Merge branch 'main' of github.com:Monadical-SAS/reflector into jose/markers

2025-12-21 12:49:06 +00:00 · 2023-08-16 15:29:00 -05:00
parent d471cdd0a5 33ab54a626
commit 3b204a0a8d
39 changed files with 1214 additions and 888 deletions
--- a/.github/workflows/deploy.yml
+++ b/.github/workflows/deploy.yml
@@ -0,0 +1,44 @@
+name: Deploy to Amazon ECS
+
+on: [deployment, workflow_dispatch]
+
+env:
+  # 384658522150.dkr.ecr.us-east-1.amazonaws.com/reflector
+  AWS_REGION: us-east-1
+  ECR_REPOSITORY: reflector
+
+jobs:
+  deploy:
+    runs-on: ubuntu-latest
+
+    permissions:
+      deployments: write
+      contents: read
+
+    steps:
+      - uses: actions/checkout@v3
+
+      - name: Configure AWS credentials
+        uses: aws-actions/configure-aws-credentials@0e613a0980cbf65ed5b322eb7a1e075d28913a83
+        with:
+          aws-access-key-id: ${{ secrets.AWS_ACCESS_KEY_ID }}
+          aws-secret-access-key: ${{ secrets.AWS_SECRET_ACCESS_KEY }}
+          aws-region: ${{ env.AWS_REGION }}
+
+      - name: Login to Amazon ECR
+        id: login-ecr
+        uses: aws-actions/amazon-ecr-login@62f4f872db3836360b72999f4b87f1ff13310f3a
+
+      - name: Set up QEMU
+        uses: docker/setup-qemu-action@v2
+
+      - name: Set up Docker Buildx
+        uses: docker/setup-buildx-action@v2
+
+      - name: Build and push
+        id: docker_build
+        uses: docker/build-push-action@v4
+        with:
+          context: server
+          push: true
+          tags: ${{ steps.login-ecr.outputs.registry }}/${{ env.ECR_REPOSITORY }}:latest
--- a/server/.gitignore
+++ b/server/.gitignore
@@ -175,3 +175,6 @@ test_samples/
 .vscode/
 artefacts/
 audio_*.wav
+
+# ignore local database
+reflector.sqlite3
--- a/server/env.example
+++ b/server/env.example
@@ -0,0 +1,80 @@
+#
+# This file serve as an example of possible configuration
+# All the settings are described here: reflector/settings.py
+#
+
+## =======================================================
+## Database
+## =======================================================
+
+#DATABASE_URL=sqlite://./reflector.db
+#DATABASE_URL=postgresql://reflector:reflector@localhost:5432/reflector
+
+
+## =======================================================
+## Transcription backend
+##
+## Check reflector/processors/audio_transcript_* for the
+## full list of available transcription backend
+## =======================================================
+
+## Using local whisper (default)
+#TRANSCRIPT_BACKEND=whisper
+#WHISPER_MODEL_SIZE=tiny
+
+## Using serverless modal.com (require reflector-gpu-modal deployed)
+#TRANSCRIPT_BACKEND=modal
+#TRANSCRIPT_URL=https://xxxxx--reflector-transcriber-web.modal.run
+#TRANSCRIPT_MODAL_API_KEY=xxxxx
+
+## Using serverless banana.dev (require reflector-gpu-banana deployed)
+## XXX this service is buggy do not use at the moment
+## XXX it also require the audio to be saved to S3
+#TRANSCRIPT_BACKEND=banana
+#TRANSCRIPT_URL=https://reflector-gpu-banana-xxxxx.run.banana.dev
+#TRANSCRIPT_BANANA_API_KEY=xxx
+#TRANSCRIPT_BANANA_MODEL_KEY=xxx
+#TRANSCRIPT_STORAGE_AWS_ACCESS_KEY_ID=xxx
+#TRANSCRIPT_STORAGE_AWS_SECRET_ACCESS_KEY=xxx
+#TRANSCRIPT_STORAGE_AWS_BUCKET_NAME="reflector-bucket/chunks"
+
+## =======================================================
+## LLM backend
+##
+## Check reflector/llm/* for the full list of available
+## llm backend implementation
+## =======================================================
+
+## Use oobagooda (default)
+#LLM_BACKEND=oobagooda
+#LLM_URL=http://xxx:7860/api/generate/v1
+
+## Using serverless modal.com (require reflector-gpu-modal deployed)
+#LLM_BACKEND=modal
+#LLM_URL=https://xxxxxx--reflector-llm-web.modal.run
+#LLM_MODAL_API_KEY=xxx
+
+## Using serverless banana.dev (require reflector-gpu-banana deployed)
+## XXX this service is buggy do not use at the moment
+#LLM_BACKEND=banana
+#LLM_URL=https://reflector-gpu-banana-xxxxx.run.banana.dev
+#LLM_BANANA_API_KEY=xxxxx
+#LLM_BANANA_MODEL_KEY=xxxxx
+
+## Using OpenAI
+#LLM_BACKEND=openai
+#LLM_OPENAI_KEY=xxx
+#LLM_OPENAI_MODEL=gpt-3.5-turbo
+
+## Using GPT4ALL
+#LLM_BACKEND=openai
+#LLM_URL=http://localhost:4891/v1/completions
+#LLM_OPENAI_MODEL="GPT4All Falcon"
+
+## =======================================================
+## Sentry
+## =======================================================
+
+## Sentry DSN configuration
+#SENTRY_DSN=
+
--- a/server/gpu/modal/README.md
+++ b/server/gpu/modal/README.md
@@ -0,0 +1,92 @@
+# Reflector GPU implementation - Transcription and LLM
+
+This repository hold an API for the GPU implementation of the Reflector API service,
+and use [Modal.com](https://modal.com)
+
+- `reflector_llm.py` - LLM API
+- `reflector_transcriber.py` - Transcription API
+
+## Modal.com deployment
+
+Create a modal secret, and name it `reflector-gpu`.
+It should contain an `REFLECTOR_APIKEY` environment variable with a value.
+
+The deployment is done using [Modal.com](https://modal.com) service.
+
+```
+$ modal deploy reflector_transcriber.py
+...
+└── 🔨 Created web => https://xxxx--reflector-transcriber-web.modal.run
+
+$ modal deploy reflector_llm.py
+...
+└── 🔨 Created web => https://xxxx--reflector-llm-web.modal.run
+```
+
+Then in your reflector api configuration `.env`, you can set theses keys:
+
+```
+TRANSCRIPT_BACKEND=modal
+TRANSCRIPT_URL=https://xxxx--reflector-transcriber-web.modal.run
+TRANSCRIPT_MODAL_API_KEY=REFLECTOR_APIKEY
+
+LLM_BACKEND=modal
+LLM_URL=https://xxxx--reflector-llm-web.modal.run
+LLM_MODAL_API_KEY=REFLECTOR_APIKEY
+```
+
+## API
+
+Authentication must be passed with the `Authorization` header, using the `bearer` scheme.
+
+```
+Authorization: bearer <REFLECTOR_APIKEY>
+```
+
+### Warmup (both)
+
+`POST /warmup`
+
+**response**
+```
+{
+    "status": "ok"
+}
+```
+
+### LLM
+
+`POST /llm`
+
+**request**
+```
+{
+    "prompt": "xxx"
+}
+```
+
+**response**
+```
+{
+    "text": "xxx completed"
+}
+```
+
+### Transcription
+
+`POST /transcribe`
+
+**request** (multipart/form-data)
+
+- `file` - audio file
+- `language` - language code (e.g. `en`)
+
+**response**
+```
+{
+    "text": "xxx",
+    "words": [
+        {"text": "xxx", "start": 0.0, "end": 1.0}
+    ]
+}
+```
--- a/server/gpu/modal/reflector_llm.py
+++ b/server/gpu/modal/reflector_llm.py
@@ -0,0 +1,170 @@
+"""
+Reflector GPU backend - LLM
+===========================
+
+"""
+
+import os
+from modal import Image, method, Stub, asgi_app, Secret
+
+
+# LLM
+LLM_MODEL: str = "lmsys/vicuna-13b-v1.5"
+LLM_LOW_CPU_MEM_USAGE: bool = False
+LLM_TORCH_DTYPE: str = "bfloat16"
+LLM_MAX_NEW_TOKENS: int = 300
+
+IMAGE_MODEL_DIR = "/model"
+
+stub = Stub(name="reflector-llm")
+
+
+def download_llm():
+    from huggingface_hub import snapshot_download
+
+    print("Downloading LLM model")
+    snapshot_download(LLM_MODEL, local_dir=IMAGE_MODEL_DIR)
+    print("LLM model downloaded")
+
+
+def migrate_cache_llm():
+    """
+    XXX The cache for model files in Transformers v4.22.0 has been updated.
+    Migrating your old cache. This is a one-time only operation. You can
+    interrupt this and resume the migration later on by calling
+    `transformers.utils.move_cache()`.
+    """
+    from transformers.utils.hub import move_cache
+
+    print("Moving LLM cache")
+    move_cache()
+    print("LLM cache moved")
+
+
+llm_image = (
+    Image.debian_slim(python_version="3.10.8")
+    .apt_install("git")
+    .pip_install(
+        "transformers",
+        "torch",
+        "sentencepiece",
+        "protobuf",
+        "einops==0.6.1",
+        "hf-transfer~=0.1",
+        "huggingface_hub==0.16.4",
+    )
+    .env({"HF_HUB_ENABLE_HF_TRANSFER": "1"})
+    .run_function(download_llm)
+    .run_function(migrate_cache_llm)
+)
+
+
+@stub.cls(
+    gpu="A100",
+    timeout=60 * 5,
+    container_idle_timeout=60 * 5,
+    concurrency_limit=2,
+    image=llm_image,
+)
+class LLM:
+    def __enter__(self):
+        import torch
+        from transformers import AutoModelForCausalLM, AutoTokenizer
+        from transformers.generation import GenerationConfig
+
+        print("Instance llm model")
+        model = AutoModelForCausalLM.from_pretrained(
+            IMAGE_MODEL_DIR,
+            torch_dtype=getattr(torch, LLM_TORCH_DTYPE),
+            low_cpu_mem_usage=LLM_LOW_CPU_MEM_USAGE,
+        )
+
+        # generation configuration
+        print("Instance llm generation config")
+        model.config.max_new_tokens = LLM_MAX_NEW_TOKENS
+        gen_cfg = GenerationConfig.from_model_config(model.config)
+        gen_cfg.max_new_tokens = LLM_MAX_NEW_TOKENS
+
+        # load tokenizer
+        print("Instance llm tokenizer")
+        tokenizer = AutoTokenizer.from_pretrained(LLM_MODEL)
+
+        # move model to gpu
+        print("Move llm model to GPU")
+        model = model.cuda()
+
+        print("Warmup llm done")
+        self.model = model
+        self.tokenizer = tokenizer
+        self.gen_cfg = gen_cfg
+
+    def __exit__(self, *args):
+        print("Exit llm")
+
+    @method()
+    def warmup(self):
+        print("Warmup ok")
+        return {"status": "ok"}
+
+    @method()
+    def generate(self, prompt: str):
+        print(f"Generate {prompt=}")
+        # tokenize prompt
+        input_ids = self.tokenizer.encode(prompt, return_tensors="pt").to(
+            self.model.device
+        )
+        output = self.model.generate(input_ids, generation_config=self.gen_cfg)
+
+        # decode output
+        response = self.tokenizer.decode(output[0].cpu(), skip_special_tokens=True)
+        print(f"Generated {response=}")
+        return {"text": response}
+
+
+# -------------------------------------------------------------------
+# Web API
+# -------------------------------------------------------------------
+
+
+@stub.function(
+    container_idle_timeout=60 * 10,
+    timeout=60 * 5,
+    secrets=[
+        Secret.from_name("reflector-gpu"),
+    ],
+)
+@asgi_app()
+def web():
+    from fastapi import FastAPI, HTTPException, status, Depends
+    from fastapi.security import OAuth2PasswordBearer
+    from pydantic import BaseModel
+
+    llmstub = LLM()
+
+    app = FastAPI()
+    oauth2_scheme = OAuth2PasswordBearer(tokenUrl="token")
+
+    def apikey_auth(apikey: str = Depends(oauth2_scheme)):
+        if apikey != os.environ["REFLECTOR_GPU_APIKEY"]:
+            raise HTTPException(
+                status_code=status.HTTP_401_UNAUTHORIZED,
+                detail="Invalid API key",
+                headers={"WWW-Authenticate": "Bearer"},
+            )
+
+    class LLMRequest(BaseModel):
+        prompt: str
+
+    @app.post("/llm", dependencies=[Depends(apikey_auth)])
+    async def llm(
+        req: LLMRequest,
+    ):
+        func = llmstub.generate.spawn(prompt=req.prompt)
+        result = func.get()
+        return result
+
+    @app.post("/warmup", dependencies=[Depends(apikey_auth)])
+    async def warmup():
+        return llmstub.warmup.spawn().get()
+
+    return app
--- a/server/gpu/modal/reflector_transcriber.py
+++ b/server/gpu/modal/reflector_transcriber.py
@@ -0,0 +1,173 @@
+"""
+Reflector GPU backend - transcriber
+===================================
+"""
+
+import tempfile
+import os
+from modal import Image, method, Stub, asgi_app, Secret
+from pydantic import BaseModel
+
+
+# Whisper
+WHISPER_MODEL: str = "large-v2"
+WHISPER_COMPUTE_TYPE: str = "float16"
+WHISPER_NUM_WORKERS: int = 1
+WHISPER_CACHE_DIR: str = "/cache/whisper"
+
+stub = Stub(name="reflector-transcriber")
+
+
+def download_whisper():
+    from faster_whisper.utils import download_model
+
+    download_model(WHISPER_MODEL, local_files_only=False)
+
+
+whisper_image = (
+    Image.debian_slim(python_version="3.10.8")
+    .apt_install("git")
+    .pip_install(
+        "faster-whisper",
+        "requests",
+        "torch",
+    )
+    .run_function(download_whisper)
+    .env(
+        {
+            "LD_LIBRARY_PATH": (
+                "/usr/local/lib/python3.10/site-packages/nvidia/cudnn/lib/:"
+                "/opt/conda/lib/python3.10/site-packages/nvidia/cublas/lib/"
+            )
+        }
+    )
+)
+
+
+@stub.cls(
+    gpu="A10G",
+    container_idle_timeout=60,
+    image=whisper_image,
+)
+class Whisper:
+    def __enter__(self):
+        import torch
+        import faster_whisper
+
+        self.use_gpu = torch.cuda.is_available()
+        device = "cuda" if self.use_gpu else "cpu"
+        self.model = faster_whisper.WhisperModel(
+            WHISPER_MODEL,
+            device=device,
+            compute_type=WHISPER_COMPUTE_TYPE,
+            num_workers=WHISPER_NUM_WORKERS,
+        )
+
+    @method()
+    def warmup(self):
+        return {"status": "ok"}
+
+    @method()
+    def transcribe_segment(
+        self,
+        audio_data: str,
+        audio_suffix: str,
+        timestamp: float = 0,
+        language: str = "en",
+    ):
+        with tempfile.NamedTemporaryFile("wb+", suffix=f".{audio_suffix}") as fp:
+            fp.write(audio_data)
+
+            segments, _ = self.model.transcribe(
+                fp.name,
+                language=language,
+                beam_size=5,
+                word_timestamps=True,
+                vad_filter=True,
+                vad_parameters={"min_silence_duration_ms": 500},
+            )
+
+            transcript = ""
+            words = []
+            if segments:
+                segments = list(segments)
+
+                for segment in segments:
+                    transcript += segment.text
+                    for word in segment.words:
+                        words.append(
+                            {
+                                "text": word.word,
+                                "start": round(timestamp + word.start, 3),
+                                "end": round(timestamp + word.end, 3),
+                            }
+                        )
+            return {
+                "text": transcript,
+                "words": words,
+            }
+
+
+# -------------------------------------------------------------------
+# Web API
+# -------------------------------------------------------------------
+
+
+@stub.function(
+    container_idle_timeout=60,
+    timeout=60,
+    secrets=[
+        Secret.from_name("reflector-gpu"),
+    ],
+)
+@asgi_app()
+def web():
+    from fastapi import FastAPI, UploadFile, Form, Depends, HTTPException, status
+    from fastapi.security import OAuth2PasswordBearer
+    from typing_extensions import Annotated
+
+    transcriberstub = Whisper()
+
+    app = FastAPI()
+
+    oauth2_scheme = OAuth2PasswordBearer(tokenUrl="token")
+
+    def apikey_auth(apikey: str = Depends(oauth2_scheme)):
+        if apikey != os.environ["REFLECTOR_GPU_APIKEY"]:
+            raise HTTPException(
+                status_code=status.HTTP_401_UNAUTHORIZED,
+                detail="Invalid API key",
+                headers={"WWW-Authenticate": "Bearer"},
+            )
+
+    class TranscriptionRequest(BaseModel):
+        timestamp: float = 0
+        language: str = "en"
+
+    class TranscriptResponse(BaseModel):
+        result: str
+
+    @app.post("/transcribe", dependencies=[Depends(apikey_auth)])
+    async def transcribe(
+        file: UploadFile,
+        timestamp: Annotated[float, Form()] = 0,
+        language: Annotated[str, Form()] = "en",
+    ):
+        audio_data = await file.read()
+        audio_suffix = file.filename.split(".")[-1]
+        assert audio_suffix in ["wav", "mp3", "ogg", "flac"]
+
+        func = transcriberstub.transcribe_segment.spawn(
+            audio_data=audio_data,
+            audio_suffix=audio_suffix,
+            language=language,
+            timestamp=timestamp,
+        )
+        result = func.get()
+        return result
+
+    @app.post("/warmup", dependencies=[Depends(apikey_auth)])
+    async def warmup():
+        return transcriberstub.warmup.spawn().get()
+
+    return app
--- a/server/poetry.lock
+++ b/server/poetry.lock
@@ -274,6 +274,21 @@ files = [
 [package.dependencies]
 frozenlist = ">=1.1.0"

