The plugin interface has been designed to mimic the hardware rack that is being used. There are three main sections of the interface – Top Panel, Sidebar, and Active Rack.
The top panel (A) contains overall controls and utilities for enabling the connection to the server. The input gain knob (1) is a general purpose gain at the input to the plugin. The local and remote buffer graphs (2) display the size over time of the sample buffers used to mitigate internet dropouts. Audio is flowing properly as long as these buffers have samples. The mix (3) knob is used to mix the dry audio passing through the plugin, with the resulting wet audio that has been processed in the hardware rack. The buffer setting (4) can be used to increase the sample buffers if dropouts become recurring or frequent. See the Buffer/Delay tab on this website for more information. The higher the value, the less chance that a dropout will occur, but the more latency that will be present between adjusting a front panel setting and subsequently hearing that change. The format (5) is used to set the audio codec that will be used for bi-directional audio across the internet. The uncompressed formats do not compromise the audio quality, but require a more stable connection between the client computer and the server. The compressed formats can be used if connection quality is not ideal, and the final audio capture can be done at full uncompressed resolution using the offline processing tool described later in this manual. The email (6) and connect button are used to establish the connection with the server. When not connected, the plugin will pass through audio unaltered. When connected, the audio will stream to the server. Finally the output gain (7) is another utility gain that can be used to adjust the output level of the plugin to properly match the input of the next item in the audio chain.
The sidebar (B) on the left side of the plugin lists all of the individual Access Analog equipment located at the server. After connecting to the server with the proper email in the top panel, this list will reflect the hardware units that are available to that specific email address at the current time. Hardware units at the server that are not reserved by the email address will be greyed out and not accessible for insertion into the active rack on the right. Note that all hardware units are separately listed, even if they are the same model, so that each can be used individually in the rack. Likewise, the web reservation for the email address applied to a specific hardware unit.
The active rack (C) on the right side of the plugin displays the active signal chain that the plugin is currently using. Audio flows from top to bottom. Hardware units can be dragged from the sidebar into the active rack at any time (8) – when playback is idle or active. We encourage you to manipulate the rack while audio is playing to better dial in the sound you are looking for. Drag operations are guided by green highlights to notify where the unit will be placed when dropped. Hardware units can be removed from the rack using the red close button located at the top of each unit. Hardware units can be dropped in below existing rack items, above, or in between. At this time, you should remove the unit from the rack and re-insert it if you want to change the order. [Coming soon: you will be able to drag within the rack to reorder units].
As mentioned before, audio flows from top to bottom. For each “row” within the rack, the incoming audio can either be processed by a single hardware unit, or a pair of units placed side-by-side. If you want to drag two units side-by-side, simply drop the second unit near the side of the first unit (the green highlight will guide you). The plugin will process mono and stereo audio through a variety of single and side-by-side hardware units. For most people, you can rely on the plugin to “do the right thing” for routing your audio and accommodating mono/stereo mismatches between the audio and the hardware units. If you are feeling adventurous and want more detail, see the examples in the tab on Audio Routing, but is is not necessary to know this detail in order to operate the plugin.
When equipment is added to the rack in a side-by-side configuration (9), one of the units will be expanded with the front panel showing, while the other is “minimized”. To view the other front panel, just click on the minimized unit.
Each item in the active rack represents an individual hardware unit. The operation of these controls within that unit are exactly the same as if you were using the physical unit directly. If you are already familiar with the hardware, you are all set. If you are not familiar with the hardware, you can read the manual online or on the Access Analog website to understand the controls. Each equipment panel has a red close button at the top left (10) to remove the unit from the active rack. Access Analog factory presets are available at the top right corner of the equipment panel (11). These presets will snap all physical hardware controls to predefined positions that can accomplish specific audio processing effects (a first for real analog audio hardware). Coming soon: all rack configurations and all equipment settings within the rack will be save-able in user presets using the host audio preset interface. Similarly, all controls will be automatable through the host.
The offline processor is used to process your audio in a non-real-time fashion. This is useful when you have a limited internet connection and you are forced to use a compressed audio format in order to get real time streaming working through the plugin. The offline processor is located at the bottom of the list of equipment in the sidebar. It is used like an actual hardware unit, but it is only a software processor and has no actual hardware counterpart. The offline processor is modeled after an audio recorder. It records audio locally, uploads it to the server uncompressed, triggers the server to process the audio, and finally downloads the uncompressed processed audio from the server.
The offline processor can be placed anywhere in the rack. It does nothing to the audio by just being in the rack, and it does not matter where in the rack it is inserted. In order for it to process the audio offline, you must interact with the controls. Offline processing is best covered as a sequence of steps to perform sequentially.
1. Start with playback idle and locate your DAW play head at the start of the audio you want to process.
2. Setup the active rack with the hardware you want to use. Use it in real time as many times as you want to dial in the settings. Do not worry if you use a compressed audio format during this time, the final audio will be uncompressed.
3. The arm button (1) is used to arm the processor to record audio, so click that next.
4. Start playback in your DAW. The offline processor will begin recording the uncompressed audio at the input to the plugin.
5. During recording, the signal is still being passed through the hardware at the server, using your currently selected audio format. This gives you real time feedback of the processed audio. Feel free to move the equipment knobs and switches during recording. These changes will be recorded as well and accurately re-played when the audio is processed at the server later.
6. When you get to the end of the audio you want to process, stop playback. You have now captured the raw audio that you want to process.
