There are six main sections of the interface:
- Top Panel
- Routing Panel
- Offline Processor
The top panel contains overall controls and utilities for enabling the connection to the server.
- The email and password are the credentials used when you reserved time on the website. The password was emailed as part of your first reservation.
- The server menu shows the list of available servers in the cloud. Choose the server that corresponds to the equipment that you reserved on the website. Currently there is only one Access Analog server.
- The connect/disconnect button is used to establish the connection with the server once the email and password are entered. When not connected, the plugin will pass through audio unaltered. When connected, the audio will stream to the server. You may connect to the server at any time and stream unprocessed audio, even without a reservation.
- The buffer setting can be used to increase the sample buffers if dropouts become recurring or frequent. See the Buffering/Latency tab on this website for more information. Higher buffer values have a better chance of reducing buffer underflows, but they cause more latency between adjusting a front panel setting and subsequently hearing that change. This latency is approximately half of the total buffer size value.
- The transmit format is used to set the format that will be used for the bi-directional audio stream across the internet. The uncompressed formats require a more stable connection between the client computer and the server. The compressed formats can be used if connection quality is limited. See the separate tab on Format for more information.
- The buffer graph displays the size over time of the sample buffers used to handle internet disruptions. The client and server buffers are overlayed on top of each other and a solid orange color indicates both have samples and audio should be streaming. See the Buffering/Latency tab for more details.
- The mix 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 sidebar on the left side of the plugin lists all of the equipment located at the server to which you are connected. After connecting to the server with the proper credentials in the top panel, this list will show all of the equipment that is located at that server. Each piece of hardware can be in one of five states:
- Unavailable – the equipment is currently in use by another user, or it has been taken offline for repair and maintenance.
- Available – the equipment is not in use and available for reservation on the website. Clicking the hardware will take you to the reservation page.
- Reserved – the equipment is currently reserved for you and available to be dragged into the routing panel for active use.
- In Rack – the equipment is currently reserved for you and it is already being actively used in the routing panel or in another instance of the plugin.
- Update Plugin – the equipment was added to the service after your version of the plugin was released. Go to the Downloads page on the website and install the latest plugin.
The routing panel is used to show the path of mono and stereo audio through the system, build analog chains with multiple pieces of equipment, and select equipment so that you can use the front panel in the equipment section. The source audio coming into the plugin is represented on the left side of the panel. The left audio channel is the top row and the right audio channel is the bottom row.
- The first piece of equipment of your analog chain is inserted into the left column of the panel by dragging your reserved equipment from the sidebar. Dragging equipment to subsequent columns to the right will insert them later in the analog chain and audio will flow through them in series from left to right. Clicking on an equipment name will highlight it and show the corresponding equipment faceplate in the equipment panel so that you can adjust the controls. A stereo device will occupy both the left and right boxes of a column.
- Mono pieces can be inserted on either the left or right audio channel. When two mono pieces are inserted in the same column, they are in a “dual mono” configuration where the left audio goes through one device and the right audio goes through the second device.
- If both pieces of equipment are the same model, they can be linked with the link button. When linked, control changes to the device on the left channel (using the equipment panel described later) will be duplicated to the box on the right. When linked, control changes to the box on the right channel can be adjusted independently of the left channel to compensate for minor hardware differences between the two devices.
- A gain slider is provided for each box in the analog chain. This gain adjusts the level of the signal being sent to the box. A mix knob is also provided to mix the signal from the box with a dry signal that bypasses that box.
- The “In” button gives the ability to easily insert and remove a box from the audio path while audio is playing. Similarly, the solo button will insert the box into the audio path while removing all other boxes.
- The final gain slider adjusts the level of the signal after the last box in the analog chain and just after the final A/D conversion. This effectively adjusts the output of the plugin and will always be the last slider on the right, regardless of the number of boxes in the chain.
- Meters (0 dBFS) show the RMS signal level at all stages of the audio path. If a sample reaches full scale, the clip indicator will light. Click on the meter to clear the clip indication. The clipping is handled by the A/D converter and can sometimes be desirable at moderate levels depending on your specific application.
Equipment can be removed from the chain by clicking the ‘x’ to the top right of the equipment name. Any columns that are empty will simply pass through audio unchanged. You can drag and drop equipment from one box to another inside the equipment panel in real time with audio playing to compare different analog chain equipment sequences with ease. Devices will swap positions or move to new points in the analog chain and the hardware will immediately be re-routed at the server to reflect the new configuration.
The equipment panel shows the likeness of the equipment currently highlighted in the routing panel.
- The remaining time shows the time left for this device in the current reservation that you made on the website.
- The title shows the equipment name which indicates the model and instance. If there is more than one device of a particular model at the server, they are numbered #1, #2, etc.
- The preset menu contains a list of factory presets for the device, followed by user presets. Add a user preset by setting the controls of the front panel as desired, then click the ‘+’ button and name the preset. User presets can be removed by selected them and then clicking the ‘-‘ button.