+[[package]]
+name = "aiosqlite"
+version = "0.19.0"
+description = "asyncio bridge to the standard sqlite3 module"
+optional = false
+python-versions = ">=3.7"
+files = [
+    {file = "aiosqlite-0.19.0-py3-none-any.whl", hash = "sha256:edba222e03453e094a3ce605db1b970c4b3376264e56f32e2a4959f948d66a96"},
+    {file = "aiosqlite-0.19.0.tar.gz", hash = "sha256:95ee77b91c8d2808bd08a59fbebf66270e9090c3d92ffbf260dc0db0b979577d"},
+]
+
+[package.extras]
+dev = ["aiounittest (==1.4.1)", "attribution (==1.6.2)", "black (==23.3.0)", "coverage[toml] (==7.2.3)", "flake8 (==5.0.4)", "flake8-bugbear (==23.3.12)", "flit (==3.7.1)", "mypy (==1.2.0)", "ufmt (==2.1.0)", "usort (==1.0.6)"]
+docs = ["sphinx (==6.1.3)", "sphinx-mdinclude (==0.5.3)"]
+
 [[package]]
 name = "annotated-types"
 version = "0.5.0"
@@ -316,6 +331,59 @@ files = [
    {file = "async_timeout-4.0.2-py3-none-any.whl", hash = "sha256:8ca1e4fcf50d07413d66d1a5e416e42cfdf5851c981d679a09851a6853383b3c"},
 ]

