CAIMAN-ASR server release bundle

Release name: myrtle-asr-server-<version>.run

This release bundle contains all the software needed to run the Myrtle.ai CAIMAN-ASR server in a production environment. This includes the server docker image, a simple Python client, and scripts to start and stop the server. Additionally, it contains a script to compile a hardware checkpoint into a CAIMAN-ASR checkpoint. Three model architectures are supported:

testing
base
large

Note

The testing config is not recommended for production use. See details here

The CAIMAN-ASR server supports two backends: CPU and FPGA. The CPU backend is not real time, but can be useful for testing on a machine without an Achronix Speedster7t PCIe card installed. The FPGA backend is able to support 2000 concurrent transcription streams per card with the base model and 800 with the large model.

Quick start: CPU backend

Load the CAIMAN-ASR server docker image:
```
docker load -i docker-asr-server.tgz
```

Start the CAIMAN-ASR server with the hardware checkpoint:

./start_asr_server.sh --rnnt-checkpoint compile-model-checkpoint/hardware_checkpoint.testing.example.pt --cpu-backend

Once the server prints "Starting server on port 3030", you can start the simple client. This will send a librispeech example wav to the CAIMAN-ASR server and print the transcription:
```
cd simple_client
./build.sh # only needed once to install dependencies
./run.sh
cd ..
```
To detach from the running docker container without killing it, use ctrl+p followed by ctrl+q.
Stop the CAIMAN-ASR server(s)
```
./kill_asr_servers.sh
```

Quick start: FPGA backend

If you are setting up the server from scratch you will need to flash the Achronix Speedster7t FPGA with the provided bitstream. If you have a demo system provided by Myrtle.ai or Achronix, the bitstream will already be flashed. See the Programming the card section for instructions on flashing the FPGA before continuing.

Note

Unlike with the CPU backend, you will need to compile the hardware checkpoint into a CAIMAN-ASR checkpoint (step 2 below). For more details on this process, see the Compiling weights section.

Load the CAIMAN-ASR server docker image
```
docker load -i docker-asr-server.tgz
```

Compile an example hardware checkpoint to a CAIMAN-ASR checkpoint

cd compile-model-checkpoint
./build_docker.sh
./run_docker.sh hardware_checkpoint.testing.example.pt caiman_asr_checkpoint.testing.example.pt
cd ..

Start the CAIMAN-ASR server with the CAIMAN-ASR checkpoint (use --card-id 1 to use the second card). --license-dir should point to the directory containing your license files. See the Licensing section for more information.
```
./start_asr_server.sh --rnnt-checkpoint compile-model-checkpoint/caiman_asr_checkpoint.testing.example.pt --license-dir "./licenses/" --card-id 0
```
To detach from the running docker container without killing it, use ctrl+p followed by ctrl+q.
Once the server prints "Starting server on port 3030", you can start the simple client. This will send a librispeech example wav to the CAIMAN-ASR server and print the transcription:
```
cd simple_client
./build.sh  # only needed once to install dependencies
./run.sh
cd ..
```
Stop the CAIMAN-ASR server(s)
```
./kill_asr_servers.sh
```

Connecting to the websocket API

The websocket endpoint is at ws://localhost:3030. See Websocket API for full documentation of the websocket interface.

The code in simple_client/simple_client.py is a simple example of how to connect to the CAIMAN-ASR server using the websocket API. The code snippets below are taken from this file, and demonstrate how to connect to the server in Python.

Initially the client needs to open a websocket connection to the server.

ws = websocket.WebSocket()
ws.connect(
    "ws://localhost:3030/asr/v0.1/stream?content_type=audio/x-raw;format=S16LE;channels=1;rate=16000"
)

Then the client can send audio data to the server.

for i in range(0, len(samples), samples_per_frame):
    payload = samples[i : i + samples_per_frame].tobytes()
    ws.send(payload, websocket.ABNF.OPCODE_BINARY)

The client can receive the server's response on the same websocket connection. Sending and receiving can be interleaved.

msg = ws.recv()
print(json.loads(msg)["alternatives"][0]["transcript"], end="", flush=True)

When the audio stream is finished the client should send a blank frame to the server to signal the end of the stream.

ws.send("", websocket.ABNF.OPCODE_BINARY)

The server will then send the final transcriptions and close the connection.

The server consumes audio in 60ms frames, so for optimal latency the client should send audio in 60ms frames. If the client sends audio in smaller chunks the server will wait for a complete frame before processing it. If the client sends audio in larger chunks there will be a latency penalty as the server waits for the next frame to arrive.

A more advanced client example in Rust is provided in caiman-asr-client; see Testing inference performance for more information.