MAG Buffer setting. Thread starter steve99 Start date Aug 16, Hi I have had my Mag for a couple of days. I was watching football and it started freezing etc. I changed the internal portal buffer setting to ms which has sorted it.
My question is: I see there is another buffer setting on the Local settings menu which can from 1 to What is this buffer? Funnily enough I was aware of the buffer setting in settings but of the internal portal buffer. Curious to find out where that one can be accessed? PCM Registered. Does anyone actually know, what Buffer Setting and Buffering do?
PCM said:. You must log in or register to reply here.Keep in touch and stay productive with Teams and Officeeven when you're working remotely. Learn More. Learn how to collaborate with Office Tech support scams are an industry-wide issue where scammers trick you into paying for unnecessary technical support services. You can help protect yourself from scammers by verifying that the contact is a Microsoft Agent or Microsoft Employee and that the phone number is an official Microsoft global customer service number.
My Win 7 Internet TV stutters during playback. I assume this is because my DSL connection is around kbps or so, and I think that the "minimum required" bandwidth is around kbps Unfortunately, with "Internet TV", the buffer seems to be limited in size.
As a result, it will only buffer a couple of minutes before it stops downloading.
This then gives me about 5 minutes of watching before it starts stuttering again. Is there any way to increase the buffer size so that I can watch videos without repeatedly pausing and restarting? Thanks, Judd Windows 7, Home Premium, bit.
The internet TV does require higher bandwidth. The general accepted kbps download rate needed to stream high definition video iswhere as standard definition TV can range from to kbps depending on the resolution on screen. So I would suggest you to use the below mentioned methods and check. Method 1: You may reduce the hardware acceleration and check if this helps. Close Expression Media if it is open. Click Start and then click Control Panel. Click Appearance and Personalizationand then click Personalization.
Under Personalize appearance and soundsclick Display Settings. Move the Hardware acceleration slider to Mediumand then click OK three times. Close Personalize appearance and sounds. By default the hardware acceleration will be high, so reduce the hardware acceleration to medium level. Method 2: I would suggest you to change your decrease the screen resolution of your computer and check if this helps you to get a proper view of the Internet TV.
You may refer to the below link for assistance on changing screen resolution. If decreasing the screen resolution helps then it is possible that you may need a more powerful video card to help you resolve the issue completely. Hope this helps. Let us know the results. Thanks, Meghmala — Microsoft Support.
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Audio Latency – How To Optimize Your Audio Interface By Tweaking Your Buffer Size
System Folder: This folder stores all configuration files. Temporary Folder: This folder stores temporary files. Each instance of xTeVe has its own serial number and creates its own temporary folder. Any other IP address or hostname of the computer can be used.
The number of tuners determines how many concurrent streams are available. Some streaming providers allow only a certain number of parallel streams, then this value should be selected. XEPG only. Tuner: Number of concurrent streams for this playlist. Only with activated buffer. In percent. Streams assigned to a group can be filtered more easily. The automatic assignment is performed only new channels and also only once automatically.
Helpful for streams where the channel name changes regularly. If the stream does not contain this ID, the channel name is used for channel assignment. If the channel name changes during an update, this is a new channel for xTeVe and the old one will be deleted. The port may differ depending on the model and firmware. The client Plex, Emby, xteve. To reduce the number of streams, filter rules can be created. There are two types of filters:. For another group a new filter has to be created.
The number in bracket indicates how many streams are in this group. Include: The channel name must contain one of these words.If you have an application installed using USB method stuck at "Loading"when opening VODs movies on your TV, just re-download and reinstall the USB app from the application website again or uncheck the "Save online" checkbox when uploading your playlist.
Make sure the userwidget directory is located in the root of your USB drive. The application will appear on My Apps screen among other apps. Otherwise, there is no solution available. The application will appear on a Samsung Apps screen among other apps. You can also use Sync server IP address to use in developer mode. You can upload your own IPTV playlist by following these instructions.
It's possibile to set buffer size in app's settings for your streams in case you have frequent buffering problems. It is not guaranteed that it will help your slow connection or providers' stream issue.
You can lock your MAC address in application settings by using the Lock MAC button to avoid your playlist being reset by somebody else or if you shared your MAC address with some third party. URL user-agent: Star Wars.In some cases, when clients have low internet bandwidth, video streams buffer badly causing lagging and freezing, but a simple tweak inside the preferences section can remedy this for the most part and in most cases result in a completely lag free streaming.
The software interface on both of these platforms are almost the same and the tweak we are about to show you gets done the same way on both. To reveal Developers Options select Show All. Scroll down to to the bottom of the page.
Network Caching is what we are looking for. By default it is set to ms which means 1 second. The blue arrow shows the default position of the slider. The value needs to be changed to ms 10 seconds. Easiest way is to click between one of those 0 in the default value and add another 0 so it becomes ms. We insist on this because the value field is small and not showing the full input in it. Now click Save and retry your online feed.
