diff --git a/README.md b/README.md
index fc7254c..26f8db0 100644
--- a/README.md
+++ b/README.md
@@ -50,6 +50,7 @@ This repository refines the timestamps of openAI's Whisper model via forced alig
 
 <h2 align="left", id="highlights">New🚨</h2>
 
+- Batch processing: Add `--vad_filter --parallel_bs [int]` for transcribing long audio file in batches (only supported with VAD filtering). Replace `[int]` with a batch size that fits your GPU memory, e.g. `--parallel_bs 16`.
 - VAD filtering: Voice Activity Detection (VAD) from [Pyannote.audio](https://huggingface.co/pyannote/voice-activity-detection) is used as a preprocessing step to remove reliance on whisper timestamps and only transcribe audio segments containing speech. add `--vad_filter` flag, increases timestamp accuracy and robustness (requires more GPU mem due to 30s inputs in wav2vec2)
 - Character level timestamps (see `*.char.ass` file output)
 - Diarization (still in beta, add `--diarize`)
diff --git a/whisperx/__init__.py b/whisperx/__init__.py
index b897f01..1d361e4 100644
--- a/whisperx/__init__.py
+++ b/whisperx/__init__.py
@@ -11,7 +11,7 @@ from tqdm import tqdm
 from .audio import load_audio, log_mel_spectrogram, pad_or_trim
 from .decoding import DecodingOptions, DecodingResult, decode, detect_language
 from .model import Whisper, ModelDimensions
-from .transcribe import transcribe, transcribe_with_vad
+from .transcribe import transcribe, transcribe_with_vad, transcribe_with_vad_parallel
 from .alignment import load_align_model, align
 _MODELS = {
     "tiny.en": "https://openaipublic.azureedge.net/main/whisper/models/d3dd57d32accea0b295c96e26691aa14d8822fac7d9d27d5dc00b4ca2826dd03/tiny.en.pt",
diff --git a/whisperx/transcribe.py b/whisperx/transcribe.py
index 25fbcad..44df48a 100644
--- a/whisperx/transcribe.py
+++ b/whisperx/transcribe.py
@@ -346,6 +346,230 @@ def transcribe_with_vad(
     return output
 
