REPLAYGAIN tags are metadata stored in your audio files (same thing as Author / Album title / Track name / etc.) that contain information about (cannot remember properly HOW this works, but the principle is this...) the highest peak the track can reach, in terms of dB (or sound pressure / volume, as you may like); so when some software wants to play tracks that are at different (absolute) sound levels, it can readjust the volume of each track (you will lose bit-perfectness, of course...), making it lower (most of the times) so to have ALL the tracks peak out at the same apparent volume. Now, I am not a big fan of shuffle among albums of the same artist, let alone albums of different artists / genres - I am an old generation "pick an album and play it from the beginning to the end", so that REPLAYGAIN is nowhere near a necessity. It makes perfect sense, and IMHO is a must-have, if you are used to play different - sometimes very much different - tracks; in terms of volume, that is. HTH
Today, shuffle has regaled me with Elvis (both), Randy Newman, The Who, Wilco, Garland Jeffreys, Junior Sample, and Bruckner No.9. You have to embrace the chaos!
1: Using the volume encoder for selecting and scrolling through the library 2: a way to tell the difference between power on and power off. 3: Screen saver stays visible during power ON (extension of 2 4: DTS decoding from Flac containers 5: Atmos for HDMI output 6: 7.1 support for HDMI output 7: A true Transport mode. Analog off, volume settings invisible. 8: Replay gain scanner function 9: rotary encoder color selection. A color for on and a color for stanby 10: Logo's on album pictures which indicates codec/channel count 11: Preload option for Tidal. 12: Tidal Atmos 13: don't ask for WOL just WOL. Make an option to enable WOL
4. DTS wrapped in FLAC is a crime since FLAC is lossless but DTS (core) isnt't. Please don't support. Those are the same people who convert MP3 to FLAC and then distribute that garbage.
Hahaha putting it into a non compressible container is even a bigger crime. But you got a point there But to serious: Putting DTS in a flac container will give a file size of a CD. The alternative is extracting it to PCM that will be 3-4 times the size of a CD Based on the reported 32bits it will get even bigger than 3-4 times of a compressed CD.
To remain strict to the average compression rate of a FLAC file... 1 CD > 1/2 CD, so... DTS is normally (16-bit talk here...) 3 to 4 (5.1 or 7.1 as limits) times the size of a CD. Compressing it to FLAC would give 1.5 to 2 times a CD. Still a good ratio. BUT... Garbage. And dishonesty. Still. As pointed out by @Jorgo it is an abuse, and a silly thing to use a - notoriously - lossless container to store - notoriously - lossy audio. You wanna stay on the safe side? Convert everything to AAC: it is compressed almost at the same ratio, if not better, and you won't be breaking any "unwritten law". Don't clutter FLAC, please.
Why would you be able to lossless compress an stereo audio cd but not an DTS audio cd? It is the same redbook data. It only needs to be decoded by a DTS decoder. Shure DTS music disk is inferiour compared to SACD .
You can, but there is more data, that is, more channels, to be correct up to 6 or 8. This alone makes it 3 to 4 times the size of a regular CD. Compress it (max ratio nowadays it 1 => 1/2) and still get 1.5 to 2 full CD size. or, about 1.2 to 1.6 GB. Compressed. BUT, as I said, the main thing is that DTS is not lossless. So using MP3, or even better, AAC lossy compression you get the same quality as the original, but in less space. Much less space.
pffft It is a CD, so 74oMB. It is multichannel, it is compressed : it is DTS on a CD. It is sold as such. It is what it is. There is no need to compare, judge or convert (And as a cd you can put it in a flac or wav file. So lots of people did)
It is the CONCEPT that is wrong. You do not put lossy music into a lossless container. You put lossy music into lossy containers, such as MP3, OGG, AAC. But as far as the music is played within your walls, do whatever you like. Just bear in mind that you are doing it the wrong way.
The biggest request of them all for me is a backup/restore function solely for favorites. The user experience on this device for me is all about playing my favorite songs. It took a very long time to go through my music collection and set the favorites. While the database system on the DMP-A6 is pretty robust, it is not infallible and there are still some kinks being ironed out that require the database being reset. Of course, in that case all my favorites are lost and with it hours and hours of work. So a favorites backup is sorely needed.
The same goes for playlist I think. Also being able to edit a playlist (including sorting/randomizing) would be welcome.
You always need to have the "slave" device startup _after_ the master. So, if you stream from TV to the Eversolo, the TV MUST be powered on FIRST. If you are streaming from the Eversolo to your HT, then the Eversolo MUST be on BEFORE you turn the HT on. Bear in mind, this is kind of a "rukle of thumb". It may not always be mandatory. It has been for sure at the beginning. Nowadays it should be not, but as I see the behavior of many devices it is, IMHO, still a good practice... If this doesn't solve the issues, then there is some problem in the handshake: poor cable quality? cable too long? not adequate (just video cable / only audio cable / etc.) HTH
Thanks. But my observation is that switching sample rate results in a handshake. A handshake the player does not wait for. The player also seem to ignore the fixed sample rate setting. Internalplayer- hdmi - HT(av7704 ) Tested with dts, flac 5.1, dsd64 5.1 Only on stereo the fixed sample rate seems active
The player is not the one ought to wait... as it is IT that changes the SR. The destination device should comply... Or, I didn't correctly get you / you didn't correctly express yourself. The fact that only STEREO works with fixed SR, seems to me that the other multichannel layers are either unsupported (in terms of handshake) or they do not support fixed sample rate resampling at all. This could be an interesting question to our fellow master @Zidoo Support-Kim See if he can chime in and clarify things.
HDMI is like a tango it takes to to get it perfect. When there is no change in freqency (bit width????) it all runs smoothly. The Zidoo z3000 has an option to wait a certain amount of time to give the hdmi recipient the time to adapt to the new stream. In that time the zidoo waits for the hdmi to be stabilised/initialised/(whatever) In case of a sample rate change it would be nice if the eversolo waits aswell. The other part of this question is why is there no fixed samplerate output for multichannel? Then this problem would only noticeable at the start of a playlist.
No. When switching music at different sample rates, the streamer negotiates with the AV decoder for a handshake in real time. Fixed sample rate setting is valid for PCM2.0 output. Thanks for your trust. Shout out to the guidance of our tech team. BTW, I’m a “she” not a “he” : )
