At what bitrate would you consider Opus universally transparent?To me 128 Kbps opus is Transparent for 5.1 tracks and 64 Kbps is transparent for 2.0 tracks
I went for 160 kbps on my whole library for that reason, didn't want to have to encode twice.
Denon D7000, by far. Not only are they detailed and reveal differences between, say 320 kbps MP3 and FLAC, quite easily, they are extremely musical and pleasing to the ear. When used with proper equipment (not a phone or integrated sound card), the sound is big, warm, engaging and cohesive. I use them with a Abrahamsen DAC and Vincent hybrid vacuum tube amplifier.
For listening on the go, I use Sennheiser Momentum 2. For this purpose, they are excellent phones, as they can be driven quite easily by an iPhone and the sound is still full and cohesive.
I also have Sennheiser HD650 and Beyerdynamic DT770 (80 ohm). They sound muddy and closed in comparison. As for Sennheiser HD800, they are extremely detailed and have a huge sound stage, but they sound thin, analytical and less engaging to my ears. To get any bass at all you have to play ridiculous high volume levels. They do not play music the way it's meant to be heard. They are however very useful as a tool to reveal differences in formats and bitrates.
3rd Party Plugins - (fb2k) / Re: JScript Panel 2.0 and Powershell to extract cover art from youtube or audio fileLast post by MordredKLB -
Ever since Sony introduced their Street Style/behind-the-neck headphones (I think this was 2000), I've only used those. They're cheap (MDR‑G45LP around $15), light and you can wear them with a bicycle helmet. I just wish the cord was on the right side. I cannot speak for quality because that's all I've used.
Seems like a similar bug. Same optimizer in a newer form in VS2017 after all. In a typical Mozilla fashion their solution seemed to be to fuck everyone who doesn't have a new machine and just require SSE. Peter's solution is to compile the problematic code with /d2SSAOptimizer- parameters to get working code until MS releases a fixed version.
HD800 for home, AKG K7XX at work. Various IEMs for mobile but CustomArts CIEM mostly.
Last post by [JAZ] -
Ok, here are my answers:
1) The default most probably depends on the driver. Integrated soundcards usually set the default to 48Khz.
Now, let me explain what this default option is:
Since Windows Vista (where WASAPI first appeared), the "MME/WaveOut" and Directsound APIs became just a wrapper on top of WASAPI.
- WASAPI, by default, works at the default sample rate, and sources from MME and Directsound get automatically resampled by the wrappers to the WASAPI default sample rate in order to get mixed.
- WASAPI shared-mode sources do not get automatic resampler. Concretely, it's the application responsibility to resample to the default samplerate and use the system-determined buffer sizes/latencies. Then, WASAPI mixes them as-is and sends it to the soundcard.
- For WASAPI exclusive-mode sources, WASAPI mixing gets bypassed, and the application takes responsibility to negotiate with the soundcard what samplerates and bit depths it supports and to use one of them.
Note: At the beginning, there was a problem with this resampler especially when resampling from MME/Waveout, and Microsoft released a patch in order to improve the quality of the resampling. Nowadays, the resampling of those sources is good, but several factors can affect the final result, so comparisons always need to verify that everything is setup correctly.
2) Differences shouldn't be obvious, since usually, there shouldn't be any, other than different roundings.
In case of foobar 2000, I should mention that it uses WASAPI-exclusive, so that's why it blocks other applications, but that also means that the soundcard receives the audio at the sample rate and bit depth of the source (if supported).
I use Foobar2000 with directsound also to allow other applications to access the soundcard. With some other applications I've been using WASAPI shared for the low latency: ( I could have 20ms on a 10 year old core 2 duo, and can have 10ms on a core i7, but I can have even 1ms with ASIO with the i7).
3) oh wait. there's no 3..
4) I would say yes. If that is the sampling rate of most of your audio, why not? That means that there will be less resampling steps.
5) I sort of explained this on point 2. WASAPI output on foobar2000 can send the audio directly to the soundcard without additional resampling and changing bit sizes. That doesn't imply that it will sound better, because the benefits might be simply theoretical.
It looks like the frequency content is mirrored above the ~28-29kHz tone.
I got a pair of TFZ Exclusive 1 from Massdrop. They're pretty good.