 
+def transcribe_with_vad_parallel(
+    model: "Whisper",
+    audio: Union[str, np.ndarray, torch.Tensor],
+    vad_pipeline,
+    mel = None,
+    verbose: Optional[bool] = None,
+    batch_size = -1,
+    **kwargs
+):
+    """
+    Transcribe per VAD segment
+    """
+
+    if mel is None:
+        mel = log_mel_spectrogram(audio)
+    
+    output = {"segments": []}
+
+    vad_segments = vad_pipeline(audio)
+    # merge segments to approx 30s inputs to make whisper most appropraite
+    vad_segments = merge_chunks(vad_segments)
+
+    ################################
+    ### START of parallelization ###
+    ################################
+
+    # pad mel to a same length
+    start_seconds = [i['start'] for i in vad_segments]
+    end_seconds = [i['end'] for i in vad_segments]
+    duration_list = np.array(end_seconds) - np.array(start_seconds)
+    max_length = round(30 / (HOP_LENGTH / SAMPLE_RATE))
+    offset_list = np.array(start_seconds)
+    chunks = []
+
+    for start_ts, end_ts in zip(start_seconds, end_seconds):
+        start_ts = round(start_ts / (HOP_LENGTH / SAMPLE_RATE))
+        end_ts = round(end_ts / (HOP_LENGTH / SAMPLE_RATE))
+        chunk = mel[:, start_ts:end_ts]
+        chunk = torch.nn.functional.pad(chunk, (0, max_length-chunk.shape[-1]))
+        chunks.append(chunk)
+    
+    mel_chunk = torch.stack(chunks, dim=0).to(model.device)
+    # using 'decode_options1': only support single temperature decoding (no fallbacks)
+    # result_list2 = model.decode(mel_chunk, decode_options1)
+
+    # prepare DecodingOptions
+    temperatures = kwargs.pop("temperature", None)
+    compression_ratio_threshold = kwargs.pop("compression_ratio_threshold", None)
+    logprob_threshold = kwargs.pop("logprob_threshold", None)
+    no_speech_threshold = kwargs.pop("no_speech_threshold", None)
+    condition_on_previous_text = kwargs.pop("condition_on_previous_text", None)
+    initial_prompt = kwargs.pop("initial_prompt", None)
+    
+    t = 0  # TODO: does not upport temperature sweeping
+    if t > 0:
+        # disable beam_size and patience when t > 0
+        kwargs.pop("beam_size", None)
+        kwargs.pop("patience", None)
+    else:
+        # disable best_of when t == 0
+        kwargs.pop("best_of", None)
+
+    options = DecodingOptions(**kwargs, temperature=t)
+    mel_chunk_batches = torch.split(mel_chunk, split_size_or_sections=batch_size)
+    decode_result = []
+    for mel_chunk_batch in mel_chunk_batches:
+        decode_result.extend(model.decode(mel_chunk_batch, options))
+    
+    ##############################
+    ### END of parallelization ###
+    ##############################
+
+    # post processing: get segments rfom batch-decoded results
+    input_stride = exact_div(
+        N_FRAMES, model.dims.n_audio_ctx
+    )  # mel frames per output token: 2
+    language = kwargs["language"]
+    task = kwargs["task"]
+    tokenizer = get_tokenizer(model.is_multilingual, language=language, task=task)
+    result_segments = post_process_results(
+        decode_result, 
+        duration_list, 
+        offset_list,
+        input_stride,
+        language,
+        tokenizer,
+        no_speech_threshold=no_speech_threshold,
+        logprob_threshold=logprob_threshold,
+        verbose=verbose)
+    
+    # post processing: collect outputs
+    assert len(result_segments) == len(vad_segments)
+    for sdx, (seg_t, result) in enumerate(zip(vad_segments, result_segments)):
+        seg_t["text"] = result["text"]
+        output["segments"].append(
+            {
+                "start": seg_t["start"],
+                "end": seg_t["end"],
+                "language": result["language"],
+                "text": result["text"],
+                "seg-text": [x["text"] for x in result["segments"]],
+                "seg-start": [x["start"] for x in result["segments"]],
+                "seg-end": [x["end"] for x in result["segments"]],
+                }
+            )
+    
+    output["language"] = output["segments"][0]["language"]
+
+    return output
+
+
+def post_process_results(
+    result_list, 
+    duration_list, 
+    offset_list,
+    input_stride,
+    language,
+    tokenizer,
+    no_speech_threshold = None,
+    logprob_threshold = None,
+    verbose: Optional[bool] = None,
+    ):
+
+    seek = 0
+    time_precision = (
+        input_stride * HOP_LENGTH / SAMPLE_RATE
+    )  # time per output token: 0.02 (seconds)
+    all_tokens = []
+    all_segments = []
+    outputs = []
+
+    def add_segment(
+        *, start: float, end: float, text_tokens: torch.Tensor, result: DecodingResult
+    ):
+        text = tokenizer.decode([token for token in text_tokens if token < tokenizer.eot])
+        if len(text.strip()) == 0:  # skip empty text output
+            return
+
+        all_segments.append(
+            {
+                "id": len(all_segments),
+                "seek": seek,
+                "start": start,
+                "end": end,
+                "text": text,
+                "tokens": text_tokens.tolist(),
+                "temperature": result.temperature,
+                "avg_logprob": result.avg_logprob,
+                "compression_ratio": result.compression_ratio,
+                "no_speech_prob": result.no_speech_prob,
+            }
+        )
+        if verbose:
+            print(f"[{format_timestamp(start)} --> {format_timestamp(end)}] {text}")
+
+    # process the output
+    for result, segment_duration, timestamp_offset in zip(result_list, duration_list, offset_list):
+        all_tokens = []
+        all_segments = []
+
+        # segment_duration = segment.shape[-1] * HOP_LENGTH / SAMPLE_RATE
+        segment_shape = int(segment_duration / (HOP_LENGTH / SAMPLE_RATE))
+        tokens = torch.tensor(result.tokens)
+
+        if no_speech_threshold is not None:
+            # no voice activity check
+            should_skip = result.no_speech_prob > no_speech_threshold
+            if logprob_threshold is not None and result.avg_logprob > logprob_threshold:
+                # don't skip if the logprob is high enough, despite the no_speech_prob
+                should_skip = False
+
+            if should_skip:
+                seek += segment_shape  # fast-forward to the next segment boundary
+                continue
+        
+        timestamp_tokens: torch.Tensor = tokens.ge(tokenizer.timestamp_begin)
+        consecutive = torch.where(timestamp_tokens[:-1] & timestamp_tokens[1:])[0].add_(1)
+        
+        if len(consecutive) > 0:  # if the output contains two consecutive timestamp tokens
+            last_slice = 0
+            for current_slice in consecutive:
+                sliced_tokens = tokens[last_slice:current_slice]
+                start_timestamp_position = (
+                    sliced_tokens[0].item() - tokenizer.timestamp_begin
+                )
+                end_timestamp_position = (
+                    sliced_tokens[-1].item() - tokenizer.timestamp_begin
+                )
+                add_segment(
+                    start=timestamp_offset + start_timestamp_position * time_precision,
+                    end=timestamp_offset + end_timestamp_position * time_precision,
+                    text_tokens=sliced_tokens[1:-1],
+                    result=result,
+                )
+                last_slice = current_slice
+            last_timestamp_position = (
+                tokens[last_slice - 1].item() - tokenizer.timestamp_begin
+            )
+            seek += last_timestamp_position * input_stride
+            all_tokens.extend(tokens[: last_slice + 1].tolist())
+        else:
+            duration = segment_duration
+            timestamps = tokens[timestamp_tokens.nonzero().flatten()]
+            if len(timestamps) > 0 and timestamps[-1].item() != tokenizer.timestamp_begin:
+                # no consecutive timestamps but it has a timestamp; use the last one.
+                # single timestamp at the end means no speech after the last timestamp.
+                last_timestamp_position = timestamps[-1].item() - tokenizer.timestamp_begin
+                duration = last_timestamp_position * time_precision
+
+            add_segment(
+                start=timestamp_offset,
+                end=timestamp_offset + duration,
+                text_tokens=tokens,
+                result=result,
+            )
+
+            seek += segment_shape
+            all_tokens.extend(tokens.tolist())
+
+        outputs.append(dict(text=tokenizer.decode(all_tokens), segments=all_segments, language=language))
+
+    return outputs
+
+
 def cli():
     from . import available_models
 