+[[package]]
+name = "asyncpg"
+version = "0.28.0"
+description = "An asyncio PostgreSQL driver"
+optional = false
+python-versions = ">=3.7.0"
+files = [
+    {file = "asyncpg-0.28.0-cp310-cp310-macosx_10_9_x86_64.whl", hash = "sha256:0a6d1b954d2b296292ddff4e0060f494bb4270d87fb3655dd23c5c6096d16d83"},
+    {file = "asyncpg-0.28.0-cp310-cp310-macosx_11_0_arm64.whl", hash = "sha256:0740f836985fd2bd73dca42c50c6074d1d61376e134d7ad3ad7566c4f79f8184"},
+    {file = "asyncpg-0.28.0-cp310-cp310-manylinux_2_17_aarch64.manylinux2014_aarch64.whl", hash = "sha256:e907cf620a819fab1737f2dd90c0f185e2a796f139ac7de6aa3212a8af96c050"},
+    {file = "asyncpg-0.28.0-cp310-cp310-manylinux_2_17_x86_64.manylinux2014_x86_64.whl", hash = "sha256:86b339984d55e8202e0c4b252e9573e26e5afa05617ed02252544f7b3e6de3e9"},
+    {file = "asyncpg-0.28.0-cp310-cp310-musllinux_1_1_aarch64.whl", hash = "sha256:0c402745185414e4c204a02daca3d22d732b37359db4d2e705172324e2d94e85"},
+    {file = "asyncpg-0.28.0-cp310-cp310-musllinux_1_1_x86_64.whl", hash = "sha256:c88eef5e096296626e9688f00ab627231f709d0e7e3fb84bb4413dff81d996d7"},
+    {file = "asyncpg-0.28.0-cp310-cp310-win32.whl", hash = "sha256:90a7bae882a9e65a9e448fdad3e090c2609bb4637d2a9c90bfdcebbfc334bf89"},
+    {file = "asyncpg-0.28.0-cp310-cp310-win_amd64.whl", hash = "sha256:76aacdcd5e2e9999e83c8fbcb748208b60925cc714a578925adcb446d709016c"},
+    {file = "asyncpg-0.28.0-cp311-cp311-macosx_10_9_x86_64.whl", hash = "sha256:a0e08fe2c9b3618459caaef35979d45f4e4f8d4f79490c9fa3367251366af207"},
+    {file = "asyncpg-0.28.0-cp311-cp311-macosx_11_0_arm64.whl", hash = "sha256:b24e521f6060ff5d35f761a623b0042c84b9c9b9fb82786aadca95a9cb4a893b"},
+    {file = "asyncpg-0.28.0-cp311-cp311-manylinux_2_17_aarch64.manylinux2014_aarch64.whl", hash = "sha256:99417210461a41891c4ff301490a8713d1ca99b694fef05dabd7139f9d64bd6c"},
+    {file = "asyncpg-0.28.0-cp311-cp311-manylinux_2_17_x86_64.manylinux2014_x86_64.whl", hash = "sha256:f029c5adf08c47b10bcdc857001bbef551ae51c57b3110964844a9d79ca0f267"},
+    {file = "asyncpg-0.28.0-cp311-cp311-musllinux_1_1_aarch64.whl", hash = "sha256:ad1d6abf6c2f5152f46fff06b0e74f25800ce8ec6c80967f0bc789974de3c652"},
+    {file = "asyncpg-0.28.0-cp311-cp311-musllinux_1_1_x86_64.whl", hash = "sha256:d7fa81ada2807bc50fea1dc741b26a4e99258825ba55913b0ddbf199a10d69d8"},
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 version = "0.14.0"
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+description = "Database Abstraction Library"
+optional = false
+python-versions = "!=3.0.*,!=3.1.*,!=3.2.*,!=3.3.*,!=3.4.*,!=3.5.*,>=2.7"
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+    {file = "SQLAlchemy-1.4.49.tar.gz", hash = "sha256:06ff25cbae30c396c4b7737464f2a7fc37a67b7da409993b182b024cec80aed9"},
+]
+
+[package.dependencies]
+greenlet = {version = "!=0.4.17", markers = "python_version >= \"3\" and (platform_machine == \"win32\" or platform_machine == \"WIN32\" or platform_machine == \"AMD64\" or platform_machine == \"amd64\" or platform_machine == \"x86_64\" or platform_machine == \"ppc64le\" or platform_machine == \"aarch64\")"}
+
+[package.extras]
+aiomysql = ["aiomysql", "greenlet (!=0.4.17)"]
+aiosqlite = ["aiosqlite", "greenlet (!=0.4.17)", "typing-extensions (!=3.10.0.1)"]
+asyncio = ["greenlet (!=0.4.17)"]
+asyncmy = ["asyncmy (>=0.2.3,!=0.2.4)", "greenlet (!=0.4.17)"]
+mariadb-connector = ["mariadb (>=1.0.1,!=1.1.2)"]
+mssql = ["pyodbc"]
+mssql-pymssql = ["pymssql"]
+mssql-pyodbc = ["pyodbc"]
+mypy = ["mypy (>=0.910)", "sqlalchemy2-stubs"]
+mysql = ["mysqlclient (>=1.4.0)", "mysqlclient (>=1.4.0,<2)"]
+mysql-connector = ["mysql-connector-python"]
+oracle = ["cx-oracle (>=7)", "cx-oracle (>=7,<8)"]
+postgresql = ["psycopg2 (>=2.7)"]
+postgresql-asyncpg = ["asyncpg", "greenlet (!=0.4.17)"]
+postgresql-pg8000 = ["pg8000 (>=1.16.6,!=1.29.0)"]
+postgresql-psycopg2binary = ["psycopg2-binary"]
+postgresql-psycopg2cffi = ["psycopg2cffi"]
+pymysql = ["pymysql", "pymysql (<1)"]
+sqlcipher = ["sqlcipher3-binary"]
+
 [[package]]
 name = "stamina"
 version = "23.1.0"
@@ -2996,4 +3234,4 @@ multidict = ">=4.0"
 [metadata]
 lock-version = "2.0"
 python-versions = "^3.11"
-content-hash = "c9924049dacf7310590416f096f5b20f6ed905d8a50edf5e8afcf2c28b70799f"
+content-hash = "ea523f9b74581a7867097a6249d416d8836f4daaf33fde65ea343e4d3502c71c"
--- a/server/pyproject.toml
+++ b/server/pyproject.toml
@@ -23,6 +23,8 @@ fastapi = "^0.100.1"
 sentry-sdk = {extras = ["fastapi"], version = "^1.29.2"}
 httpx = "^0.24.1"
 fastapi-pagination = "^0.12.6"
+databases = {extras = ["aiosqlite", "asyncpg"], version = "^0.7.0"}
+sqlalchemy = "<1.5"


 [tool.poetry.group.dev.dependencies]
--- a/server/reflector/app.py
+++ b/server/reflector/app.py
@@ -2,6 +2,7 @@ from fastapi import FastAPI
 from fastapi.middleware.cors import CORSMiddleware
 from fastapi_pagination import add_pagination
 from fastapi.routing import APIRoute
+import reflector.db  # noqa
 from reflector.views.rtc_offer import router as rtc_offer_router
 from reflector.views.transcripts import router as transcripts_router
 from reflector.events import subscribers_startup, subscribers_shutdown
--- a/server/reflector/db/init.py
+++ b/server/reflector/db/init.py
@@ -0,0 +1,42 @@
+import databases
+import sqlalchemy
+from reflector.events import subscribers_startup, subscribers_shutdown
+from reflector.settings import settings
+
+
+database = databases.Database(settings.DATABASE_URL)
+metadata = sqlalchemy.MetaData()
+
+
+transcripts = sqlalchemy.Table(
+    "transcript",
+    metadata,
+    sqlalchemy.Column("id", sqlalchemy.String, primary_key=True),
+    sqlalchemy.Column("name", sqlalchemy.String),
+    sqlalchemy.Column("status", sqlalchemy.String),
+    sqlalchemy.Column("locked", sqlalchemy.Boolean),
+    sqlalchemy.Column("duration", sqlalchemy.Integer),
+    sqlalchemy.Column("created_at", sqlalchemy.DateTime),
+    sqlalchemy.Column("summary", sqlalchemy.String, nullable=True),
+    sqlalchemy.Column("topics", sqlalchemy.JSON),
+    sqlalchemy.Column("events", sqlalchemy.JSON),
+    # with user attached, optional
+    sqlalchemy.Column("user_id", sqlalchemy.String),
+)
+
+engine = sqlalchemy.create_engine(
+    settings.DATABASE_URL, connect_args={"check_same_thread": False}
+)
+metadata.create_all(engine)
+
+
+async def database_connect():
+    await database.connect()
+
+
+async def database_disconnect():
+    await database.disconnect()
+
+
+subscribers_startup.append(database_connect)
+subscribers_shutdown.append(database_disconnect)
--- a/server/reflector/models.py
+++ b/server/reflector/models.py
@@ -1,211 +0,0 @@
-"""
-Collection of data classes for streamlining and rigidly structuring
-the input and output parameters of functions
-"""
-
-import datetime
-from dataclasses import dataclass
-from typing import List
-from sortedcontainers import SortedDict
-
-import av
-
-
-@dataclass
-class TitleSummaryInput:
-    """
-    Data class for the input to generate title and summaries.
-    The outcome will be used to send query to the LLM for processing.
-    """
-
-    input_text = str
-    transcribed_time = float
-    prompt = str
-    data = dict
-
-    def __init__(self, transcribed_time, input_text=""):
-        self.input_text = input_text
-        self.transcribed_time = transcribed_time
-        self.prompt = f"""
-            ### Human:
-            Create a JSON object as response.The JSON object must have 2 fields:
-            i) title and ii) summary.For the title field,generate a short title
-            for the given text. For the summary field, summarize the given text
-            in three sentences.
-
-            {self.input_text}
-
-            ### Assistant:
-            """
-        self.data = {"prompt": self.prompt}
-        self.headers = {"Content-Type": "application/json"}
-
-
-@dataclass
-class IncrementalResult:
-    """
-    Data class for the result of generating one title and summaries.
-    Defines how a single "topic" looks like.
-    """
-
-    title = str
-    description = str
-    transcript = str
-    timestamp = str
-
-    def __init__(self, title, desc, transcript, timestamp):
-        self.title = title
-        self.description = desc
-        self.transcript = transcript
-        self.timestamp = timestamp
-
-
-@dataclass
-class TitleSummaryOutput:
-    """
-    Data class for the result of all generated titles and summaries.
-    The result will be sent back to the client
-    """
-
-    cmd = str
-    topics = List[IncrementalResult]
-
-    def __init__(self, inc_responses):
-        self.topics = inc_responses
-        self.cmd = "UPDATE_TOPICS"
-
-    def get_result(self) -> dict:
-        """
-        Return the result dict for displaying the transcription
-        :return:
-        """
-        return {"cmd": self.cmd, "topics": self.topics}
-
-
-@dataclass
-class ParseLLMResult:
-    """
-    Data class to parse the result returned by the LLM while generating title
-    and summaries. The result will be sent back to the client.
-    """
-
-    title = str
-    description = str
-    transcript = str
-    timestamp = str
-
-    def __init__(self, param: TitleSummaryInput, output: dict):
-        self.title = output["title"]
-        self.transcript = param.input_text
-        self.description = output.pop("summary")
-        self.timestamp = str(datetime.timedelta(seconds=round(param.transcribed_time)))
-
-    def get_result(self) -> dict:
-        """
-        Return the result dict after parsing the response from LLM
-        :return:
-        """
-        return {
-            "title": self.title,
-            "description": self.description,
-            "transcript": self.transcript,
-            "timestamp": self.timestamp,
-        }
-
-
-@dataclass
-class TranscriptionInput:
-    """
-    Data class to define the input to the transcription function
-    AudioFrames -> input
-    """
-
-    frames = List[av.audio.frame.AudioFrame]
-
-    def __init__(self, frames):
-        self.frames = frames
-
-
-@dataclass
-class TranscriptionOutput:
-    """
-    Dataclass to define the result of the transcription function.
-    The result will be sent back to the client
-    """
-
-    cmd = str
-    result_text = str
-
-    def __init__(self, result_text):
-        self.cmd = "SHOW_TRANSCRIPTION"
-        self.result_text = result_text
-
-    def get_result(self) -> dict:
-        """
-        Return the result dict for displaying the transcription
-        :return:
-        """
-        return {"cmd": self.cmd, "text": self.result_text}
-
-
-@dataclass
-class FinalSummaryResult:
-    """
-    Dataclass to define the result of the final summary function.
-    The result will be sent back to the client.
-    """
-
-    cmd = str
-    final_summary = str
-    duration = str
-
-    def __init__(self, final_summary, time):
-        self.duration = str(datetime.timedelta(seconds=round(time)))
-        self.final_summary = final_summary
-        self.cmd = "DISPLAY_FINAL_SUMMARY"
-
-    def get_result(self) -> dict:
-        """
-        Return the result dict for displaying the final summary
-        :return:
-        """
-        return {
-            "cmd": self.cmd,
-            "duration": self.duration,
-            "summary": self.final_summary,
-        }
-
-
-class BlackListedMessages:
-    """
-    Class to hold the blacklisted messages. These messages should be filtered
-    out and not sent back to the client as part of the transcription.
-    """
-
-    messages = [
-        " Thank you.",
-        " See you next time!",
-        " Thank you for watching!",
-        " Bye!",
-        " And that's what I'm talking about.",
-    ]
-
-
-@dataclass
-class TranscriptionContext:
-    transcription_text: str
-    last_transcribed_time: float
-    incremental_responses: List[IncrementalResult]
-    sorted_transcripts: dict
-    data_channel: None  # FIXME
-    logger: None
-    status: str
-
-    def __init__(self, logger):
-        self.transcription_text = ""
-        self.last_transcribed_time = 0.0
-        self.incremental_responses = []
-        self.data_channel = None
-        self.sorted_transcripts = SortedDict()
-        self.status = "idle"
-        self.logger = logger
--- a/server/reflector/processors/init.py
+++ b/server/reflector/processors/init.py
@@ -1,5 +1,6 @@
 from .base import Processor, ThreadedProcessor, Pipeline  # noqa: F401
 from .types import AudioFile, Transcript, Word, TitleSummary, FinalSummary  # noqa: F401
+from .audio_file_writer import AudioFileWriterProcessor  # noqa: F401
 from .audio_chunker import AudioChunkerProcessor  # noqa: F401
 from .audio_merge import AudioMergeProcessor  # noqa: F401
 from .audio_transcript import AudioTranscriptProcessor  # noqa: F401
--- a/server/reflector/processors/audio_file_writer.py
+++ b/server/reflector/processors/audio_file_writer.py
@@ -0,0 +1,39 @@
+from reflector.processors.base import Processor
+import av
+from pathlib import Path
+
+
+class AudioFileWriterProcessor(Processor):
+    """
+    Write audio frames to a file.
+    """
+
+    INPUT_TYPE = av.AudioFrame
+    OUTPUT_TYPE = av.AudioFrame
+
+    def __init__(self, path: Path | str):
+        super().__init__()
+        if isinstance(path, str):
+            path = Path(path)
+        self.path = path
+        self.out_container = None
+        self.out_stream = None
+
+    async def _push(self, data: av.AudioFrame):
+        if not self.out_container:
+            self.path.parent.mkdir(parents=True, exist_ok=True)
+            self.out_container = av.open(self.path.as_posix(), "w", format="wav")
+            self.out_stream = self.out_container.add_stream(
+                "pcm_s16le", rate=data.sample_rate
+            )
+            for packet in self.out_stream.encode(data):
+                self.out_container.mux(packet)
+        await self.emit(data)
+
+    async def _flush(self):
+        if self.out_container:
+            for packet in self.out_stream.encode(None):
+                self.out_container.mux(packet)
+            self.out_container.close()
+            self.out_container = None
+            self.out_stream = None
--- a/server/reflector/processors/audio_merge.py
+++ b/server/reflector/processors/audio_merge.py
@@ -1,7 +1,8 @@
 from reflector.processors.base import Processor
 from reflector.processors.types import AudioFile
-from pathlib import Path
-import wave
+from time import monotonic_ns
+from uuid import uuid4
+import io
 import av


