The Time Machine 720p Dual Audio

[Aili Peal]

Ihave read one post where the poster says Cubase will only allow him to select 1 interface at a time of the two he has connected to his system. A response to that post, there is possibly a third party driver that allows multiple audio interfaces.

Anyways I am right at the start of upgrading my studio from Cubasis on my ipad to Cubase on my laptop. I am trying Cubase AI as I got a free copy of it with my UR44. before I commit dollars for the pro ver I want to know I can expand primarily (at this stage) physical inputs up to 12-16.

I just started to clean out my old VCD collection and wanted to rip them to my hard drive. The problem is that its all Chinese VCD cartoons which has dual audio tracks. A mandarin and cantonese track. I tried looking around for a VCD ripper that can rip these .DAT files and isolate only 1 audio stream. I tried the program Handbrake and VCDGear, but they rip the entire video track with both languages heard at once. I want to be able to just rip 1 audio stream so my kids can easily click on the file and watch them without having to fiddle with the left and right audio selections.

You can configure and add the following Citrix policy settings to a policy that optimizes HDX audio features. For usage details plus relationships and dependencies with other policy settings, see Audio policy settings and Bandwidth policy settings and Multi-stream connections policy settings.

UDP audio encryption using DTLS is available only between Citrix Gateway and Citrix Workspace app. Therefore, sometimes it might be preferable to use TCP transport. TCP supports end-to-end TLS encryption from the VDA to Citrix Workspace app.

The loss tolerant mode supports audio. This feature increases the user experience for real-time streaming and improves audio quality over EDT when users are connecting through networks with high latency and packet loss.

Enlightened Data Transport (EDT) is a Citrix-proprietary transport protocol that delivers a superior user experience on challenging long-haul connections while maintaining server scalability. Loss tolerant mode is a feature of Citrix Gateway service that uses the loss tolerant mode as the transport protocol to maintain a stable connection even in the face of network congestion. This ensures a consistent and stable experience for remote workers. During normal conditions, both EDT and the loss tolerant mode provide similar results. However, during network conditions with packet loss, loss tolerant mode provides a better audio experience compared to EDT. This makes it an essential feature for remote workers who rely on real-time multimedia for their work.

In general, higher sound quality consumes more bandwidth and server CPU utilization by sending more audio data to user devices. Sound compression allows you to balance sound quality against overall session performance; use Citrix policy settings to configure the compression levels to apply to sound files.

By default, the Audio quality policy setting is set to High - high definition audio when TCP transport is used. The policy is set to Medium - optimized-for-speech when UDP transport (recommended) is used. The High Definition audio setting provides high fidelity stereo audio, but consumes more bandwidth than other quality settings. Do not use this audio quality for non-optimized voice chat or video chat applications (such as softphones). The reason is that it might introduce latency into the audio path that is not suitable for real-time communications. We recommend the optimized for speech policy setting for real-time audio, regardless of the selected transport protocol.

When the bandwidth is limited, for example satellite or dial-up connections, reducing audio quality to Low consumes the least possible bandwidth. In this situation, create separate policies for users on low-bandwidth connections so that users on high-bandwidth connections are not adversely impacted.

To allow users to receive audio from an application on a server through speakers or other sound devices on the user device, leave the Client audio redirection setting at Allowed. This is the default.

The Audio Plug N Play policy setting allows or prevents the use of multiple audio devices to record and play sound. This setting is Enabled by default. Audio Plug N Play enables audio devices to be recognized. The devices are recognized even if they are not plugged in until after the user session has started.

By default, Audio over User Datagram Protocol (UDP) Real-time Transport is allowed (when selected at the time of installation). It opens up a UDP port on the server for connections that use Audio over UDP Real-time Transport. If there is network congestion or packet loss, we recommend configuring UDP/RTP for audio to ensure the best possible user experience. For any real time audio such as softphone applications, UDP audio is preferred to EDT. UDP allows for packet loss without retransmission, ensuring that no latency is added on connections with high packet loss.

When Citrix Gateway is not in the path, audio data transmitted with UDP is not encrypted. If Citrix Gateway is configured to access Citrix Virtual Apps and Desktops resources, then audio traffic between the endpoint device and Citrix Gateway is secured using DTLS protocol.