@@ -362,6 +586,7 @@ def cli():
     # vad params
     parser.add_argument("--vad_filter", action="store_true", help="Whether to first perform VAD filtering to target only transcribe within VAD. Produces more accurate alignment + timestamp, requires more GPU memory & compute.")
     parser.add_argument("--vad_input", default=None, type=str)
+    parser.add_argument("--parallel_bs", default=-1, type=int, help="Enable parallel transcribing if > 1")
     # diarization params
     parser.add_argument("--diarize", action='store_true')
     parser.add_argument("--min_speakers", default=None, type=int)
@@ -408,6 +633,7 @@ def cli():
     hf_token: str = args.pop("hf_token")
     vad_filter: bool = args.pop("vad_filter")
     vad_input: bool = args.pop("vad_input")
+    parallel_bs: int = args.pop("parallel_bs")
 
     diarize: bool = args.pop("diarize")
     min_speakers: int = args.pop("min_speakers")
@@ -454,8 +680,12 @@ def cli():
 
     for audio_path in args.pop("audio"):
         if vad_filter:
-            print("Performing VAD...")
-            result = transcribe_with_vad(model, audio_path, vad_pipeline, temperature=temperature, **args)
+            if parallel_bs > 1:
+                print("Performing VAD and parallel transcribing ...")
+                result = transcribe_with_vad_parallel(model, audio_path, vad_pipeline, temperature=temperature, batch_size=parallel_bs, **args)
+            else:
+                print("Performing VAD...")
+                result = transcribe_with_vad(model, audio_path, vad_pipeline, temperature=temperature, **args)
         else:
             print("Performing transcription...")
             result = transcribe(model, audio_path, temperature=temperature, **args)