@@ -24,24 +25,27 @@ class AudioMergeProcessor(Processor):
        sample_width = frame.format.bytes

        # create audio file
-        from time import monotonic_ns
-        from uuid import uuid4
-
        uu = uuid4().hex
-        path = Path(f"audio_{monotonic_ns()}_{uu}.wav")
-        with wave.open(path.as_posix(), "wb") as wf:
-            wf.setnchannels(channels)
-            wf.setsampwidth(sample_width)
-            wf.setframerate(sample_rate)
+        fd = io.BytesIO()
+
+        out_container = av.open(fd, "w", format="wav")
+        out_stream = out_container.add_stream("pcm_s16le", rate=sample_rate)
        for frame in data:
-                wf.writeframes(frame.to_ndarray().tobytes())
+            for packet in out_stream.encode(frame):
+                out_container.mux(packet)
+        for packet in out_stream.encode(None):
+            out_container.mux(packet)
+        out_container.close()
+        fd.seek(0)

        # emit audio file
        audiofile = AudioFile(
-            path=path,
+            name=f"{monotonic_ns()}-{uu}.wav",
+            fd=fd,
            sample_rate=sample_rate,
            channels=channels,
            sample_width=sample_width,
            timestamp=data[0].pts * data[0].time_base,
        )
+
        await self.emit(audiofile)
--- a/server/reflector/processors/audio_transcript_modal.py
+++ b/server/reflector/processors/audio_transcript_modal.py
@@ -48,9 +48,9 @@ class AudioTranscriptModalProcessor(AudioTranscriptProcessor):

    async def _transcript(self, data: AudioFile):
        async with httpx.AsyncClient() as client:
-            self.logger.debug(f"Try to transcribe audio {data.path.name}")
+            self.logger.debug(f"Try to transcribe audio {data.name}")
            files = {
-                "file": (data.path.name, data.path.open("rb")),
+                "file": (data.name, data.fd),
            }
            response = await retry(client.post)(
                self.transcript_url,
--- a/server/reflector/processors/types.py
+++ b/server/reflector/processors/types.py
@@ -1,16 +1,41 @@
-from pydantic import BaseModel
+from pydantic import BaseModel, PrivateAttr
 from pathlib import Path
+import tempfile
+import io


 class AudioFile(BaseModel):
-    path: Path
+    name: str
    sample_rate: int
    channels: int
    sample_width: int
    timestamp: float = 0.0

+    _fd: io.BytesIO = PrivateAttr(None)
+    _path: Path = PrivateAttr(None)
+
+    def __init__(self, fd, **kwargs):
+        super().__init__(**kwargs)
+        self._fd = fd
+
+    @property
+    def fd(self):
+        self._fd.seek(0)
+        return self._fd
+
+    @property
+    def path(self):
+        if self._path is None:
+            # write down to disk
+            filename = tempfile.NamedTemporaryFile(suffix=".wav", delete=False).name
+            self._path = Path(filename)
+            with self._path.open("wb") as f:
+                f.write(self._fd.getbuffer())
+        return self._path
+
    def release(self):
-        self.path.unlink()
+        if self._path:
+            self._path.unlink()


