I've seen many Bluetooth products measured and reviewed in these pages. Bluetooth headphones all rely on compression and they are the fastest growing segment in audio, so compression is history?
I mostly used open reel for air checks. Met opera broadcasts and jazz shows from the local (low power) college station. As I recall (doing this from memory, and it was many years ago) my B77 four track allowed individual programs for each track, so you could get a lot of time on a large reel. At 3 1/4 ips it was suitable for FM mono broadcasts (stereo reception was never guaranteed because I was a long way from the tower).Unfortunately all of the tape copies I have from that era were recorded on Ampex 456 'Grand Master' tape, and are now totally unplayable because of sticky shed syndrome. I tried baking a couple but they were too far gone, and the music wasn't all that great to justify all the work.
Yes. Those products are now legacy for Hi-Fi use with the arrival of aptX Lossless for mainstream chips.
Come on. it's just announced, no products support it. even AptX HD and LDAC is just a tiny fraction of the market... Wait a bit for "legacy" Now I do use lossless when I am connected to wifi and my transducers are linked to the source with a wire... But there is plenty of use cases where it is convenient not to, and honestly, the difference between lossless and a good compression algorithm at a decent bit rate is tiny at best, indiscernable for many. there is so much more important stuff in the chain to worry about, starting with the transducers... Compressed music can sound very good, when you stop looking at numbers and just enjoy. It isn't the bottle neck of any reproduction chain.
I am not in a rush to buy a new BT device to the urgency that I will buy one with no support for uncompressed audio transfer. Are you?
I disagree vehemently but everyone to themselves.
There are a few differentiation tests between lossless and various bitrate compression available on the net. Some will pass some will fail to distinguish, but anybody that find that it's night and day and one is enjoyable vs the other bad sounding (for 320kbps+) would be of bad. faith. the fact that I'm writing that on a thread about R2R which many are able to find good sounding is a proof of that.
Ha! You do realise that you are on ASR, right? We differentiate SINAD between -123 and -121dB...
Sure, if you say so.
as we say, looping the loop. I was talking listening experience, you are talking numbers.
Here is a view of the mixing console. I made this from scratch - the circuit design, etching the circuit boards, assembling the module chassis, the woodwork, cutting the front panels, and even silk screening the text to the front of those panels. Everything. It took me almost a year to construct, but I learned a hell of a lot in the process.
That is a lot of work. Bravo!Here is a view of the mixing console. I made this from scratch - the circuit design, etching the circuit boards, assembling the module chassis, the woodwork, cutting the front panels, and even silk screening the text to the front of those panels. Everything. It took me almost a year to construct, but I learned a hell of a lot in the process.
View attachment 156384
Oh its long gone - unfortunately. I sold it when I closed down the studio. Its a split console with a separate 8 channel monitor section, 4 effects send busses, 16 input channels and 8 output summing busses. It had 4 channel monitoring output capability in case I wanted to do surround monitoring, but I never used it and only monitored in stereo. It was predominately designed around the National Semiconductor LM 381 opamp, one of the first truly low noise ICs available. The EQs were 5 fixed bands and used LCRs in the feedback loop. I didn't have much in the way of measurement equipment when I built it, but I do remember that it was extremely quiet and had a lot of headroom. I miss it, but realistically it weighed a ton and more capability is crammed into even the simplest DAWs nowadays.
LCRs in the feedback loop are cool. How were the microphone amplifiers implemented?Oh its long gone - unfortunately. I sold it when I closed down the studio. Its a split console with a separate 8 channel monitor section, 4 effects send busses, 16 input channels and 8 output summing busses. It had 4 channel monitoring output capability in case I wanted to do surround monitoring, but I never used it and only monitored in stereo. It was predominately designed around the National Semiconductor LM 381 opamp, one of the first truly low noise ICs available. The EQs were 5 fixed bands and used LCRs in the feedback loop. I didn't have much in the way of measurement equipment when I built it, but I do remember that it was extremely quiet and had a lot of headroom. I miss it, but realistically it weighed a ton and more capability is crammed into even the simplest DAWs nowadays.
And yes, it was a shit-wad of work!
I used 1:10 ratio microphone input transformers from Triad which had 10dB and 20dB pads ahead of it, along with 6.8k resistors furnishing +48V phantom power. This fed an LM 381 which had its gain varied between 20dB to 60dB via the feed back loop. This fed a largely passive matrix for channel assignment. The faders I bought from Tascam.
You think audio is all I do?Who said Hi-Fi is cheap?
However, even an entry level phone (64GB) can store 4,000 hours (100 albums) worth of uncompressed music. I don't think you have a case for compressed even on a mobile device.
Clever design. Transformer amplification vs active amplification topology was behind the famous desk sound debate of Neve vs (early) SSL. Neve used transformer while SSL used a parallel transistor pair in front of the op amps. Neves were universally liked while SSLs were found to sound harsh.
