diff --git a/Dockerfile b/Dockerfile
index 42d638c..5466ea8 100644
--- a/Dockerfile
+++ b/Dockerfile
@@ -7,6 +7,7 @@ RUN export DEBIAN_FRONTEND=noninteractive \
     && apt-get -qq update \
     && apt-get -qq install --no-install-recommends \
     ffmpeg \
+    git \
     && rm -rf /var/lib/apt/lists/*
 
 RUN python3 -m venv $POETRY_VENV \
@@ -24,4 +25,10 @@ COPY --from=swagger-ui /usr/share/nginx/html/swagger-ui-bundle.js swagger-ui-ass
 RUN poetry config virtualenvs.in-project true
 RUN poetry install
 
+RUN /app/.venv/bin/pip install pandas transformers nltk pyannote.audio
+RUN git clone --depth 1 https://github.com/m-bain/whisperX.git \
+    && cd whisperX \
+    && $POETRY_VENV/bin/pip install -e .
+
+EXPOSE 9000
 ENTRYPOINT ["gunicorn", "--bind", "0.0.0.0:9000", "--workers", "1", "--timeout", "0", "app.webservice:app", "-k", "uvicorn.workers.UvicornWorker"]
diff --git a/Dockerfile.gpu b/Dockerfile.gpu
index c3a8eee..5b31896 100644
--- a/Dockerfile.gpu
+++ b/Dockerfile.gpu
@@ -11,6 +11,7 @@ RUN export DEBIAN_FRONTEND=noninteractive \
     python${PYTHON_VERSION}-venv \
     python3-pip \
     ffmpeg \
+    git \ 
     && rm -rf /var/lib/apt/lists/*
 
 RUN ln -s -f /usr/bin/python${PYTHON_VERSION} /usr/bin/python3 && \
@@ -35,6 +36,13 @@ COPY --from=swagger-ui /usr/share/nginx/html/swagger-ui.css swagger-ui-assets/sw
 COPY --from=swagger-ui /usr/share/nginx/html/swagger-ui-bundle.js swagger-ui-assets/swagger-ui-bundle.js
 
 RUN poetry install
-RUN $POETRY_VENV/bin/pip install torch==1.13.0+cu117 -f https://download.pytorch.org/whl/torch
+RUN /app/.venv/bin/pip install torch torchaudio pandas transformers nltk pyannote.audio \
+    --index-url https://download.pytorch.org/whl/cu118 \
+    --index-url https://pypi.org/simple/
 
-CMD gunicorn --bind 0.0.0.0:9000 --workers 1 --timeout 0 app.webservice:app -k uvicorn.workers.UvicornWorker
+RUN git clone --depth 1 https://github.com/m-bain/whisperX.git \
+    && cd whisperX \
+    && $POETRY_VENV/bin/pip install --no-dependencies -e .
+
+EXPOSE 9000
+CMD gunicorn --bind 0.0.0.0:9000 --workers 1 --timeout 0 app.webservice:app -k uvicorn.workers.UvicornWorker
\ No newline at end of file
diff --git a/README.md b/README.md
index 75bd94c..3eacd71 100644
--- a/README.md
+++ b/README.md
@@ -11,6 +11,7 @@ Current release (v1.2.0) supports following whisper models:
 
 - [openai/whisper](https://github.com/openai/whisper)@[v20230918](https://github.com/openai/whisper/releases/tag/v20230918)
 - [guillaumekln/faster-whisper](https://github.com/guillaumekln/faster-whisper)@[0.9.0](https://github.com/guillaumekln/faster-whisper/releases/tag/v0.9.0)
+- [whisperX](https://github.com/m-bain/whisperX)@[v3.1.1](https://github.com/m-bain/whisperX/releases/tag/v3.1.1)
 