 class Word(BaseModel):
--- a/server/reflector/server.py
+++ b/server/reflector/server.py
@@ -1,381 +0,0 @@
-import argparse
-import asyncio
-import datetime
-import json
-import os
-import wave
-import uuid
-from concurrent.futures import ThreadPoolExecutor
-from typing import NoReturn, Union
-
-import aiohttp_cors
-import av
-import requests
-from aiohttp import web
-from aiortc import MediaStreamTrack, RTCPeerConnection, RTCSessionDescription
-from aiortc.contrib.media import MediaRelay
-from faster_whisper import WhisperModel
-
-from reflector.models import (
-    BlackListedMessages,
-    FinalSummaryResult,
-    ParseLLMResult,
-    TitleSummaryInput,
-    TitleSummaryOutput,
-    TranscriptionInput,
-    TranscriptionOutput,
-    TranscriptionContext,
-)
-from reflector.logger import logger
-from reflector.utils.run_utils import run_in_executor
-from reflector.settings import settings
-
-# WebRTC components
-pcs = set()
-relay = MediaRelay()
-executor = ThreadPoolExecutor()
-
-# Transcription model
-model = WhisperModel("tiny", device="cpu", compute_type="float32", num_workers=12)
-
-# LLM
-LLM_URL = settings.LLM_URL
-if not LLM_URL:
-    assert settings.LLM_BACKEND == "oobagooda"
-    LLM_URL = f"http://{settings.LLM_HOST}:{settings.LLM_PORT}/api/v1/generate"
-logger.info(f"Using LLM [{settings.LLM_BACKEND}]: {LLM_URL}")
-
-
-def parse_llm_output(
-    param: TitleSummaryInput, response: requests.Response
-) -> Union[None, ParseLLMResult]:
-    """
-    Function to parse the LLM response
-    :param param:
-    :param response:
-    :return:
-    """
-    try:
-        output = json.loads(response.json()["results"][0]["text"])
-        return ParseLLMResult(param, output)
-    except Exception:
-        logger.exception("Exception while parsing LLM output")
-        return None
-
-
-def get_title_and_summary(
-    ctx: TranscriptionContext, param: TitleSummaryInput
-) -> Union[None, TitleSummaryOutput]:
-    """
-    From the input provided (transcript), query the LLM to generate
-    topics and summaries
-    :param param:
-    :return:
-    """
-    logger.info("Generating title and summary")
-
-    # TODO : Handle unexpected output formats from the model
-    try:
-        response = requests.post(LLM_URL, headers=param.headers, json=param.data)
-        output = parse_llm_output(param, response)
-        if output:
-            result = output.get_result()
-            ctx.incremental_responses.append(result)
-            return TitleSummaryOutput(ctx.incremental_responses)
-    except Exception:
-        logger.exception("Exception while generating title and summary")
-        return None
-
-
-def channel_send(channel, message: str) -> NoReturn:
-    """
-    Send text messages via the data channel
-    :param channel:
-    :param message:
-    :return:
-    """
-    if channel:
-        channel.send(message)
-
-
-def channel_send_increment(
-    channel, param: Union[FinalSummaryResult, TitleSummaryOutput]
-) -> NoReturn:
-    """
-    Send the incremental topics and summaries via the data channel
-    :param channel:
-    :param param:
-    :return:
-    """
-    if channel and param:
-        message = param.get_result()
-        channel.send(json.dumps(message))
-
-
-def channel_send_transcript(ctx: TranscriptionContext) -> NoReturn:
-    """
-    Send the transcription result via the data channel
-    :param channel:
-    :return:
-    """
-    if not ctx.data_channel:
-        return
-    try:
-        least_time = next(iter(ctx.sorted_transcripts))
-        message = ctx.sorted_transcripts[least_time].get_result()
-        if message:
-            del ctx.sorted_transcripts[least_time]
-            if message["text"] not in BlackListedMessages.messages:
-                ctx.data_channel.send(json.dumps(message))
-        # Due to exceptions if one of the earlier batches can't return
-        # a transcript, we don't want to be stuck waiting for the result
-        # With the threshold size of 3, we pop the first(lost) element
-        else:
-            if len(ctx.sorted_transcripts) >= 3:
-                del ctx.sorted_transcripts[least_time]
-    except Exception:
-        logger.exception("Exception while sending transcript")
-
-
-def get_transcription(
-    ctx: TranscriptionContext, input_frames: TranscriptionInput
-) -> Union[None, TranscriptionOutput]:
-    """
-    From the collected audio frames create transcription by inferring from
-    the chosen transcription model
-    :param input_frames:
-    :return:
-    """
-    ctx.logger.info("Transcribing..")
-    ctx.sorted_transcripts[input_frames.frames[0].time] = None
-
-    # TODO: Find cleaner way, watch "no transcription" issue below
-    # Passing IO objects instead of temporary files throws an error
-    # Passing ndarray (type casted with float) does not give any
-    # transcription. Refer issue,
-    # https://github.com/guillaumekln/faster-whisper/issues/369
-    audio_file = "test" + str(datetime.datetime.now())
-    wf = wave.open(audio_file, "wb")
-    wf.setnchannels(settings.AUDIO_CHANNELS)
-    wf.setframerate(settings.AUDIO_SAMPLING_RATE)
-    wf.setsampwidth(settings.AUDIO_SAMPLING_WIDTH)
-
-    for frame in input_frames.frames:
-        wf.writeframes(b"".join(frame.to_ndarray()))
-    wf.close()
-
-    result_text = ""
-
-    try:
-        segments, _ = model.transcribe(
-            audio_file,
-            language="en",
-            beam_size=5,
-            vad_filter=True,
-            vad_parameters={"min_silence_duration_ms": 500},
-        )
-        os.remove(audio_file)
-        segments = list(segments)
-        result_text = ""
-        duration = 0.0
-        for segment in segments:
-            result_text += segment.text
-            start_time = segment.start
-            end_time = segment.end
-            if not segment.start:
-                start_time = 0.0
-            if not segment.end:
-                end_time = 5.5
-            duration += end_time - start_time
-
-        ctx.last_transcribed_time += duration
-        ctx.transcription_text += result_text
-
-    except Exception:
-        logger.exception("Exception while transcribing")
-
-    result = TranscriptionOutput(result_text)
-    ctx.sorted_transcripts[input_frames.frames[0].time] = result
-    return result
-
-
-def get_final_summary_response(ctx: TranscriptionContext) -> FinalSummaryResult:
-    """
-    Collate the incremental summaries generated so far and return as the final
-    summary
-    :return:
-    """
-    final_summary = ""
-
-    # Collate inc summaries
-    for topic in ctx.incremental_responses:
-        final_summary += topic["description"]
-
-    response = FinalSummaryResult(final_summary, ctx.last_transcribed_time)
-
-    with open(
-        "./artefacts/meeting_titles_and_summaries.txt", "a", encoding="utf-8"
-    ) as file:
-        file.write(json.dumps(ctx.incremental_responses))
-
-    return response
-
-
-class AudioStreamTrack(MediaStreamTrack):
-    """
-    An audio stream track.
-    """
-
-    kind = "audio"
-
-    def __init__(self, ctx: TranscriptionContext, track):
-        super().__init__()
-        self.ctx = ctx
-        self.track = track
-        self.audio_buffer = av.AudioFifo()
-
-    async def recv(self) -> av.audio.frame.AudioFrame:
-        ctx = self.ctx
-        frame = await self.track.recv()
-        self.audio_buffer.write(frame)
-
-        if local_frames := self.audio_buffer.read_many(
-            settings.AUDIO_BUFFER_SIZE, partial=False
-        ):
-            whisper_result = run_in_executor(
-                get_transcription,
-                ctx,
-                TranscriptionInput(local_frames),
-                executor=executor,
-            )
-            whisper_result.add_done_callback(
-                lambda f: channel_send_transcript(ctx) if f.result() else None
-            )
-
-        if len(ctx.transcription_text) > 25:
-            llm_input_text = ctx.transcription_text
-            ctx.transcription_text = ""
-            param = TitleSummaryInput(
-                input_text=llm_input_text, transcribed_time=ctx.last_transcribed_time
-            )
-            llm_result = run_in_executor(
-                get_title_and_summary, ctx, param, executor=executor
-            )
-            llm_result.add_done_callback(
-                lambda f: channel_send_increment(ctx.data_channel, llm_result.result())
-                if f.result()
-                else None
-            )
-        return frame
-
-
-async def offer(request: requests.Request) -> web.Response:
-    """
-    Establish the WebRTC connection with the client
-    :param request:
-    :return:
-    """
-    params = await request.json()
-    offer = RTCSessionDescription(sdp=params["sdp"], type=params["type"])
-
-    # client identification
-    peername = request.transport.get_extra_info("peername")
-    if peername is not None:
-        clientid = f"{peername[0]}:{peername[1]}"
-    else:
-        clientid = uuid.uuid4()
-
-    # create a context for the whole rtc transaction
-    # add a customised logger to the context
-    ctx = TranscriptionContext(logger=logger.bind(client=clientid))
-
-    # handle RTC peer connection
-    pc = RTCPeerConnection()
-    pcs.add(pc)
-
-    @pc.on("datachannel")
-    def on_datachannel(channel) -> NoReturn:
-        ctx.data_channel = channel
-        ctx.logger = ctx.logger.bind(channel=channel.label)
-        ctx.logger.info("Channel created by remote party")
-
-        @channel.on("message")
-        def on_message(message: str) -> NoReturn:
-            ctx.logger.info(f"Message: {message}")
-            if json.loads(message)["cmd"] == "STOP":
-                # Placeholder final summary
-                response = get_final_summary_response()
-                channel_send_increment(channel, response)
-                # To-do Add code to stop connection from server side here
-                # But have to handshake with client once
-
-            if isinstance(message, str) and message.startswith("ping"):
-                channel_send(channel, "pong" + message[4:])
-
-    @pc.on("connectionstatechange")
-    async def on_connectionstatechange() -> NoReturn:
-        ctx.logger.info(f"Connection state changed: {pc.connectionState}")
-        if pc.connectionState == "failed":
-            await pc.close()
-            pcs.discard(pc)
-
-    @pc.on("track")
-    def on_track(track) -> NoReturn:
-        ctx.logger.info(f"Track {track.kind} received")
-        pc.addTrack(AudioStreamTrack(ctx, relay.subscribe(track)))
-
-    await pc.setRemoteDescription(offer)
-
-    answer = await pc.createAnswer()
-    await pc.setLocalDescription(answer)
-    return web.Response(
-        content_type="application/json",
-        text=json.dumps(
-            {"sdp": pc.localDescription.sdp, "type": pc.localDescription.type}
-        ),
-    )
-
-
-async def on_shutdown(application: web.Application) -> NoReturn:
-    """
-    On shutdown, the coroutines that shutdown client connections are
-    executed
-    :param application:
-    :return:
-    """
-    coroutines = [pc.close() for pc in pcs]
-    await asyncio.gather(*coroutines)
-    pcs.clear()
-
-
-def create_app() -> web.Application:
-    """
-    Create the web application
-    """
-    app = web.Application()
-    cors = aiohttp_cors.setup(
-        app,
-        defaults={
-            "*": aiohttp_cors.ResourceOptions(
-                allow_credentials=True, expose_headers="*", allow_headers="*"
-            )
-        },
-    )
-
-    offer_resource = cors.add(app.router.add_resource("/offer"))
-    cors.add(offer_resource.add_route("POST", offer))
-    app.on_shutdown.append(on_shutdown)
-    return app
-
-
-if __name__ == "__main__":
-    parser = argparse.ArgumentParser(description="WebRTC based server for Reflector")
-    parser.add_argument(
-        "--host", default="0.0.0.0", help="Server host IP (def: 0.0.0.0)"
-    )
-    parser.add_argument(
-        "--port", type=int, default=1250, help="Server port (def: 1250)"
-    )
-    args = parser.parse_args()
-    app = create_app()
-    web.run_app(app, access_log=None, host=args.host, port=args.port)
--- a/server/reflector/settings.py
+++ b/server/reflector/settings.py
@@ -6,6 +6,12 @@ class Settings(BaseSettings):

    OPENMP_KMP_DUPLICATE_LIB_OK: bool = False

+    # Database
+    DATABASE_URL: str = "sqlite:///./reflector.sqlite3"
+
+    # local data directory (audio for no)
+    DATA_DIR: str = "./data"
+
    # Whisper
    WHISPER_MODEL_SIZE: str = "tiny"
    WHISPER_REAL_TIME_MODEL_SIZE: str = "tiny"
@@ -27,7 +33,7 @@ class Settings(BaseSettings):
    AUDIO_BUFFER_SIZE: int = 256 * 960

    # Audio Transcription
-    # backends: whisper, banana
+    # backends: whisper, banana, modal
    TRANSCRIPT_BACKEND: str = "whisper"
    TRANSCRIPT_URL: str | None = None
    TRANSCRIPT_TIMEOUT: int = 90
@@ -49,6 +55,7 @@ class Settings(BaseSettings):
    TRANSCRIPT_STORAGE_AWS_SECRET_ACCESS_KEY: str | None = None