- The equipment controls occupy the main area of the window. These control the robotics at the server. Use the equipment manual online (links on our equipment page) if you want to dive deep and understand how these controls work.
The offline processor is located at the bottom of the plugin and can be used to capture a clip of audio locally at the plugin, upload it to the server, process it at the server, and collect the full 96 kHz, 24 bit samples back at the plugin. The nature of this process makes it unaffected by internet interruptions. This method can be used on any internet connection to safely capture the final processed audio for your reservation. The real time streaming format at the top of the plugin does not affect offline processing, which is always processed at 96 kHz, 24 bit. Offline processing is best covered as a sequence of steps to perform sequentially.
- Start with playback idle and locate your DAW play head at the start of the audio you want to process.
- 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 full resolution as noted above.
- The arm button (1) is used to arm the processor to record audio, so click that next.
- Start playback in your DAW. The offline processor will begin recording the uncompressed audio at the input to the plugin.
- 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.
- 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.
- 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.
- Once the server finishes processing your stored audio, the interface will display the audio and the “Processing…” message will go away.
- 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.
- Check this box to show the equipment thumbnails in the sidebar.
- This checkbox can stop the flow of audio to the server. This is reserved for future use and should always be checked under normal operation.
- This checkbox can override the address of the server. This is reserved for future use and should not be checked under normal opration.
- This checkbox can turn on some debug logging for the plugin. Use this to create a log file that can be emailed to technical support when issues arise.
The system allows the user to select the audio format that will be used to transmit the data across the internet and back. This format can be chosen to match the sample rate of the user session, or it can be adjusted to match the quality and bandwidth of the internet connection.
When the sample rate on both sides of the SRC are the same, then no conversion takes place. Thus, when the session sample rate and transmit sample rate are the same, there is no sample rate conversion at the plugin. Similarly, when the transmit sample rate is 96 kHz, there is no conversion at the server.
The transmit sample rate should not be set higher than the DAW session sample rate, as that would use more bandwidth and result in no benefit.
The hardware A/D converter is set to 96 kHz.
When offline bounce is used, the local samples in the plugin are converted to lossless 96 kHz, 24 bit samples. These are then uploaded and processed at the server with no further loss or conversion. Thus, offline bounce looks like this:
The system employs two buffers to manage the inconsistencies of the internet. There is one buffer located at the server which receives samples from the client plugin. There is another buffer located in the plugin that receives samples back from the server. The user can select the entire system buffer size at the top of the plugin, between 300ms and 2500ms, and this is split between the two buffers.
The buffer graph is located at the top of the plugin. This shows the size of both the plugin and server buffer relative to the total system buffer setting. The two graphs are overlayed on top of each other and show the fill percentage of both buffers over time. For best streaming, each buffer should be approximately half full, resulting in a bright orange graph that is half of the height of the graph, as shown. If one graph is much larger than the other, you can use the reset button below the graph to clear up the issue.
Inserting the Analog Matrix plugin on the master track of a session is always possible 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
You will need recording software that uses the AAX, VST, VST3, or AU plugin formats on either the Mac or Windows platform.
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:
- Lossless, 44.1 kHz, 24 bit
- Lossless, 48 kHz, 24 bit
- Lossless, 88.2 kHz, 24 bit
- Lossless, 96 kHz, 24 bit
- Lossless, 44.1 kHz, 16 bit
- Lossless, 48 kHz, 16 bit
- Lossless, 88.2 kHz, 16 bit
- Lossless, 96 kHz, 16 bit
- Compressed, 48 kHz, 512 kBit/sec
- Compressed, 48 kHz, 256 kBit/sec
- Compressed, 48 kHz, 128 kBit/sec
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, the offline processing tool is provided to allow real time system usage with the compressed audio, but the final bounce or save can be done offline at lossless, 96 kHz, 24 bit resolution. See the user manual offline processing topic 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 – “Buffering/Latency” – for a discussion of buffer size and delay compensation.
What Format Should I Use?
You can first try the 24 bit format that matches the sample rate of your session. If the streaming is intermittent, you can drop the sample rate and/or use the 16 bit format. If the streaming is still intermittent, you should use a compressed format for real time streaming. When you are ready to capture your processed audio, you can use of the offline processor described in the user manual to get full resolution 96 kHz audio.
How is My Digital Audio Being Delivered to Analog Hardware?
We employ real time A/D and D/A conversion at the server using a Lynx Aurora(n) converter.
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.
Can I Create My Own Presets?
Yes! Use the ‘+’ button at the top of each individual equipment panel to create presets for that piece of equipment.
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 firstname.lastname@example.org and let us know.
What is a Lynx Loopback Cable?
We have two loopback XLR cables connecting two outputs back into two inputs of the Lynx converter that are available for reservation at no cost. When you reserve these and drag them into your plugin rack, you will stream audio through the cables just as if they were an analog unit. These cables allow you to learn the system and fine tune your streaming parameters without having to spend money. You can also use them to analyze the Lynx conversion.