Sample rate conversion quality windows xp

Certainly, when I attempted to upload a 48kHz clip, it was converted to At However, I suspect that the default rate using the Fast Track will indeed be Note that the Inmatrix Zoom Player support personnel have reproduced the problem as well. They can hear the artifact, if they select WaveOut for the output renderer. The same problem occurs with the HD Audio Class driver. I tested with YouTube. However, the class driver defaults to I forget what I had it set to before the install, but whatever it was it wasn't bit, so I assume that the class driver install did change the settings.

I'm doing a system restore and will report back. Given that Windows 7 does not change the hardware config on the fly like XP used to , I would have thought that a default rate of 48kHz and bit would make a LOT more sense, because otherwise 48kHz material will be downsampled to So, as far as I can see, the default settings that the Realtek driver used 48kHz, bit made sense.

Will I get a warning if Windows is not going to increase the hardware rate above 48kHz, or will it silently downsample to 48kHz? Anyway, a different topic I guess. EDIT: the audio interface was set to However, I've repeated the audio class driver install, with the Realtek settings being on 48kHz, bit the default for my system , and as expected, it was changed to I can hear the problem on their demo tracks.

It sounds noticably cleaner when I set my audio interface to Is there a registry setting for the WaveOut sample rate conversion quality? That utility also fails on my system, at least in the same way, if the sample rates do not match.

They thought I was half mad for suggesting that the sample rates could possibly not match. Is your sample rate conversion really THAT good? Will it retain every last Hz of bandwidth, and every last bit of detail, of HD content? I couldn't immediately see how to use the Passmark SoundCheck to actually perform analysis of the soundcard, so I downloaded "Sound Card Analyzer 1. As expected, this too doesn't work properly on my Windows 7 system, unless I make sure that the sample rate that is configured in the application matches that set in the Windows properties.

If they don't match, the results are non representative of the hardware capabilities. For example, I'm able to put in a sample rate of into the program, and into Windows, and the frequency response will be restricted to circa 20kHz but with a strange spike right up near 48kHz due to resampling artifacts or whatever I've tried this on both the internal Realtek and the external M-Audio Fast Track Ultra with similar results.

Using the Realtek, I could not get the right channel to work, and I could not force the application to only test one channel, however the results appeared as if it may have been ignoring one channel.

I did configure the interface for line-in, rather than mic, but I just couldn't get stereo to work. Other test results are of course affected by sample rate discrepancies, such as THD. Note that the sound level check pre-test did not work properly when the sample rates did not match.

For example, the -6dB test would produce a 0dB input level, which is a huge error. So, any user of this utility would most likely be alerted to a serious problem before doing the tests proper, which is good. I did the tests despite this issue. The point I'm trying to make is that on XP and earlier, we didn't have to do this crosschecking, and the uninitiated may get into strife because of this change in behaviour.

So I was wrong. I really expected XP to reconfigure the hardware. I haven't tried other APIs yet and I haven't been able to install it on Windows 7 yet - the install barfs. Thanks for the info Matthew. In any case, that's not the main issue at the moment. The main reason I came here is to report what I feel is inadequate fidelity in YouTube, and, it seems, other streaming media.

I didn't go looking for this problem - I noticed it straight away, when I viewed my first YouTube clip which was normal music. I've double checked that my default sample rate is 48kHz on this particular machine, by doing a fresh install using the supplied Recovery Media.

Windows Starter. I apologise if I've mentioned too many issues in this thread. I thought the sample rate setting issue would be very clear cut, but it's not. Maybe I'm making too much of it. Maybe most folks realise that internet audio is often lossily compressed, and not to expect too much. It's bugging me a lot though.

It's bad enough for me to feel the need to add a note to anything I would upload, cautioning my listeners to check their sample rate, if they are running Windows 7 or Vista, I assume.

This is silly. I've re-tested the Passmark SoundCheck utility. Thankfully, it is in fact behaving much closer to how I always expected it would behave, on Windows XP.

If it is not a supported rate, the hardware rate remains unchanged. I also re-tested it on Windows 7. The hardware will remain on the sample rate set in the Windows advanced properties for the audio interface. I have "Allow applications to take exclusive control of this device", and "Give exclusive mode applications priority" both checked. Can you play your test tone in the following two configurations, capturing the output of the audio engine both times this is what we send to the sound card and then mail me the resulting.

