remoteAudio

remoteAudio is an audio streaming application, built for Amateur Radio purposes, written in Go. The most typical use case for this software is the remote operation of an amateur radio station.

ADVICE: This project is under development. The parameters, ICD and the behaviour is still not stable and subject to change.

Supported Codecs

OPUS (default)
PCM (variable samplerate and 8/12/16/24 bit depth)

Supported Transportation Protocols

MQTT
P2P protocol (UDP/TCP/ZEROMQ) will be added later

Supported Platforms

RemoteAudio has been tested on the following platforms:

AMD64
i386
ARMv6
ARMv8

and the following operating Systems:

Linux (Ubuntu, Raspian)
MacOS (Sierra)

Download

You can download compiled binaries for MacOS, Linux (ARM/AMD64) from the releases page.

How build remoteAudio from source on Linux (Ubuntu > 12.04)

Dependencies

Make sure you have the latest Go version installed
Download and install the latest version of the Protocol buffers compiler
Install ````pkg-config, libsamplerate``` and ```libopus```

    $ sudo apt install pkg-config libsamplerate0 libsamplerate0-dev libopusfile-dev libopus-dev libportaudio2 portaudio19-dev

Install the gogo protocol buffers plugin

    $ go get github.com/gogo/protobuf/protoc-gen-gofast

Download and install rice command line tool

    $ go get github.com/GeertJohan/go.rice/rice

Download the remoteAudio sources and the packages it depends on

    $ go get -d github.com/dh1tw/remoteAudio

as an alternative you can build the protocol buffers compiler also from source. This repository contains a build script install-protobuf.sh

Build remoteAudio

    $ cd $GOPATH/src/github.com/dh1tw/remoteAudio
    $ make build

Install remoteAudio on your Systems

    $ make install

Requirements

In order to operate remoteAudio you need to either run your own MQTT Broker (Mosquitto is a good choice) or connect to a public broker, like iot.eclipse.org or test.mosquitto.org. The load of these brokers and their ping to your place obviously will influence the latency. These public brokers are good for inital tests, however they are sometimes overloaded.

Execute Audio Server

    $ remoteAudio server mqtt

Execute Audio Client

    $ remoteAudio client mqtt

List audio devices

If you are not sure about the name of your audio devices and their parameters, you can easily list that information:

    $ remoteAudio enumerate

Both, the server and the client provide extensive configuration possibilities, either through a configuration file (TOML|YAML|JSON), typically located in your home directory /home/your_user/.remoteAudio.toml. or through pflags.

An example configuration file is included in the repository.

All parameters can be set through pflags. The following example shows the options for $ remoteAudio server mqtt --help:


MQTT Server for bi-directional audio streaming

Usage:
  remoteAudio server mqtt [flags]

Flags:
  -p, --broker-port int     Broker Port (default 1883)
  -u, --broker-url string   Broker URL (default "localhost")
  -Y, --radio string        Radio ID (default "myradio")
  -X, --station string      Your station callsign (default "mystation")

Global Flags:
  -f, --audio-frame-length int             Amount of audio samples in one frame (default 480)
  -C, --codec string                       Audio codec (default "opus")
      --config string                      config file (default is $HOME/.remoteAudio.yaml)
      --input-device-channels string       Input Channels (default "mono")
      --input-device-latency duration      Input latency (default 5ms)
  -i, --input-device-name string           Input device (default "default")
      --input-device-samplingrate float    Input device sampling rate (default 48000)
      --opus-application string            profile for opus encoder (default "restricted_lowdelay")
      --opus-bitrate int                   Bitrate (bits/sec) generated by the opus encoder (default 32000)
      --opus-complexity int                Computational complexity of opus encoder (default 9)
      --opus-max-bandwidth string          maximum bandwidth of opus encoder (default "wideband")
      --output-device-channels string      Output Channels (default "stereo")
      --output-device-latency duration     Output latency (default 5ms)
  -o, --output-device-name string          Output device (default "default")
      --output-device-samplingrate float   Output device sampling rate (default 48000)
      --pcm-bitdepth int                   pcm audio bit depth (8, 12, 16, 24 bit) (default 16)
      --pcm-channels string                pcm audio Channels (default "stereo")
      --pcm-resampling-quality int         pcm resampling quality (default 1)
      --pcm-samplingrate float             pcm sampling rate (default 16000)
  -R, --rx-buffer-length int               Buffer length (in frames) for incoming Audio packets (default 10)
  -U, --user-id string                     Your User ID - required for TX

The Client provides a minimal Web Interface for basic control of the client and server side audio streams. Open a Webbrowser at: http://localhost:6060 to access the WebUI.

In any case, the client and server will accept almost any kind of audio frames without any configuration. Internally remoteAudio picks the right codec, resamples and adjust to the local audio output device.

However if the buffer size does not correspond to the internal buffer size, the stream has to be restarted which might result in a small delay.

Troubleshooting

remoteAudio does it's best to check if your sound hardware is compatible with the parameters you have set. However it's not entirely possible to check all the Settings.

Inexpensive (USB) Soundcards usually operate at 48kHz. They play the audio in Stereo and Record the Audio in MONO.

Feel free to open an issue if you encounter problems.

hb9fxq / remoteAudio