7. If everything looks right, click process (2) to send the audio to the server for processing, or click cancel (3) and return to step 1. When you click process, the offline processing interface will display “Processing…”. During this time you can continue working locally, and the plugin will simply pass through the local audio unchanged, because the server is busy processing your stored audio. Note that the server is processing your audio at full 24 bit lossless resolution, regardless of your format setting and internet connection.
8. Once the server finishes processing your stored audio, the interface will display the audio and the “Processing…” message will go away.
9. At this point you can click save (4) to save the processed audio to a wav file that can be imported into your DAW. Alternately you can click cancel (3) to return to step 1.
Inserting the Analog Matrix plugin on the master track of a session is always allowed at any buffer size setting because there are no parallel tracks to the master track and compensation is not necessary.
For individual tracks, each DAW provides plugin delay compensation. This can be used to compensate for the Analog Matrix buffer size setting and keep the track in sync with the session. Analog Matrix provides a selectable buffer size setting that represents the total buffer size, or delay, through the system. Each DAW has it’s own maximum delay compensation detailed below. The Analog Matrix plugin can be used on any individual track as long as the buffer size is less than the maximum delay compensation for that DAW, and delay compensation is enabled. If the Analog Matrix plugin requires larger buffer sizes for stable audio streaming, and the buffer size is larger than the maximum delay compensation, the track with the plugin will be out of sync with parallel tracks in the session. If you are processing the master track, then there are no parallel tracks, and so this doesn’t affect you.
If you are processing an individual track, you can compensate for this by manually moving the audio of the track earlier in the timeline. If delay compensation is not enabled, you would move the audio earlier by the buffer amount. If delay compensation is enabled, you would move the audio earlier by the buffer size minus the maximum delay compensation value.
Digital Audio Workstation
Max Delay Compensation
Since this is real hardware, mono processing cannot be simply duplicated as with digital plugins, unless you have two mono hardware units. We want to allow maximum flexibility in allowing you to route your audio through the hardware, so just about any combination is possible. For this reason, the routing can get complicated to explain, so here are some basic rules followed by some examples:
- If a mono signal goes into a stereo unit, the mono signal will be duplicated at the input, and the output will be stereo to the next link in the audio chain.
- If a stereo signal goes into a mono unit, the left/right signals will be summed to mono before entering the unit.
- If the output of the rack is stereo, but the output of the plugin is mono, then only the left channel will be passed to the output of the plugin.
- Side-by-side mono units behave like a stereo unit, and will process left and right signal accordingly, as expected.
- Side-by-side stereo units will use only their left channels to process the incoming left and right signal.
You will need recording software that uses the AAX, VST, VST3, or AU plugin formats on either the Mac or Windows platform. We have done testing with Pro Tools, Logic, and Reaper at this time, but expect more confidence in other applications as beta testing progresses.
You will need a robust internet connection. In general, wifi is not ideal for streaming low latency audio completely error free, but if you have a strong signal then it is usually fine. Since this is a professional audio application, we cannot lose even one sample of audio. So, the requirements for this streaming application are a bit more stringent than something like Netflix, which can drop a frame here or there without any noticeable effect to the application.
We transmit audio across the internet in one of the following formats, selectable by the user:
- 48kHz, 24bit PCM
- 48kHz, 16bit PCM
- Compressed 512 kBit
- Compressed 256 kBit
- Compressed 128 kBit
The compressed formats reduce the requirements on the internet connection but may not be desirable for professional results. If this is the only choice for a given user internet connection, there is a tool provided to allow real time usage with the compressed audio, but the final bounce or save can be done offline at full PCM resolution. See the user manual for more details.
You can measure your connection bandwidth at speedtest.net.
Do I Need An Account?
No account is needed to reserve equipment. However, an account will be automatically created upon your first reservation. You will receive an email with credentials to login to your account if you so choose. If you are logged in to your account you will be able to view your past and upcoming reservations.
What Buffer Size Should I Use?
See the separate tab – “Latency/Delay” – for a discussion of buffer size and delay compensation.
What Format Should I Use?
The uncompressed 24 bit format offers uncompromised audio quality, but it is the most demanding on your internet connection. Start with this value and see how the audio streams. If you get buffer underflows consistently, try one of the other formats. The choices are arranged from highest bandwidth (highest quality) to lowest bandwidth (lowest quality). If you use anything other than 24 bit uncompressed, but you still desire the highest quality audio for your final audio, you should make use of the offline processor described in the user manual.
How is My Digital Audio Being Delivered to Analog Hardware?
We utilize real time A/D and D/A conversion at the server. We are currently using an Antelope Audio Orion 32 unit. We will notify the email list about any new A/D units that we acquire as we expand in the future.
Has the Analog Hardware Been Compromised?
Never. We use a wide range of cutting edge robotics to control the hardware from the front panel, just like you would if you were interacting with it manually. All plugin meters reflect the physical needles and LED’s of the actual equipment. There is no digital simulation in any of our products.
How is Audio Routed in the Rack When Mixing Mono and Stereo Audio and Equipment?
See the separate tab – “Audio Routing” – for a discussion on audio routing.
Can I Create My Own Presets?
Coming soon. The factory presets in the rack are built into the plugin and cannot be changed. You will be able to create your own presets for the entire rack by utilizing the preset functionality of your recording software.
Can Anyone Steal or Listen to My Audio?
Your audio is never routed anywhere on our server except through the analog unit that you are using. So, we can’t hear your audio, and it is no more vulnerable to theft than any other unencrypted data going across the internet. Encryption increases the load on the internet connection, so we do not offer it at this time, but it could be added in the future.
Can I Put in a Request For Access Analog to Add a Specific Piece of Equipment?
Of course! We want to hear what you are interested in. Please email email@example.com and let us know.