    # LLM
+    # available backend: openai, banana, modal, oobagooda
    LLM_BACKEND: str = "oobagooda"

    # LLM common configuration
--- a/server/reflector/views/rtc_offer.py
+++ b/server/reflector/views/rtc_offer.py
@@ -2,17 +2,18 @@ import asyncio
 from fastapi import Request, APIRouter
 from reflector.events import subscribers_shutdown
 from pydantic import BaseModel
-from reflector.models import TranscriptionContext
 from reflector.logger import logger
 from aiortc import RTCPeerConnection, RTCSessionDescription, MediaStreamTrack
 from json import loads, dumps
 from enum import StrEnum
+from pathlib import Path
 import av
 from reflector.processors import (
    Pipeline,
    AudioChunkerProcessor,
    AudioMergeProcessor,
    AudioTranscriptAutoProcessor,
+    AudioFileWriterProcessor,
    TranscriptLinerProcessor,
    TranscriptTopicDetectorProcessor,
    TranscriptFinalSummaryProcessor,
@@ -25,6 +26,15 @@ sessions = []
 router = APIRouter()


+class TranscriptionContext(object):
+    def __init__(self, logger):
+        self.logger = logger
+        self.pipeline = None
+        self.data_channel = None
+        self.status = "idle"
+        self.topics = []
+
+
 class AudioStreamTrack(MediaStreamTrack):
    """
    An audio stream track.
@@ -64,7 +74,11 @@ class PipelineEvent(StrEnum):


 async def rtc_offer_base(
-    params: RtcOffer, request: Request, event_callback=None, event_callback_args=None
+    params: RtcOffer,
+    request: Request,
+    event_callback=None,
+    event_callback_args=None,
+    audio_filename: Path | None = None,
 ):
    # build an rtc session
    offer = RTCSessionDescription(sdp=params.sdp, type=params.type)
@@ -73,7 +87,6 @@ async def rtc_offer_base(
    peername = request.client
    clientid = f"{peername[0]}:{peername[1]}"
    ctx = TranscriptionContext(logger=logger.bind(client=clientid))
-    ctx.topics = []

    async def update_status(status: str):
        changed = ctx.status != status
@@ -151,14 +164,18 @@ async def rtc_offer_base(

    # create a context for the whole rtc transaction
    # add a customised logger to the context
-    ctx.pipeline = Pipeline(
+    processors = []
+    if audio_filename is not None:
+        processors += [AudioFileWriterProcessor(path=audio_filename)]
+    processors += [
        AudioChunkerProcessor(),
        AudioMergeProcessor(),
        AudioTranscriptAutoProcessor.as_threaded(callback=on_transcript),
        TranscriptLinerProcessor(),
        TranscriptTopicDetectorProcessor.as_threaded(callback=on_topic),
        TranscriptFinalSummaryProcessor.as_threaded(callback=on_final_summary),
-    )
+    ]
+    ctx.pipeline = Pipeline(*processors)
    # FIXME: warmup is not working well yet
    # await ctx.pipeline.warmup()

--- a/server/reflector/views/transcripts.py
+++ b/server/reflector/views/transcripts.py
@@ -5,13 +5,20 @@ from fastapi import (
    WebSocket,
    WebSocketDisconnect,
 )
+from fastapi.responses import FileResponse
+from starlette.concurrency import run_in_threadpool
 from pydantic import BaseModel, Field
-from uuid import UUID, uuid4
+from uuid import uuid4
 from datetime import datetime
 from fastapi_pagination import Page, paginate
 from reflector.logger import logger
+from reflector.db import database, transcripts
+from reflector.settings import settings
 from .rtc_offer import rtc_offer_base, RtcOffer, PipelineEvent
 from typing import Optional
+from pathlib import Path
+from tempfile import NamedTemporaryFile
+import av


 router = APIRouter()
@@ -21,6 +28,10 @@ router = APIRouter()
 # ==============================================================


+def generate_uuid4():
+    return str(uuid4())
+
+
 def generate_transcript_name():
    now = datetime.utcnow()
    return f"Transcript {now.strftime('%Y-%m-%d %H:%M:%S')}"
@@ -31,7 +42,7 @@ class TranscriptText(BaseModel):


 class TranscriptTopic(BaseModel):
-    id: UUID = Field(default_factory=uuid4)
+    id: str = Field(default_factory=generate_uuid4)
    title: str
    summary: str
    transcript: str
@@ -48,7 +59,7 @@ class TranscriptEvent(BaseModel):


 class Transcript(BaseModel):
-    id: UUID = Field(default_factory=uuid4)
+    id: str = Field(default_factory=generate_uuid4)
    name: str = Field(default_factory=generate_transcript_name)
    status: str = "idle"
    locked: bool = False
@@ -70,21 +81,87 @@ class Transcript(BaseModel):
        else:
            self.topics.append(topic)

+    def events_dump(self, mode="json"):
+        return [event.model_dump(mode=mode) for event in self.events]
+
+    def topics_dump(self, mode="json"):
+        return [topic.model_dump(mode=mode) for topic in self.topics]
+
+    def convert_audio_to_mp3(self):
+        fn = self.audio_mp3_filename
+        if fn.exists():
+            return
+
+        logger.info(f"Converting audio to mp3: {self.audio_filename}")
+        inp = av.open(self.audio_filename.as_posix(), "r")
+
+        # create temporary file for mp3
+        with NamedTemporaryFile(suffix=".mp3", delete=False) as tmp:
+            out = av.open(tmp.name, "w")
+            stream = out.add_stream("mp3")
+            for frame in inp.decode(audio=0):
+                frame.pts = None
+                for packet in stream.encode(frame):
+                    out.mux(packet)
+            for packet in stream.encode(None):
+                out.mux(packet)
+            out.close()
+
+            # move temporary file to final location
+            Path(tmp.name).rename(fn)
+
+    def unlink(self):
+        self.data_path.unlink(missing_ok=True)
+
+    @property
+    def data_path(self):
+        return Path(settings.DATA_DIR) / self.id
+
+    @property
+    def audio_filename(self):
+        return self.data_path / "audio.wav"
+
+    @property
+    def audio_mp3_filename(self):
+        return self.data_path / "audio.mp3"
+

 class TranscriptController:
-    transcripts: list[Transcript] = []
+    async def get_all(self) -> list[Transcript]:
+        query = transcripts.select()
+        results = await database.fetch_all(query)
+        return results

-    def get_all(self) -> list[Transcript]:
-        return self.transcripts
+    async def get_by_id(self, transcript_id: str) -> Transcript | None:
+        query = transcripts.select().where(transcripts.c.id == transcript_id)
+        result = await database.fetch_one(query)
+        if not result:
+            return None
+        return Transcript(**result)

-    def get_by_id(self, transcript_id: UUID) -> Transcript | None:
-        return next((t for t in self.transcripts if t.id == transcript_id), None)
+    async def add(self, name: str):
+        transcript = Transcript(name=name)
+        query = transcripts.insert().values(**transcript.model_dump())
+        await database.execute(query)
+        return transcript

-    def add(self, transcript: Transcript):
-        self.transcripts.append(transcript)
+    async def update(self, transcript: Transcript, values: dict):
+        query = (
+            transcripts.update()
+            .where(transcripts.c.id == transcript.id)
+            .values(**values)
+        )
+        await database.execute(query)
+        for key, value in values.items():
+            setattr(transcript, key, value)

-    def remove(self, transcript: Transcript):
-        self.transcripts.remove(transcript)
+    async def remove_by_id(self, transcript_id: str) -> None:
+        transcript = await self.get_by_id(transcript_id)
+        if not transcript:
+            return
+        transcript.unlink()
+        query = transcripts.delete().where(transcripts.c.id == transcript_id)
+        await database.execute(query)


 transcripts_controller = TranscriptController()
@@ -96,7 +173,7 @@ transcripts_controller = TranscriptController()


 class GetTranscript(BaseModel):
-    id: UUID
+    id: str
    name: str
    status: str
    locked: bool
@@ -123,15 +200,12 @@ class DeletionStatus(BaseModel):

@router.get("/transcripts", response_model=Page[GetTranscript])
 async def transcripts_list():
-    return paginate(transcripts_controller.get_all())
+    return paginate(await transcripts_controller.get_all())


@router.post("/transcripts", response_model=GetTranscript)
 async def transcripts_create(info: CreateTranscript):
-    transcript = Transcript()
-    transcript.name = info.name
-    transcripts_controller.add(transcript)
-    return transcript
+    return await transcripts_controller.add(info.name)


 # ==============================================================
@@ -140,54 +214,72 @@ async def transcripts_create(info: CreateTranscript):


@router.get("/transcripts/{transcript_id}", response_model=GetTranscript)
-async def transcript_get(transcript_id: UUID):
-    transcript = transcripts_controller.get_by_id(transcript_id)
+async def transcript_get(transcript_id: str):
+    transcript = await transcripts_controller.get_by_id(transcript_id)
    if not transcript:
        raise HTTPException(status_code=404, detail="Transcript not found")
    return transcript


@router.patch("/transcripts/{transcript_id}", response_model=GetTranscript)
-async def transcript_update(transcript_id: UUID, info: UpdateTranscript):
-    transcript = transcripts_controller.get_by_id(transcript_id)
+async def transcript_update(transcript_id: str, info: UpdateTranscript):
+    transcript = await transcripts_controller.get_by_id(transcript_id)
    if not transcript:
        raise HTTPException(status_code=404, detail="Transcript not found")
+    values = {}
    if info.name is not None:
-        transcript.name = info.name
+        values["name"] = info.name
    if info.locked is not None:
-        transcript.locked = info.locked
+        values["locked"] = info.locked
+    await transcripts_controller.update(transcript, values)
    return transcript


@router.delete("/transcripts/{transcript_id}", response_model=DeletionStatus)
-async def transcript_delete(transcript_id: UUID):
-    transcript = transcripts_controller.get_by_id(transcript_id)
+async def transcript_delete(transcript_id: str):
+    transcript = await transcripts_controller.get_by_id(transcript_id)
    if not transcript:
        raise HTTPException(status_code=404, detail="Transcript not found")
-    transcripts_controller.remove(transcript)
+    await transcripts_controller.remove_by_id(transcript.id)
    return DeletionStatus(status="ok")


@router.get("/transcripts/{transcript_id}/audio")
-async def transcript_get_audio(transcript_id: UUID):
-    transcript = transcripts_controller.get_by_id(transcript_id)
+async def transcript_get_audio(transcript_id: str):
+    transcript = await transcripts_controller.get_by_id(transcript_id)
    if not transcript:
        raise HTTPException(status_code=404, detail="Transcript not found")