 
 ## Quick Usage
diff --git a/app/faster_whisper/core.py b/app/faster_whisper/core.py
index aa6563a..987b64b 100644
--- a/app/faster_whisper/core.py
+++ b/app/faster_whisper/core.py
@@ -25,6 +25,7 @@ def transcribe(
         language: Union[str, None],
         initial_prompt: Union[str, None],
         word_timestamps: Union[bool, None],
+        options: Union[dict, None],
         output,
 ):
     options_dict = {"task": task}
diff --git a/app/mbain_whisperx/core.py b/app/mbain_whisperx/core.py
new file mode 100644
index 0000000..9dd2a59
--- /dev/null
+++ b/app/mbain_whisperx/core.py
@@ -0,0 +1,105 @@
+import os
+from typing import BinaryIO, Union
+from io import StringIO
+from threading import Lock
+import torch
+import whisperx
+import whisper
+from whisperx.utils import SubtitlesWriter, ResultWriter, WriteTXT, WriteSRT, WriteVTT, WriteTSV, WriteJSON
+
+model_name= os.getenv("ASR_MODEL", "base")
+hf_token= os.getenv("HF_TOKEN", "")
+x_models = dict()
+
+if torch.cuda.is_available():
+    device = "cuda"
+    model = whisperx.load_model(model_name, device=device)
+    if hf_token != "":
+        diarize_model = whisperx.DiarizationPipeline(use_auth_token=hf_token, device=device)
+else:
+    device = "cpu"
+    model = whisperx.load_model(model_name, device=device)
+    if hf_token != "":
+        diarize_model = whisperx.DiarizationPipeline(use_auth_token=hf_token, device=device)
+model_lock = Lock()
+
+def transcribe(
+    audio,
+    task: Union[str, None],
+    language: Union[str, None],
+    initial_prompt: Union[str, None],
+    word_timestamps: Union[bool, None],
+    options: Union[dict, None],
+    output
+):
+    options_dict = {"task" : task}
+    if language:
+        options_dict["language"] = language
+    if initial_prompt:
+        options_dict["initial_prompt"] = initial_prompt
+    with model_lock:
+        result = model.transcribe(audio, **options_dict)
+
+    # Load the required model and cache it
+    # If we transcribe models in many differen languages, this may lead to OOM propblems
+    if result["language"] in x_models:
+        print('Using chached model')
+        model_x, metadata = x_models[result["language"]]
+    else:
+        print(f'Loading model {result["language"]}')
+        x_models[result["language"]] = whisperx.load_align_model(language_code=result["language"], device=device)
+        model_x, metadata = x_models[result["language"]]
+
+    # Align whisper output    
+    result = whisperx.align(result["segments"], model_x, metadata, audio, device, return_char_alignments=False)
+
+    if options["diarize"]:
+        if hf_token == "":
+            print("Warning! HF_TOKEN is not set. Diarization may not wor as expected.")
+        min_speakers = options["min_speakers"]
+        max_speakers = options["max_speakers"]
+        # add min/max number of speakers if known
+        diarize_segments = diarize_model(audio, min_speakers, max_speakers)
+        result = whisperx.assign_word_speakers(diarize_segments, result)
+
+    outputFile = StringIO()
+    write_result(result, outputFile, output)
+    outputFile.seek(0)
+
+    return outputFile
+
+def language_detection(audio):
+    # load audio and pad/trim it to fit 30 seconds
+    audio = whisper.pad_or_trim(audio)
+
+    # make log-Mel spectrogram and move to the same device as the model
+    mel = whisper.log_mel_spectrogram(audio).to(model.device)
+
+    # detect the spoken language
+    with model_lock:
+        _, probs = model.detect_language(mel)
+    detected_lang_code = max(probs, key=probs.get)
+
+    return detected_lang_code
+
+def write_result(
+    result: dict, file: BinaryIO, output: Union[str, None]
+):
+    if(output == "srt"):
+        if hf_token != "":
+            WriteSRT(SubtitlesWriter).write_result(result, file = file, options = {})
+        else:
+            WriteSRT(ResultWriter).write_result(result, file = file, options = {})
+    elif(output == "vtt"):
+        if hf_token != "":
+            WriteVTT(SubtitlesWriter).write_result(result, file = file, options = {})
+        else:
+            WriteVTT(ResultWriter).write_result(result, file = file, options = {})
+    elif(output == "tsv"):
+        WriteTSV(ResultWriter).write_result(result, file = file, options = {})
+    elif(output == "json"):
+        WriteJSON(ResultWriter).write_result(result, file = file, options = {})
+    elif(output == "txt"):
+        WriteTXT(ResultWriter).write_result(result, file = file, options = {})
+    else:
+        return 'Please select an output method!'
diff --git a/app/openai_whisper/core.py b/app/openai_whisper/core.py
index df3af05..9e1bdea 100644
--- a/app/openai_whisper/core.py
+++ b/app/openai_whisper/core.py
@@ -23,6 +23,7 @@ def transcribe(
         language: Union[str, None],
         initial_prompt: Union[str, None],
         word_timestamps: Union[bool, None],
+        options: Union[dict, None],
         output
 ):
     options_dict = {"task": task}
diff --git a/app/webservice.py b/app/webservice.py
index c3846d3..d919a4a 100644
--- a/app/webservice.py
+++ b/app/webservice.py
@@ -12,8 +12,12 @@
 from whisper import tokenizer
 
