Talk:Windows legacy audio components

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Cwolfsheep 18:03, 8 July 2006 (UTC)[reply]

Fit into Category:Music software plugin architectures?[edit]

Hello! Can anybody in the know tell me if Windows audio components/DirectSound, apart from its regular usage in games, is used by Music software applications, like sequencers and DAW to add sofware synthesizers and software effects to audio tracks? Which ones? Thanks :-) Peter S. 17:16, 8 July 2006 (UTC)[reply]

I believe they are. Adding cat. Cwolfsheep 18:03, 8 July 2006 (UTC)[reply]

- Have you got an example url of a DAW or a plugin manufacturer that uses it? Peter S. 21:21, 8 July 2006 (UTC)[reply]
- Not handy. The closest I get to editing is using VirtualDub & Audacity: there are some "DirectSound" or "DirectShow" filters used.

Professional audio software usually offers a choice between WaveOut/MME and ASIO. Certain soundcard vendors sacrifice a bit of audio quality to gain performance when implementing the DirectSound part of their drivers, leaving it a less desirable choice for such applications. JoaCHIP (talk) 16:39, 1 January 2013 (UTC)[reply]

ACM vs. DirectShow[edit]

Why are there both ACM filters and DirectShow filters, and what's better? For example, there are both ACM and DirectShow filters for the LAME MP3 codec at http://www.rarewares.org/mp3.html. --Abdull 22:47, 2 September 2006 (UTC)[reply]

Another question, about this part of the article:

However, unlike ACM and the related Video Compression Manager (VCM), DirectShow provides no means to encode files for end-users but requires developers to build end to end graphs for encoding content.

I am no audio processing newbie, but I have no clue what this sentence is trying to explain. Someone care to elaborate and maybe make it a bit clearer in the article? 62.167.77.101 (talk) 10:36, 15 August 2008 (UTC)[reply]

latency[edit]

"the latency of KMixer is around 30 ms and it cannot be reduced". This is not true. It is possible to reduce the KMixer's latency to 5-10 ms. --85.101.213.251 (talk) 23:10, 28 May 2009 (UTC)[reply]

Limitations of MME[edit]

"MME supports up to two channels of recording, 16-bit audio bit depth and sampling rates of up to 44.1 kHz with all the audio being mixed and sampled to 44.1 kHz." I have used lots of audio software that offers 24-bit 96000 Hz operation using WaveIn and WaveOut. (Samplitude, Vegas, Buzz and more) Such resampling would remove or destroy any audio content above 22050 Hz, which is not the case. I suspect this section might be either wrong or misleading, or maybe it refers to something specific? Can anyone clarify? — Preceding unsigned comment added by Joachim Michaelis (talk • contribs) 16:31, 1 January 2013 (UTC)[reply]

Dubious[edit]

Technical details of the sample rate conversion issue in Vista & Win7 are being challenged by editors. I added the original paragraph, fully sourced, which included the phrase "the internal resampler, which is no longer configurable, defaults to linear interpolation, which was the lowest-quality conversion mode that could be set in previous versions of Windows".

24.68.108.247 then qualified "linear interpolation' as "fast integer-based linear interpolation" and added a descriptive example (new sample is taken as an exact duplicate of the nearest sample instead of a varying portion of the two nearest samples). The explanation for this edit was true linear interpolation (more computation) would not cause the "audible noise" caused by fast linear interpolation.

71.167.59.95 has now tagged that example as dubious, explaining that's not what linear interpolation means. either they're using zero-order hold interpolation or your description is wrong.

All I can do is point to the sources that are already linked.

Here is what the developer said in the MSDN discussion thread:

We have found the problem, which is a bug in waveOut on Vista and Windows 7. Windows XP does not have this problem. In Windows XP, the sample rate conversion quality in KMixer is controlled by the Sound control panel: http://msdn.microsoft.com/en-us/library/ff538617%28v=VS.85%29.aspx
The document he refers to, which I also linked to, mentions the pre-Vista slider which "assigns the settings Good through Best to linear interpolation, multipoint interpolation, and high-end multipoint interpolation, respectively. Linear interpolation is the default for the DirectSound versions that ship with Microsoft Windows 98/Me and Windows 2000. In Windows XP and later, the default is high-end multipoint interpolation."
He goes on to say In Windows Vista KMixer was removed and the audio engine was moved up into user mode. The sample rate conversion quality meter was removed from the Sound control panel. Media Foundation, DirectShow, DirectSound, and waveOut each do sample rate conversion slightly differently. There is a bug in the waveOut sample rate conversion which results in a lower-quality sample rate conversion than was done in XP.
He later refers to the Vista/Win7 hotfix, the description for which says This issue occurs because the sample rate converter uses linear interpolation when it converts audio files. This behavior creates noise on the audio file that is sensitive to the human ear.

If we can't describe it in more detail, I think we should just stick to what the sources say, which would mean leaving it the way I originally had it. —mjb (talk) 23:09, 14 October 2013 (UTC)[reply]

- if this article https://www2.iis.fraunhofer.de/AAC/ie9.html is indeed describing the "Issue", then I think "poor audio quality on playback" on that page is vague. If that is really the problem (linear interpolation instead of something slightly better), then it is a separate problem. The problem I was describing is much worse, skipping interpolation entirely (requiring no floating point math at all), which produces a duplicated or skipped sample (depends on whether it is 44.1kHz to 48kHz or vice versa) every 11~9 samples or so. This creates very noticeable artifacts in audio that contains low frequencies and very little or no mid-high frequencies, such as a solo bass guitar. The sound is sort of a metallic ringing, sort of similar to a bit-crusher sound. This is a waveOut problem, and not a DirectSound problem (e.g. iTunes will sound fine playing the same audio file as waveOut when the waveOut application uses a different sample rate than the endpoint device of Windows). I could provide source code as an example of how to resample audio this way, but I don't think Microsoft is going to chime in for a citation. — Preceding unsigned comment added by 184.151.231.111 (talk) 07:19, 31 October 2014 (UTC)[reply]

- The language here is very iffy. "defaults to a fast integer-based linear interpolation (e.g. new sample is taken as an exact duplicate of the nearest sample instead of a varying portion of the two nearest samples)" You are saying "linear interpolation" but go on to describe "nearest neighbor". So which one is it? — Preceding unsigned comment added by 188.173.53.45 (talk) 14:12, 9 September 2020 (UTC)[reply]