-    # TODO: Implement audio generation
-    return HTTPException(status_code=500, detail="Not implemented")
+    if not transcript.audio_filename.exists():
+        raise HTTPException(status_code=404, detail="Audio not found")
+
+    return FileResponse(transcript.audio_filename, media_type="audio/wav")
+
+
+@router.get("/transcripts/{transcript_id}/audio/mp3")
+async def transcript_get_audio_mp3(transcript_id: str):
+    transcript = await transcripts_controller.get_by_id(transcript_id)
+    if not transcript:
+        raise HTTPException(status_code=404, detail="Transcript not found")
+
+    if not transcript.audio_filename.exists():
+        raise HTTPException(status_code=404, detail="Audio not found")
+
+    await run_in_threadpool(transcript.convert_audio_to_mp3)
+
+    return FileResponse(transcript.audio_mp3_filename, media_type="audio/mp3")


@router.get("/transcripts/{transcript_id}/topics", response_model=list[TranscriptTopic])
-async def transcript_get_topics(transcript_id: UUID):
-    transcript = transcripts_controller.get_by_id(transcript_id)
+async def transcript_get_topics(transcript_id: str):
+    transcript = await transcripts_controller.get_by_id(transcript_id)
    if not transcript:
        raise HTTPException(status_code=404, detail="Transcript not found")
    return transcript.topics


@router.get("/transcripts/{transcript_id}/events")
-async def transcript_get_websocket_events(transcript_id: UUID):
+async def transcript_get_websocket_events(transcript_id: str):
    pass


@@ -200,20 +292,20 @@ class WebsocketManager:
    def __init__(self):
        self.active_connections = {}

-    async def connect(self, transcript_id: UUID, websocket: WebSocket):
+    async def connect(self, transcript_id: str, websocket: WebSocket):
        await websocket.accept()
        if transcript_id not in self.active_connections:
            self.active_connections[transcript_id] = []
        self.active_connections[transcript_id].append(websocket)

-    def disconnect(self, transcript_id: UUID, websocket: WebSocket):
+    def disconnect(self, transcript_id: str, websocket: WebSocket):
        if transcript_id not in self.active_connections:
            return
        self.active_connections[transcript_id].remove(websocket)
        if not self.active_connections[transcript_id]:
            del self.active_connections[transcript_id]

-    async def send_json(self, transcript_id: UUID, message):
+    async def send_json(self, transcript_id: str, message):
        if transcript_id not in self.active_connections:
            return
        for connection in self.active_connections[transcript_id][:]:
@@ -227,8 +319,8 @@ ws_manager = WebsocketManager()


@router.websocket("/transcripts/{transcript_id}/events")
-async def transcript_events_websocket(transcript_id: UUID, websocket: WebSocket):
-    transcript = transcripts_controller.get_by_id(transcript_id)
+async def transcript_events_websocket(transcript_id: str, websocket: WebSocket):
+    transcript = await transcripts_controller.get_by_id(transcript_id)
    if not transcript:
        raise HTTPException(status_code=404, detail="Transcript not found")

@@ -260,7 +352,7 @@ async def handle_rtc_event(event: PipelineEvent, args, data):
    # transcript from the database for each event.
    # print(f"Event: {event}", args, data)
    transcript_id = args
-    transcript = transcripts_controller.get_by_id(transcript_id)
+    transcript = await transcripts_controller.get_by_id(transcript_id)
    if not transcript:
        return

@@ -272,6 +364,12 @@ async def handle_rtc_event(event: PipelineEvent, args, data):
    # FIXME don't do copy
    if event == PipelineEvent.TRANSCRIPT:
        resp = transcript.add_event(event=event, data=TranscriptText(text=data.text))
+        await transcripts_controller.update(
+            transcript,
+            {
+                "events": transcript.events_dump(),
+            },
+        )

    elif event == PipelineEvent.TOPIC:
        topic = TranscriptTopic(
@@ -283,14 +381,34 @@ async def handle_rtc_event(event: PipelineEvent, args, data):
        resp = transcript.add_event(event=event, data=topic)
        transcript.upsert_topic(topic)

+        await transcripts_controller.update(
+            transcript,
+            {
+                "events": transcript.events_dump(),
+                "topics": transcript.topics_dump(),
+            },
+        )
+
    elif event == PipelineEvent.FINAL_SUMMARY:
        final_summary = TranscriptFinalSummary(summary=data.summary)
        resp = transcript.add_event(event=event, data=final_summary)
-        transcript.summary = final_summary
+        await transcripts_controller.update(
+            transcript,
+            {
+                "events": transcript.events_dump(),
+                "summary": final_summary.summary,
+            },
+        )

    elif event == PipelineEvent.STATUS:
        resp = transcript.add_event(event=event, data=data)
-        transcript.status = data.value
+        await transcripts_controller.update(
+            transcript,
+            {
+                "events": transcript.events_dump(),
+                "status": data.value,
+            },
+        )

    else:
        logger.warning(f"Unknown event: {event}")
@@ -302,9 +420,9 @@ async def handle_rtc_event(event: PipelineEvent, args, data):

@router.post("/transcripts/{transcript_id}/record/webrtc")
 async def transcript_record_webrtc(
-    transcript_id: UUID, params: RtcOffer, request: Request
+    transcript_id: str, params: RtcOffer, request: Request
 ):
-    transcript = transcripts_controller.get_by_id(transcript_id)
+    transcript = await transcripts_controller.get_by_id(transcript_id)
    if not transcript:
        raise HTTPException(status_code=404, detail="Transcript not found")

@@ -317,4 +435,5 @@ async def transcript_record_webrtc(
        request,
        event_callback=handle_rtc_event,
        event_callback_args=transcript_id,
+        audio_filename=transcript.audio_filename,
    )
--- a/server/test.db
+++ b/server/test.db
--- a/server/tests/test_basic_rtc.py
+++ b/server/tests/test_basic_rtc.py
@@ -1,63 +0,0 @@
-import pytest
-from unittest.mock import patch
-
-
-@pytest.mark.asyncio
-async def test_basic_rtc_server(aiohttp_server, event_loop):
-    # goal is to start the server, and send rtc audio to it
-    # validate the events received
-    import argparse
-    import json
-    from pathlib import Path
-    from reflector.server import create_app
-    from reflector.stream_client import StreamClient
-    from reflector.models import TitleSummaryOutput
-    from aiortc.contrib.signaling import add_signaling_arguments, create_signaling
-
-    # customize settings to have a mock LLM server
-    with patch("reflector.server.get_title_and_summary") as mock_llm:
-        # any response from mock_llm will be test topic
-        mock_llm.return_value = TitleSummaryOutput(["topic_test"])
-
-        # create the server
-        app = create_app()
-        server = await aiohttp_server(app)
-        url = f"http://{server.host}:{server.port}/offer"
-
-        # create signaling
-        parser = argparse.ArgumentParser()
-        add_signaling_arguments(parser)
-        args = parser.parse_args(["-s", "tcp-socket"])
-        signaling = create_signaling(args)
-
-        # create the client
-        path = Path(__file__).parent / "records" / "test_mathieu_hello.wav"
-        client = StreamClient(signaling, url=url, play_from=path.as_posix())
-        await client.start()
-
-        # we just want the first transcription
-        # and topic update messages
-
-        marks = {
-            "SHOW_TRANSCRIPTION": False,
-            "UPDATE_TOPICS": False,
-        }
-
-        async for rawmsg in client.get_reader():
-            msg = json.loads(rawmsg)
-            cmd = msg["cmd"]
-            if cmd == "SHOW_TRANSCRIPTION":
-                assert "text" in msg
-                assert "want to share my incredible experience" in msg["text"]
-            elif cmd == "UPDATE_TOPICS":
-                assert "topics" in msg
-                assert "topic_test" in msg["topics"]
-            marks[cmd] = True
-
-            # break if we have all the events we need
-            if all(marks.values()):
-                break
-
-        # stop the server
-        await server.close()
-        await client.stop()
--- a/server/tests/test_transcripts_rtc_ws.py
+++ b/server/tests/test_transcripts_rtc_ws.py
@@ -70,11 +70,15 @@ async def dummy_llm():


@pytest.mark.asyncio
-async def test_transcript_rtc_and_websocket(dummy_transcript, dummy_llm):
+async def test_transcript_rtc_and_websocket(tmpdir, dummy_transcript, dummy_llm):
    # goal: start the server, exchange RTC, receive websocket events
    # because of that, we need to start the server in a thread
    # to be able to connect with aiortc

+    from reflector.settings import settings
+
+    settings.DATA_DIR = Path(tmpdir)
+
    # start server
    host = "127.0.0.1"
    port = 1255
@@ -188,3 +192,13 @@ async def test_transcript_rtc_and_websocket(dummy_transcript, dummy_llm):
    resp = await ac.get(f"/transcripts/{tid}")
    assert resp.status_code == 200
    assert resp.json()["status"] == "ended"
+
+    # check that audio is available
+    resp = await ac.get(f"/transcripts/{tid}/audio")
+    assert resp.status_code == 200
+    assert resp.headers["Content-Type"] == "audio/wav"
+
+    # check that audio/mp3 is available
+    resp = await ac.get(f"/transcripts/{tid}/audio/mp3")
+    assert resp.status_code == 200
+    assert resp.headers["Content-Type"] == "audio/mp3"
--- a/www/app/layout.js
+++ b/www/app/layout.js
@@ -1,4 +1,4 @@
-import "./globals.scss";
+import "./styles/globals.scss";
 import { Roboto } from "next/font/google";