Play the tone first; then start the capture app; then press Enter to stop the app; finally stop the tone. Re: SoundCheck: in Windows 7 you can use "exclusive mode" playback to play to a non-default sample rate supported rates only, of course.

I gather that SoundCheck is not doing this. There's a sample "exclusive mode" player here:. Note that I would have recorded the output at the outset, if you had said that you could not reproduce the problem.

The problem is so obvious to me when playing a low frequency sine wave, that I simply didn't feel it necessary. I would have even given you an analysis of the recording to prove that the signal was being distorted. Just in case it helps anyone else, I had to run that loopback capture utility as Administrator, and it only allows a few seconds worth of audio to be captured.

It correctly restricts the available sample rates to that which the hardware supports great! Again, I was able to run the tests with the sample rate discrepancy, with no errors whatsoever, but with the expected adverse results. I thought this would be easy, by checking the frequency response results, however I'm having trouble getting it to work.

I'll do that extra test. If anyone knows of a utility that can query the RealTek HD and determine it's current sample rate, please advise - thanks. Without being able to monitor the hardware rate of the Realtek, I had to rely on frequency response to verify a discrepancy in sample rates.

I wasn't able to get meaningful results from this initially, but I've sorted out the problem and now it's working. I was not aware that the sample rates for playback and recording were independent, and I was not setting the record sample rate!

It only has one setting for sample rate. EDIT: p. If anyone has a WAV file that is playing using this app, please advise and make available if possible - thanks. Greg, I haven't had time to look into this carefully, but you do need to be very careful when using the Realtek and m-audio devices together under Windows 7.

If I then open the Windows 7 audio Control Panel and select the "Recording" tab, the m-audio asio driver gives a "sample rate unavailable for this device" error from the supposedly independent of Windows asio high level application. When this occurs depends on the particular audio app - I've been using mainly Wavosaur and Adobe Audition 3. I have been assuming that the metering part of the Win7 recording tab is driven by a small recording application inside Windows 7 and that this is in some way hooking together the MME and asio drivers.

As you can tell, I really don't understand much of this. I have posted before here and elsewhere about the fact that I have other problems caused by the Windows 7 Control Panel on some machines and with external audio devices from several manufacturers.

I haven't had many replies. I have been testing the onboard Realtek seperately to the M-Audio, and both seem to behave much the same.

So far, I haven't noticed anything that suggests that the OTHER device is interacting in some way, although I suppose anything is possible. I'll try what you are doing, though, just out of curiousity. I do know that on XP, I was able to use an ASIO app, whilst at the same time play a tune from iTunes, with both audio sources being mixed - it worked well.

I haven't tried that on Windows 7 yet. Greg, I may have confused things a bit here. The effect that I refer to above is probably unrelated to whether the Realtek device is in use or disabled. Where the two devices appear to affect each other is when I use an AMD-based laptop and a usb1.

There I can trigger horrendous audio glitching on the asio usb audio by disabling the Realtek device or by selecting any except the "Recording" tab in the Win 7 Control Panel. This is, of course the opposite of the other effect. One requires the "Recording" tab selected, the other not. My workaround for this is to insert a usb2 hub before the usb1. I really would like to know definitively why selecting tabs on the Win 7 Control Panel affects asio audio.

I'm not at all convinced that your issues are relevant to mine, so I'm not going to spend any time looking at it just at the moment - at least, not in this thread, unless others disagree with me and think that they ARE related. Audio artifacts, when the sample rate of the audio interface is set to 48kHz and the content is Affects streaming media such as YouTube etc. In the past, it was typical for these applications to change the hardware rate without any other user intervention.

Now, it seems that the user has to ensure that the sample rate is changed in the Windows advanced properties, such that it matches that set in the applications. Trying to test exclusive mode, to determine whether this mode actually does change the hardware sample rate or not. I fully expect it to! Greg, You are probably right when you say that your issues are unrelated to mine only the one that produces sample rate error messages would be relevant anyway , but I would still maintain that you need to be wary with these tests.

I have just tried loading a The m-audio control panel follows the sample rate of whichever file is being played. This seems as it should be. If I play the first YouTube file referenced in your original post, it plays correctly with the m-audio control panel set to It sometimes sounds correct with the panel set to 48, but usually there is what sounds like a machine-gun interruption of the audio.

