment.start, segment.end, segment.text))
+```
+
 ### Faster Distil-Whisper
 
 The Distil-Whisper checkpoints are compatible with the Faster-Whisper package. In particular, the latest [distil-large-v3](https://huggingface.co/distil-whisper/distil-large-v3)

benchmark/wer_benchmark.py

@@ -1,5 +1,6 @@
 import argparse
 import json
+import os
 
 from datasets import load_dataset
 from evaluate import load
@@ -26,7 +27,9 @@
 
 # define the evaluation metric
 wer_metric = load("wer")
-normalizer = EnglishTextNormalizer(json.load(open("normalizer.json")))
+
+with open(os.path.join(os.path.dirname(__file__), "normalizer.json"), "r") as f:
+    normalizer = EnglishTextNormalizer(json.load(f))
 
 
 def inference(batch):

faster_whisper/__init__.py

@@ -1,12 +1,13 @@
 from faster_whisper.audio import decode_audio
-from faster_whisper.transcribe import WhisperModel
+from faster_whisper.transcribe import BatchedInferencePipeline, WhisperModel
 from faster_whisper.utils import available_models, download_model, format_timestamp
 from faster_whisper.version import __version__
 
 __all__ = [
     "available_models",
     "decode_audio",
     "WhisperModel",
+    "BatchedInferencePipeline",
     "download_model",
     "format_timestamp",
     "__version__",

faster_whisper/audio.py

@@ -1,19 +1,7 @@
-"""We use the PyAV library to decode the audio: https://github.com/PyAV-Org/PyAV
-
-The advantage of PyAV is that it bundles the FFmpeg libraries so there is no additional
-system dependencies. FFmpeg does not need to be installed on the system.
-
-However, the API is quite low-level so we need to manipulate audio frames directly.
-"""
-
-import gc
-import io
-import itertools
-
 from typing import BinaryIO, Union
 
-import av
-import numpy as np
+import torch
+import torchaudio
 
 
 def decode_audio(
@@ -29,91 +17,42 @@ def decode_audio(
       split_stereo: Return separate left and right channels.
 
     Returns:
-      A float32 Numpy array.
+      A float32 Torch Tensor.
 
       If `split_stereo` is enabled, the function returns a 2-tuple with the
       separated left and right channels.
     """
-    resampler = av.audio.resampler.AudioResampler(
-        format="s16",
-        layout="mono" if not split_stereo else "stereo",
-        rate=sampling_rate,
-    )
-
-    raw_buffer = io.BytesIO()
-    dtype = None
 
-    with av.open(input_file, mode="r", metadata_errors="ignore") as container:
-        frames = container.decode(audio=0)
-        frames = _ignore_invalid_frames(frames)
-        frames = _group_frames(frames, 500000)
-        frames = _resample_frames(frames, resampler)
-
-        for frame in frames:
-            array = frame.to_ndarray()
-            dtype = array.dtype
-            raw_buffer.write(array)
-
-    # It appears that some objects related to the resampler are not freed
-    # unless the garbage collector is manually run.
-    del resampler
-    gc.collect()
-
-    audio = np.frombuffer(raw_buffer.getbuffer(), dtype=dtype)
-
-    # Convert s16 back to f32.
-    audio = audio.astype(np.float32) / 32768.0
+    waveform, audio_sf = torchaudio.load(input_file)  # waveform: channels X T
 
+    if audio_sf != sampling_rate:
+        waveform = torchaudio.functional.resample(
+            waveform, orig_freq=audio_sf, new_freq=sampling_rate
+        )
     if split_stereo:
-        left_channel = audio[0::2]
-        right_channel = audio[1::2]
-        return left_channel, right_channel
-
-    return audio
-
-
-def _ignore_invalid_frames(frames):
-    iterator = iter(frames)
-
-    while True:
-        try:
-            yield next(iterator)
-        except StopIteration:
-            break
-        except av.error.InvalidDataError:
-            continue
-
-
-def _group_frames(frames, num_samples=None):
-    fifo = av.audio.fifo.AudioFifo()
-
-    for frame in frames:
-        frame.pts = None  # Ignore timestamp check.
-        fifo.write(frame)
-
-        if num_samples is not None and fifo.samples >= num_samples:
-            yield fifo.read()
-
-    if fifo.samples > 0:
-        yield fifo.read()
-
+        return waveform[0], waveform[1]
 
-def _resample_frames(frames, resampler):
-    # Add None to flush the resampler.
-    for frame in itertools.chain(frames, [None]):
-        yield from resampler.resample(frame)
+    return waveform.mean(0)
 
 
 def pad_or_trim(array, length: int, *, axis: int = -1):
     """
     Pad or trim the audio array to N_SAMPLES, as expected by the encoder.
     """
+    axis = axis % array.ndim
     if array.shape[axis] > length:
-        array = array.take(indices=range(length), axis=axis)
+        idx = [Ellipsis] * axis + [slice(length)] + [Ellipsis] * (array.ndim - axis - 1)
+        return array[idx]
 
     if array.shape[axis] < length:
-        pad_widths = [(0, 0)] * array.ndim
-        pad_widths[axis] = (0, length - array.shape[axis])
-        array = np.pad(array, pad_widths)
+        pad_widths = (
+            [
+                0,
+            ]
+            * array.ndim
+            * 2
+        )
+        pad_widths[2 * axis] = length - array.shape[axis]
+        array = torch.nn.functional.pad(array, tuple(pad_widths[::-1]))
 
     return array