Lesson Plan 3
Time required: Approximately 40 minutes
Refresher on mix minus and collaboration
In the last lesson we built a mix-minus session, which is a very important concept to understand at a core level when building remote collaboration sessions. The main reason for this is that you do not wish to send the remote audio signal back to the source, or they will get feedback, and you also don’t want to record yourself on top of the remote signal. Creating a session that keeps each signal distinct and clean is critical in order to get the recording you need.
In this lesson we will discuss advanced collaboration techniques: remote transport sync for working with picture or tracking music, and how Source-Connect can help you work over the Internet when dealing with inevitable problems such as packet loss and data arriving too late or out of order.
Lecture: Remote Transport Sync
When conducting a Source-Connect session you are actually sending and receiving audio data in real time with a different part of the world possibly a very distant remote location, depending on who you are actually connecting with. Let’s examine this process: the audio is fed eg into a mic, and passes this following chain of “hops” (presented in a somewhat high level of abstraction):
-> Mic -> Audio interface -> DAW (if any) -> Source-Connect (encoding and sending) -> Internet -> Source-Connect (receiving and decoding) -> DAW (if any) -> Audio Interface -> Monitors/Headphones
As expected, all of those hops carry out a certain process that takes an amount of time. Hence, it is obvious that the audio inserted in Mic will be played back at the remote Monitors after an amount of time as well. This amount of time is usually referred to as latency. There is the audio processing related latency (audio interface and computer-related) and the transport latency (Internet-related).
Now that we have covered the concept of latency one may wonder then how it could be possible to record from a remote end (eg a voice actor) some audio that will be synced to their own local audio or picture and not coming in late by whatever the latency happens to be. Clearly both sides can not be in sync simultaneously. However maybe it’s possible to have the audio you send be in sync at the receive side or have the audio you receive be in sync on your side. This is exactly what the Remote Transport Sync function does. Essentially it allows one side (the Sender) to control position/play/stop and manipulates the time either DAW’s play/record so that it simply waits for a bit (this “bit” is directly related to the discussed latency of course even counting in the latency of the DAW itself to go into play/record) and starts playing/recording exactly at the same time that the audio that is sent from the remote end reaches the receive. Result: the received audio is recorded perfectly in sync with the local audio/picture. In a sense one side chases the other, the side doing the chasing receives audio in sync. With RTS the Sender always controls the transport and the Receiver is controlled. There are two modes of RTS defined by where the Sender presses play: For the ADR/Overdub mode the Sender presses play in the Source-Connect RTS panel. The RTS Sender chases the RTS Receiver and is used to record/monitor received remote talent audio in sync with the local time line of the RTS Sender. For Review Mode the Sender presses play in their local DAW and the Receiver chases the Sender. In this case you (the RTS Sender) get to send your audio and it will arrive synced to the remote end’s (RTS Receiver’s) timeline. This would be used eg in a case where you have the talent in your studio and want to send your own audio (ie the talent’s voice) to be heard in the remote end synced to their audio/picture, and thus being subjected to real time remote review (hence the name Review) by an engineer/producer/director etc (it’s also possible for them to record what is received in sync).
You can find very detailed information on Remote Transport Sync as well as actual real life examples of how it is used here:
and also refer to the Source-Connect manual for the utilization specifics.
Lecture: Q Manager
The Internet is used mainly for the exchange of information in the form of data. Data sent and received is broken down to small parts (that are called packets) which are then sent in a specific order to the remote end. However, due to the very nature of the Internet’s inherent transfer infrastructure, those packets are susceptible to loss/disappearance (the explanation of this can be a bit technical) or simply, arriving out of order. Of course, a packet that arrives quite late is not useful in the case of real-time audio signal communications, so this packet too is considered lost. This phenomenon is generally known as packet loss, and if you experience it in your Source-Connect session, it will be perceived as a short break/dropout in the audio you are receiving. Source-Connect Pro has tools that try to resolve this in real time (see resilience and buffer size options), however it is indeed possible that this phenomenon occurs, especially if one user does not have high enough bandwidth (the upload is usually the sensitive part bandwidth-wise). This is when the Q Manager takes over. The Q Manager is a tool that is used after the recording of a file has been made, so essentially is a post-recording tool. In short, the locally recorded file is scanned for dropouts, and if such dropouts are found, the specific parts that are missing are requested from the remote end and are used to fill in the audio gaps, thus resulting in a perfect file. This process is called Restore. There is also the Replace function, which is a Source-Connect Pro to Pro function that enables the user to actually ask and receive from the remote end the full audio content as it was entered in Source-Connect, before passing from the AAC compress/decompress chain; this audio content that has full blown PCM audio quality is then used to replace (hence the name Replace) the locally recorded file. This enables the user to have a quality of recording that is exactly the same as if they were recording locally in their studio, with eg the voice talent in their studio booth. It is important to note that the received audio can be treated as the final audio recording. In other words the replaced audio will seamlessly follow any timeline edit performed on the originally received audio that passed through the AAC codec process. In this way Source-Connect Provides a backed up, seamless and lossless remote recording solution.
For a more detailed description of the Q Manager you can review the documentation here:
and also read the complete Q Manager section in the Source-Connect manual.
How can you use recording technology to assist avoiding losing that perfect take when there are internet problems? What are some of the possible solutions available when there is not enough bandwidth for high-quality connections, but only enough for low-quality audio for monitoring?
Action: Putting it all together – Recording a session with Source-Connect
In groups, establish a connection in different rooms so you can experience what it’s like to communicate remotely without seeing each other. Here are some important considerations you should understand thoroughly, to be comfortable working remotely:
- Have a backup plan, if the Internet doesn’t work, or something else goes wrong
- Understand, and be able to explain to non-technical people, why you need headphones or a Talkback system
- Have a basic understanding about how the Internet works in regards to ports, NAT/routing and Firewalls
- Understand that packet loss and other Internet issues may cause issues during a recording session, and how to mitigate those issues with the Q Manager and back-up recordings
- Know how to make your collaborators feel comfortable when working remotely, by communicating verbally or using text messaging
- Know where to get technical help and to set up your session and test beforehand, to avoid issues during a booked recording session
Action: Advanced setups (if you have time)
Multi-connect: In Mac OSX you can duplicate the application and route the audio to your DAW so you can create your own conference mixing app. Note: take care to use different ports for each connection