 ASR_ENGINE = os.getenv("ASR_ENGINE", "openai_whisper")
+HF_TOKEN = os.getenv("HF_TOKEN", "")
+
 if ASR_ENGINE == "faster_whisper":
     from .faster_whisper.core import transcribe, language_detection
+elif ASR_ENGINE == "whisperx":
+    from .mbain_whisperx.core import transcribe, language_detection
 else:
     from .openai_whisper.core import transcribe, language_detection
 
@@ -59,16 +63,44 @@ async def index():
 
 
 @app.post("/asr", tags=["Endpoints"])
-async def asr(
-        task: Union[str, None] = Query(default="transcribe", enum=["transcribe", "translate"]),
-        language: Union[str, None] = Query(default=None, enum=LANGUAGE_CODES),
-        initial_prompt: Union[str, None] = Query(default=None),
-        audio_file: UploadFile = File(...),
-        encode: bool = Query(default=True, description="Encode audio first through ffmpeg"),
-        output: Union[str, None] = Query(default="txt", enum=["txt", "vtt", "srt", "tsv", "json"]),
-        word_timestamps: bool = Query(default=False, description="Word level timestamps")
+def asr(
+    task : Union[str, None] = Query(default="transcribe", enum=["transcribe", "translate"]),
+    language: Union[str, None] = Query(default=None, enum=LANGUAGE_CODES),
+    initial_prompt: Union[str, None] = Query(default=None),
+    audio_file: UploadFile = File(...),
+    encode : bool = Query(default=True, description="Encode audio first through ffmpeg"),
+    output : Union[str, None] = Query(default="txt", enum=["txt", "vtt", "srt", "tsv", "json"]),
+    word_timestamps : bool = Query(
+        default=False, 
+        description="World level timestamps", 
+        include_in_schema=(True if ASR_ENGINE == "faster_whisper" else False)
+    ),
+    diarize : bool = Query(
+        default=False,
+        description="Diarize the input",
+        include_in_schema=(True if ASR_ENGINE == "whisperx" and HF_TOKEN != "" else False)),
+    min_speakers : Union[int, None] = Query(
+        default=None,
+        description="Min speakers in this file",
+        include_in_schema=(True if ASR_ENGINE == "whisperx" else False)),
+    max_speakers : Union[int, None] = Query(
+        default=None,
+        description="Max speakers in this file",
+        include_in_schema=(True if ASR_ENGINE == "whisperx" else False)),
 ):
-    result = transcribe(load_audio(audio_file.file, encode), task, language, initial_prompt, word_timestamps, output)
+    result = transcribe(
+        load_audio(audio_file.file, encode),
+        task, 
+        language,
+        initial_prompt,
+        word_timestamps,
+        {
+            "diarize": diarize,
+            "min_speakers": min_speakers,
+            "max_speakers": max_speakers
+        },
+        output)
+    
     return StreamingResponse(
         result,
         media_type="text/plain",
diff --git a/docs/index.md b/docs/index.md
index 6dff661..38a9682 100644
--- a/docs/index.md
+++ b/docs/index.md
@@ -5,6 +5,7 @@ Current release (v1.2.0) supports following whisper models:
 
 - [openai/whisper](https://github.com/openai/whisper)@[v20230918](https://github.com/openai/whisper/releases/tag/v20230918)
 - [guillaumekln/faster-whisper](https://github.com/guillaumekln/faster-whisper)@[0.9.0](https://github.com/guillaumekln/faster-whisper/releases/tag/v0.9.0)
+- [whisperX](https://github.com/m-bain/whisperX)@[v3.1.1](https://github.com/m-bain/whisperX/releases/tag/v3.1.1)
 
 ## Quick Usage