In this case, the M-Audio device seems not to pick up the sample rate and match the file. I am a little dubious about the YouTube source file as it never seems to reach the stated 10kHz audio frequency. I'm also not sure that what I'm hearing is aliasing, which, on a sliding audio tone, I would have expected to sound like other tones swooping and diving in the background, rather that this which sounds more like a clocking mismatch.

Next, I loaded the two files into Adobe Audition 1. This does not reset the sample rate in the M-Audio Control panel to match the file being played, but the files play at the correct speed, so I assume some sample rate conversion is occurring.

I think DirectSound drivers are in use here. Then I generated a sliding tone in Audition at a sample rate of 48kHz and played it via Audition 1. With the M-Audio set to 48kHz, it played cleanly, set to M-Audio device is the default, and is a usb 1. Machine AMD-based dual core laptop.

The browser is Firefox. Sound is clean LF tone with very occasional clicks. Sample rate remains on Play test tone, still sounds as above. On starting the play the M-Audio CP changed to The Windows 7 CP remained at 48kHz. Closing and reopening Windows CP makes no difference. The sketch is tasteful, your authored material stylish. Paolo Rodriguez Jan 2, at am. Good way of describing, and pleasant article to obtain information on the topic of my presentation subject matter, which i am going to convey in institution of higher education.

Thanks for this! As you may have noticed, there are some improvements to the audio stack in Windows 10! I use an M-Audio , and since installing Windows 10, I can set the sound mixer to output to the sound card in 96K bit. It would only do bit before. The M-Audio actually processes internally at bit before downsampling to Lucas Sep 25, at pm.

Sorry for this I listen to music on XP and always. Matt Bentley Jul 15, at am. Almost everything about this article and the comments is incorrect. AFAIK the system does not automatically feed the soundcard with bit audio, it has to be specifically supported by the app. Teeluck Mar 11, at pm. The system, after processing the audio, feeds the final audio samples to a limiter, which will limit all clipping samples in a proper way to prevent excessive audio distortion.

The system feeds the audio device with what was selected by the the user in the device advanced settings. The final data sent is a dithered and limited as described before version as selected. My Realtek codec is as default set for 24bit 48khz. Tomasz P. Szynalski Mar 14, at pm.

The floats whether bit or other allow no clipping when mixing two sounds, because they have enormous range. Of course later you have to limit it. Leave this field empty. In my opinion, there are a few reasons why audiophiles should be happy with this change: The new audio stack automatically upconverts all streams to a bit floating-point sample depth the same that is used in professional studios and mixes them with the same precision.

Because of the amount of headroom that comes with using bit floats, there is no more clipping when playing two samples at the same time. There is also no loss of resolution when you lower the volume of a stream see below. This corresponds to dividing each sample by In an ideal world, after the volume control we would get 4. However, if the output stream has a bit depth just like the input stream, then both output samples will have to be truncated to 4. There is now no difference between the two samples, which means we have lost some resolution.

Well, these abominations should now be behind us. In Vista and Win7, each application gets its own audio stream or streams and a separate high-quality volume control , so there should no longer be any reason for application vendors to mess with the system volume or roll their own and botch the job. Whatever type of music you enjoy and however you want to hear it, we have something for you at Musical Fidelity. Musical Fidelity Driver Download For Windows 7 Now owned and operated by Audio Tuning, an independent audio specialist company based in Austria, Musical Fidelity still retains its proud British heritage, with every product we design continuing to boast the natural, authentic sound that has made our name so respected across the world.

Musical Fidelity Driver Download For Windows 10 Whatever type of music you enjoy and however you want to hear it, we have something for you at Musical Fidelity. Is there any good data on this? That totally depends on the used algorithm. That would depend on a number of things. You're correct, any integer sample rate conversion should be more or less lossless. I think floating point conversions should also be undetectable if done with good enough quality settings.

Sometimes you have to care about inter-sample overs, so decreasing gain slightly could be a good idea. Last edited: Aug 3, Veri Master Contributor. Joined Feb 6, Messages 8, Likes 9, Windows has a pretty shite resampling system. And I'd say even that is not too audible. I believe our ears are quite forgiving, which makes super-over-sampling the like a Chord M-Scaler does even more ridiculous. That's just my view on the matter, at least Veri said:.

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