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Other Lossy Codecs / Re: Quite OK Audio (QOA)... anyone ?
Last post by Klymins -
@Porcus Sampling rate is independent from quality and it is not an unusable sampling rate, some Flash games were used it in their musics.

As ADPCM-XQ does not use predictions, it can't reach QOA's quality in most cases. Also, LossyWAV uses frequency domain if I understood correctly and this makes it very complex from beginning.

I don't understand why you don't want having a 4-bit variation of QOA. I think it will be very useful as Flash ADPCM also has 2, 3, and 5 bit variations and they are very useful.
3rd Party Plugins - (fb2k) / Re: NEW ESLyric v0.5 - an alternative lyric show component
Last post by ngs428 -
On an unrelated question....   In the Lyric Search Window (right click in ESLyrics > Search), if you click on the "Search" dropdown in the top right corner, and pick "Sync Settings" what does that do? 
The check status of the lyric source is the same as the setting in the Prefences options.
There is no feature to download lyrics in batches.

Thanks for the reply @always.beta , I am not finding the setting in the Preferences options that would match that setting.  The attached picture shows the setting I am looking to see what it does.  I assume the setting must be in the "Lyric Sources" or "Search Settings" sections in preferences.  

Other Lossy Codecs / Re: Quite OK Audio (QOA)... anyone ?
Last post by Porcus -
A mighty 3 octaves down from CDDA is quite a departure, and you wouldn't expect quality reproduction of music there in any case. But it does something to your need for bitrate.

* How dows it compare to ?
* How low can you go with LossyWAV --> flac -8pb512 ?  The 1.4.3 beta has a few new presets at the low end:,109239.msg1045012.html#msg1045012
Lossless / Other Codecs / Re: HALAC (High Availability Lossless Audio Compression)
Last post by Case -
Therefore, since we load only the frames we want into memory, the memory consumption problem is also solved. You can see this by trying it on the player.
I took a look at the player sources before my initial post. Your demo player loads the entire source file into memory so it can't be used to demonstrate playing a partial track.

In other words, it is enough to change the frame start and end numbers at the bottom right and play them. If it is on a different application, the GET_WAV_FRAME function should be used. (the frame size is currently set to 1 mb).
I can only make GET_WAV_FRAME work with partial buffer. I can't keep it from crashing unless it's given a buffer containing the full HALAC file's contents.

I mean, about streaming...

If valid HALAC data(frame) is coming from any source that does not contain header information, what do we do?
My streaming comment didn't mean using HALAC for streaming a radio station or such. It was about ability to start playing a HALAC file coming through a slow pipe, for example a not-so-fast NAS storage or over HTTP. Fast enough for realtime playback but not fast enough to want to wait for full file to download before playback starts.

There must be some rules of communication between them. I tried to act thinking that this information was on both sides. That's why I created the GET_RAW_FRAME function. The information provided(number of channels, bit depth, compressed frame size...) can decode in light of.
I can't make GET_RAW_FRAME() work at all. Initially I tried feeding it partial data from the HALAC source file starting from FRAME DATAS 1 position, library instantly crashes. Then I tried giving it a buffer with the entire track loaded in the memory, still crashes with the first call.

Regarding the file format, right now application that would want to support HALAC in a manner that doesn't require loading entire track into memory needs to understand the header and parse METADATA FIRST SIZE field so it can make sure it has enough data to get access to the FRAME COUNT field, then read enough data so it has all the bytes needed to handle the framesizes. Having all size info early in the file and in common positions would simplify things. And you should have a parameter where caller can tell how large buffers you are handed, otherwise things will crash at least with bad data as you will try to read or write beyond allocated memory.

Regarding the source file info, if you ever support more than two channels you should add a function that returns the channel mask used in the source wav.

Edit: Updated the foobar2000 decoder to use HALAC 0.2.8.
Other Lossy Codecs / Re: Quite OK Audio (QOA)... anyone ?
Last post by C.R.Helmrich -
I agree with all three of you. Hence my previous two posts in this thread, where the latter includes a solution for (relatively) simple but robust noise shaping.

Psychoacoustically, a high coding SNR is needed especially in lower frequencies. In case of a CDish sampling rate, you could achieve that by noise shaping during quantization, but with lower sampling rates, you can't. So in the latter case, indeed increasing the bit-depth is the only option I see.

But I don't think the QOA developer intends to change the QOA spec in that regard, am I right?

Wiki Discussion / Re: wiki 404 Not Found
Last post by ktf -
While you're at it, maybe consider updating Mediawiki? Last time I checked it was very outdated. I don't know how many of the updates are security-related, but still.