If Audio over UDP Real-time Transport is not required for adaptive audio, Citrix recommends configuring the policy setting to Disabled. This helps avoid Citrix Workspace app clients requesting open UDP connections or triggering unwanted Citrix Workspace app client firewall configuration dialog windows to appear.

For setting details about Audio over UDP Real-time Transport, see Audio policy settings. For details about Audio UDP port range, see Multi-stream connections policy settings. Remember to enable Client audio settings on the user device.

As an Administrator, if you do not have control on endpoint devices to make these changes, use the default.ica attributes from StoreFront to enable UDP Audio. For example, for bring your own devices or home computers.

Users in audio or video conferences might hear an echo. Echoes usually occur when speakers and microphones are too close to each other. For that reason, we recommend the use of headsets for audio and video conferences.

A softphone is software acting as a phone interface. You use a softphone to make calls over the internet from a computer or other smart device. By using a softphone, you can dial phone numbers and carry out other phone-related functions using a screen.

Generic softphone support enables you to host an unmodified softphone on XenApp or XenDesktop in the data center. The audio traffic goes over the Citrix ICA protocol (preferably using UDP/RTP) to the user device running the Citrix Workspace app.

Client Hardware and Software:For optimal audio quality, we recommend the latest version of Citrix Workspace app and a good quality headset that has acoustic echo cancellation (AEC). Citrix Workspace app versions for Windows, Linux, and Mac support Voice over Internet Protocol. Also, Dell Wyse offers Voice over Internet Protocol support for ThinOS (WTOS).

CPU Considerations:Monitor CPU usage on the VDA to determine if it is necessary to assign two virtual CPUs to each virtual machine. Real-time voice and video are data intensive. Configuring two virtual CPUs reduces the thread switching latency. Therefore, we recommend that you configure two vCPUs in a Citrix Virtual Desktops VDI environment.

Citrix Gateway Protocol (CGP), which is used for the Session Reliability feature, also increases CPU consumption. On high-quality network connections, you can disable this feature to reduce CPU consumption on the VDA. Neither of the preceding steps might be necessary on a powerful server.

LAN/WAN configuration:Proper configuration of the network is critical for good real-time audio quality. Typically, you must configure virtual LANs (VLANs) because excessive broadcast packets can introduce jitter. IPv6-enabled devices might generate many broadcast packets. If IPv6 support is not needed, you can disable IPv6 on those devices. Configure to support Quality of Service.

Codec selection and bandwidth consumption:Between the user device and the VDA in the data center, we recommend using the Optimized-for-Speech codec setting, also known as Medium Quality audio. Between the VDA platform and the IP-PBX, the softphone uses whatever codec is configured or negotiated. For example:

The bidirectional Citrix Audio Virtual Channel (CTXCAM) enables audio to be delivered efficiently over the network. Generic HDX RealTime takes the audio from the user headset or microphone and compresses it. Then, it sends it over ICA to the softphone application on the virtual desktop. Likewise, the audio output of the softphone is compressed and sent in the other direction to the user headset or speakers. This compression is independent of the compression used by the softphone itself (such as G.729 or G.711). It is done using the Optimized-for-Speech codec (Medium Quality). Its characteristics are ideal for Voice over Internet Protocol. It features quick encode time, and it consumes only approximately 56 Kilobits per second of network bandwidth (28 Kbps in each direction), peak. This codec must be explicitly selected in the Studio console because it is not the default audio codec. The default is the HD Audio codec (High Quality). This codec is excellent for high fidelity stereo soundtracks but is slower to encode compared to the Optimized-for-Speech codec.

I have also started to use the envelope tool a lot when adjusting volumes between tracks, and if you do that, you have no use of the mixer board. I maintain that if you do multiple takes of the same voice as the standard procedure, as soon as you have completed the post processing and selection of that part, its probably good practise to tidy up among the tracks before you go on to another section. If not the main track window will grow more and more cluttered. Back to the mixer board, I do like the graphics of it, resembling a traditional hardware mixer board, and the better view of individual track volumes, but it also clashes a bit with the main window. Most of the time, I find myself wanting to see the waveforms because they convey so much info.

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