It should surly play better. Let us know below. Comments powered by CComment. We have gathered the answer to all of your streaming questions. Simple, to the chase and detailed way of clarification. How to improve online video buffering on VLC player using network cache? Latest Articles. Hottest Articles. Resellers Guide: How to work with Xtream Codes panel? What are basic steps?Some of the members are only used by NDIS.
The members that drivers are most likely to use are defined in the following list:. Otherwise, this parameter is NULL.
ProtocolReserved Reserved for use by protocol drivers. MiniportReserved Reserved for use by miniport drivers. SourceHandle A handle that NDIS provided to the driver in a binding or attaching operation by using one of the following driver-supplied routines:. Miniport Driver MiniportInitializeEx. Filter Driver FilterAttach. NDIS drivers should not read this handle.
Otherwise, this parameter is zero.
Mag Box Help Tutorial : Change the Buffer
There are currently no flags available to drivers. Miniport drivers write this value before completing a send operation. This information is often referred to as "out-of-band OOB data. NDIS provides macros and functions to manipulate the data at Context. You may also leave feedback directly on GitHub. Skip to main content. Exit focus mode. Related Articles Is this page helpful?
5.2.3 Latency and Buffers
Yes No. Any additional feedback? Skip Submit. Send feedback about This product This page. This page. Submit feedback. There are no open issues. View on GitHub. Is this page helpful?In Section 5. A close look at this signal path shows how delays can occur in between input and output of the audio signal, and how such delays can be minimized.
Latency is the period of time between when an audio signal enters a system and when the sound is output and can be heard. Digital audio systems introduce latency problems not present in analog systems. It takes time for a piece of digital equipment to process audio data, time that isn't required in fully analog systems where sound travels along wires as electric current at nearly the speed of light.
An immediate source of latency in a digital audio system arises from analog-to-digital and digital-to-analog conversions. Each conversion adds latency on the order of milliseconds to your system. Another factor influencing latency is buffer size. The input buffer must fill up before the digitized audio data is sent along the audio stream to output. Buffer sizes vary by your driver and system, but a size of samples would not be usual, so let's use that as an estimate. At a sampling rate of A few milliseconds of delay may not seem very much, but when multiple sounds are expected to be synchronized when they arrive at the listener, this amount of latency can be a problem, resulting in phase offsets and echoes.
Let's consider a couple of scenarios in which latency can be a problem, and then look at how the problem can be dealt with. Imagine a situation where a singer is singing live on stage. Her voice is taken in by the microphone and undergoes digital processing before it comes out the loudspeakers. In this case, the sound is not being recorded, but there's latency nonetheless. The simplest way to reduce the latency here is to avoid analog-to-digital and digital-to-analog conversions whenever possible.
Buffer size contributes to latency as well. Consider a scenario in which a singer's voice is being digitally recorded Figure 5. When an audio stream is captured in a digital device like a computer, it passes through an input buffer.
This input buffer must be large enough to hold the audio samples that are coming in while the CPU is off somewhere else doing other work. When a singer is singing into a microphone, audio samples are being collected at a fixed rate — say 44, samples per second. The singer isn't going to pause her singing and the sound card isn't going to slow down the number of samples it takes per second just because the CPU is busy.
If the input buffer is too small, samples have to be dropped or overwritten because the CPU isn't there in time to process them. If the input buffer is sufficiently large, it can hold the samples that accumulate while the CPU is busy, but the amount of time it takes to fill up the buffer is added to the latency. The singer herself will be affected by this latency is she's listening to her voice through headphones as her voice is being digitally recorded called live sound monitoring.
Software monitoring requires one analog-to-digital and one digital-to-analog conversion. Depending on the buffer size and amount of processing done, the singer may not hear herself back in the headphones until 50 to milliseconds after she sings. Even an untrained ear will perceive this latency as an audible echo, making it extremely difficult to sing on beat.
If the singer is also listening to a backing track played directly from the computer, the computer will deliver that backing track to the headphones sooner than it can deliver the audio coming in live to the computer. A backing track is a track that has already been recorded and is being played while the singer sings.
Latency in live sound monitoring can be avoided by hardware monitoring also called direct monitoring. Hardware monitoring splits the newly digitized signal before sending it into the computer, mixing it directly into the output and eliminating the longer latencies caused by analog-to-digital conversion and input buffers Figure 5.
The disadvantage of hardware monitoring is that the singer cannot hear her voice with processing such as reverb applied. When the mix knob is turned fully counterclockwise, only the direct input signals e. When the mix knob is turned fully counterclockwise, only the signal from the DAW software is heard. In general, the way to reduce latency caused by buffer size is to use the most efficient driver available for your system. ASIO drivers cut down on latency by allowing your audio application program to speak directly to the sound card, without having to go through the operating system.
Once you have the best driver in place, you can check the interface to see if the driver gives you any control over the buffer size. If you're allowed to adjust the size, you can find the optimum size mostly by trial and error.