 import Head from "next/head";
--- a/www/app/lib/random.js
+++ b/www/app/lib/random.js
@@ -17,15 +17,3 @@ export function Mulberry32(seed) {
    return ((t ^ (t >>> 14)) >>> 0) / 4294967296;
  };
 }
-
-export const formatTime = (seconds) => {
-  let hours = Math.floor(seconds / 3600);
-  let minutes = Math.floor((seconds % 3600) / 60);
-  let secs = Math.floor(seconds % 60);
-
-  let timeString = `${hours > 0 ? hours + ":" : ""}${minutes
-    .toString()
-    .padStart(2, "0")}:${secs.toString().padStart(2, "0")}`;
-
-  return timeString;
-};
--- a/www/app/lib/time.js
+++ b/www/app/lib/time.js
@@ -0,0 +1,11 @@
+export const formatTime = (seconds) => {
+  let hours = Math.floor(seconds / 3600);
+  let minutes = Math.floor((seconds % 3600) / 60);
+  let secs = Math.floor(seconds % 60);
+
+  let timeString = `${hours > 0 ? hours + ":" : ""}${minutes
+    .toString()
+    .padStart(2, "0")}:${secs.toString().padStart(2, "0")}`;
+
+  return timeString;
+};
--- a/www/app/page.js
+++ b/www/app/page.js
@@ -1,57 +1,4 @@
-"use client";
-import React, { useEffect, useState } from "react";
-import Recorder from "./components/record.js";
-import { Dashboard } from "./components/dashboard.js";
-import useWebRTC from "./components/webrtc.js";
-import useTranscript from "./components/transcript.js";
-import { useWebSockets } from "./components/websocket.js";
-import "../public/button.css";
-
-const App = () => {
-  const [stream, setStream] = useState(null);
-  const [disconnected, setDisconnected] = useState(false);
-  const useActiveTopic = useState(null);
-
-  useEffect(() => {
-    if (process.env.NEXT_PUBLIC_ENV === "development") {
-      document.onkeyup = (e) => {
-        if (e.key === "d") {
-          setDisconnected((prev) => !prev);
-        }
-      };
-    }
-  }, []);
-
-  const transcript = useTranscript();
-  const webRTC = useWebRTC(stream, transcript.response?.id);
-  const webSockets = useWebSockets(transcript.response?.id);
-
-  return (
-    <div className="flex flex-col items-center h-[100svh] bg-gradient-to-r from-[#8ec5fc30] to-[#e0c3fc42]">
-      <div className="h-[13svh] flex flex-col justify-center items-center">
-        <h1 className="text-5xl font-bold text-blue-500">Reflector</h1>
-        <p className="text-gray-500">Capture The Signal, Not The Noise</p>
-      </div>
-
-      <Recorder
-        setStream={setStream}
-        onStop={() => {
-          webRTC?.peer?.send(JSON.stringify({ cmd: "STOP" }));
-          setStream(null);
-        }}
-        topics={webSockets.topics}
-        useActiveTopic={useActiveTopic}
-      />
-      <Dashboard
-        transcriptionText={webSockets.transcriptText}
-        finalSummary={webSockets.finalSummary}
-        topics={webSockets.topics}
-        stream={stream}
-        disconnected={disconnected}
-        useActiveTopic={useActiveTopic}
-      />
-    </div>
-  );
-};
-
-export default App;
+import { redirect } from "next/navigation";
+export default async function Index({ params }) {
+  redirect("/transcripts/new");
+}
--- a/www/app/reach.png
+++ b/www/app/reach.png
--- a/www/app/styles/button.css
+++ b/www/app/styles/button.css
--- a/www/app/styles/globals.scss
+++ b/www/app/styles/globals.scss
--- a/www/app/transcripts/CustomRecordPlugin.js
+++ b/www/app/transcripts/CustomRecordPlugin.js
--- a/www/app/transcripts/dashboard.js
+++ b/www/app/transcripts/dashboard.js
--- a/www/app/transcripts/new/page.js
+++ b/www/app/transcripts/new/page.js
@@ -0,0 +1,56 @@
+"use client";
+import React, { useEffect, useState } from "react";
+import Recorder from "../recorder";
+import { Dashboard } from "../dashboard";
+import useWebRTC from "../useWebRTC";
+import useTranscript from "../useTranscript";
+import { useWebSockets } from "../useWebSockets";
+import "../../styles/button.css";
+
+const App = () => {
+  const [stream, setStream] = useState(null);
+  const [disconnected, setDisconnected] = useState(false);
+
+  useEffect(() => {
+    if (process.env.NEXT_PUBLIC_ENV === "development") {
+      document.onkeyup = (e) => {
+        if (e.key === "d") {
+          setDisconnected((prev) => !prev);
+        }
+      };
+    }
+  }, []);
+
+  const transcript = useTranscript();
+  const webRTC = useWebRTC(stream, transcript.response?.id);
+  const webSockets = useWebSockets(transcript.response?.id);
+
+  return (
+    <div className="flex flex-col items-center h-[100svh] bg-gradient-to-r from-[#8ec5fc30] to-[#e0c3fc42]">
+      <div className="h-[13svh] flex flex-col justify-center items-center">
+        <h1 className="text-5xl font-bold text-blue-500">Reflector</h1>
+        <p className="text-gray-500">Capture The Signal, Not The Noise</p>
+      </div>
+
+      <Recorder
+        setStream={setStream}
+        onStop={() => {
+          webRTC?.peer?.send(JSON.stringify({ cmd: "STOP" }));
+          setStream(null);
+        }}
+      />
+
+      <hr />
+
+      <Dashboard
+        transcriptionText={webSockets.transcriptText}
+        finalSummary={webSockets.finalSummary}
+        topics={webSockets.topics}
+        stream={stream}
+        disconnected={disconnected}
+      />
+    </div>
+  );
+};
+
+export default App;
--- a/www/app/transcripts/recorder.js
+++ b/www/app/transcripts/recorder.js
@@ -10,7 +10,7 @@ import Dropdown from "react-dropdown";
 import "react-dropdown/style.css";

 import CustomRecordPlugin from "./CustomRecordPlugin";
-import { formatTime } from "../utils";
+import { formatTime } from "../lib/time";

 const AudioInputsDropdown = (props) => {
  const [ddOptions, setDdOptions] = useState([]);
--- a/www/app/transcripts/useTranscript.js
+++ b/www/app/transcripts/useTranscript.js
--- a/www/app/transcripts/useWebRTC.js
+++ b/www/app/transcripts/useWebRTC.js
--- a/www/app/transcripts/useWebSockets.js
+++ b/www/app/transcripts/useWebSockets.js
--- a/www/pages/api/sentry-example-api.js
+++ b/www/pages/api/sentry-example-api.js
@@ -1,5 +0,0 @@
-// A faulty API route to test Sentry's error monitoring
-export default function handler(_req, res) {
-  throw new Error("Sentry Example API Route Error");
-  res.status(200).json({ name: "John Doe" });
-}
--- a/www/pages/sentry-example-page.js
+++ b/www/pages/sentry-example-page.js
@@ -1,87 +0,0 @@
-import Head from "next/head";
-import * as Sentry from "@sentry/nextjs";
-
-export default function Home() {
-  return (
-    <div>
-      <Head>
-        <title>Sentry Onboarding</title>
-        <meta name="description" content="Test Sentry for your Next.js app!" />
-      </Head>
-
-      <main
-        style={{
-          minHeight: "100vh",
-          display: "flex",
-          flexDirection: "column",
-          justifyContent: "center",
-          alignItems: "center",
-        }}
-      >
-        <h1 style={{ fontSize: "4rem", margin: "14px 0" }}>
-          <svg
-            style={{
-              height: "1em",
-            }}
-            xmlns="http://www.w3.org/2000/svg"
-            viewBox="0 0 200 44"
-          >
-            <path
-              fill="currentColor"
-              d="M124.32,28.28,109.56,9.22h-3.68V34.77h3.73V15.19l15.18,19.58h3.26V9.22h-3.73ZM87.15,23.54h13.23V20.22H87.14V12.53h14.93V9.21H83.34V34.77h18.92V31.45H87.14ZM71.59,20.3h0C66.44,19.06,65,18.08,65,15.7c0-2.14,1.89-3.59,4.71-3.59a12.06,12.06,0,0,1,7.07,2.55l2-2.83a14.1,14.1,0,0,0-9-3c-5.06,0-8.59,3-8.59,7.27,0,4.6,3,6.19,8.46,7.52C74.51,24.74,76,25.78,76,28.11s-2,3.77-5.09,3.77a12.34,12.34,0,0,1-8.3-3.26l-2.25,2.69a15.94,15.94,0,0,0,10.42,3.85c5.48,0,9-2.95,9-7.51C79.75,23.79,77.47,21.72,71.59,20.3ZM195.7,9.22l-7.69,12-7.64-12h-4.46L186,24.67V34.78h3.84V24.55L200,9.22Zm-64.63,3.46h8.37v22.1h3.84V12.68h8.37V9.22H131.08ZM169.41,24.8c3.86-1.07,6-3.77,6-7.63,0-4.91-3.59-8-9.38-8H154.67V34.76h3.8V25.58h6.45l6.48,9.2h4.44l-7-9.82Zm-10.95-2.5V12.6h7.17c3.74,0,5.88,1.77,5.88,4.84s-2.29,4.86-5.84,4.86Z M29,2.26a4.67,4.67,0,0,0-8,0L14.42,13.53A32.21,32.21,0,0,1,32.17,40.19H27.55A27.68,27.68,0,0,0,12.09,17.47L6,28a15.92,15.92,0,0,1,9.23,12.17H4.62A.76.76,0,0,1,4,39.06l2.94-5a10.74,10.74,0,0,0-3.36-1.9l-2.91,5a4.54,4.54,0,0,0,1.69,6.24A4.66,4.66,0,0,0,4.62,44H19.15a19.4,19.4,0,0,0-8-17.31l2.31-4A23.87,23.87,0,0,1,23.76,44H36.07a35.88,35.88,0,0,0-16.41-31.8l4.67-8a.77.77,0,0,1,1.05-.27c.53.29,20.29,34.77,20.66,35.17a.76.76,0,0,1-.68,1.13H40.6q.09,1.91,0,3.81h4.78A4.59,4.59,0,0,0,50,39.43a4.49,4.49,0,0,0-.62-2.28Z"
-            ></path>
-          </svg>
-        </h1>
-
-        <p>Get started by sending us a sample error:</p>
-        <button
-          type="button"
-          style={{
-            padding: "12px",
-            cursor: "pointer",
-            backgroundColor: "#AD6CAA",
-            borderRadius: "4px",
-            border: "none",
-            color: "white",
-            fontSize: "14px",
-            margin: "18px",
-          }}
-          onClick={async () => {
-            const transaction = Sentry.startTransaction({
-              name: "Example Frontend Transaction",
-            });
-
-            Sentry.configureScope((scope) => {
-              scope.setSpan(transaction);
-            });
-
-            try {
-              const res = await fetch("/api/sentry-example-api");
-              if (!res.ok) {
-                throw new Error("Sentry Example Frontend Error");
-              }
-            } finally {
-              transaction.finish();
-            }
-          }}
-        >
-          Throw error!
-        </button>
-
-        <p>
-          Next, look for the error on the{" "}
-          <a href="https://monadical.sentry.io/issues/?project=4505634666577920">
-            Issues Page
-          </a>
-          .
-        </p>
-        <p style={{ marginTop: "24px" }}>
-          For more information, see{" "}
-          <a href="https://docs.sentry.io/platforms/javascript/guides/nextjs/">
-            https://docs.sentry.io/platforms/javascript/guides/nextjs/
-          </a>
-        </p>
-      </main>
-    </div>
-  );
-}