I currently use a Roland Bridge Cast, and while it gives me all of the features I need, it genuinely sounds worse than something like a SMSL C200 or even a FiiO BR13. (and if anyone is wondering, the XLR input has noise on certain gain values for some reason). Those 2 both have a smoother sound, almost like the Bridge Cast has a bit of distortion. The biggest difference I noticed though, was that the Bridge Cast's directional imaging in FPS games is somehow worse than the other devices. It's still very playable, but I can hear a clear difference. For example: The Bridge Cast lets me know a general direction of enemy footsteps, but the other devices give nearly pinpoint accurate directionality of them. For a device that costs more than the others combined, it's kind of disappointing. I don't really know how to word this, but what measurement metric would I be looking at to determine the DAC's ability to have normal good stereo imaging?
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What causes a DAC to have bad directional imaging?
- Thread starter toasty
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I Googled the Roland. Very nice device. Firstly we would be assuming you are not imagining stuff... Frequency response would be a go to but it's a DAC and they have flat frequency response. Output impedance might be a factor but I doubt that too.
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I thought I was imagining stuff myself, so I did a few quick tests by capturing a few different gameplay audio clips that had loud, medium, and faint audio cues then playing them back with all the devices. I tested at roughly the same volume (even though I most likely did not match the loudness perfectly during that test), as well as trying each device being louder than the others and then quieter than the others. I made sure all processing was disabled when doing the tests as well. I really couldn't hear the difference between the C200 and the BR13, but the difference was noticeable for me at every volume level when switching over to the Bridge Cast. I can't find the video anymore, but I did see a review where the person had the same opinion in that other audio offerings, (in the reviews case the Motu M2 and Audient Evo 4) had better audio directionality, which made them return the Bridge Cast. That is what sparked my curiosity on if something was somehow "worse" in the device. I also reached out to Roland to ask if they were able to tell me what DAC / Amp / any type of chip that was used so that I could get a better understanding on a rough estimate of the performance of the HP out (I know implementation also matters), but they basically just said they couldn't tell me.
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- #5
I've only been in the Audio Hobby for a few years, and only a few months into learning terminology, understanding measurements, and more in-depth stuff like that. But my first thought was about if something like crosstalk could be causing it, but it really doesn't feel like stuff is leaking to the other channel. It feels like if the audio was a pie, the bridge cast has 8 "slices" of directionality, and the other devices have 32+ in addition to having a somewhat noticeable "clearer" / more "open" overall sound. (I dislike those terms though as I usually think the whole "DACs sound different" thing is a bit of a stretch sometimes)
Can you share clips demonstrating good vs bad behaviour?
I totally get your feeling. My Kenwood L-08c/L-08m had the same impression of slices of directionality and that was just an analog setup.
But if you want real answers to your question, you will need to measure yourself. We can help you if you are willing to spend the money on a E1DA Cosmos ADC or even any ADC of decent quality if the difference to your ears is large enough.
Otherwise it’s people guessing and you also guessing if they are right or wrong based upon nothing.
I was hesitant to buy the E1DA since the ADC offers no “improvement” but instead it has been one of the best products I have ever had, allowing me to sort out my audio preferences through objective means.
Take a look at the articles I have started looking at stuff like DACs and doing stuff like recording the output of different systems and then ABX’ing them or doing null comparisons. That’s what you need to do. We cannot do it for you, but we definitely can help you through the relatively short learning curve of using an ADC to test your own line level gear.
DVDdoug
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A properly functioning DAC won't affect the sound. And it's not easy for electronics to "accidently" affect imaging and soundstage.
Directional perception is tricky because it's obviously an illusion with the sound actually coming from a pair of speakers or headphones. More (surround) speakers can be a lot more realistic. If you want to hear footsteps coming from behind you, rear speakers are the obvious answer!
The big factors are the recording (or game audio), speakers and room acoustics, and your brain. It's one of the few things that can't be measured.
With headphones, it turns out that most people perceive the sound coming from inside their head!
What is a blind ABX test?
Most "audiophiles" are nuts and most of what you read is nonsense and there's a lot of undefined-nonsense terminology. This is one of the few rational-scientific resources. There's even some nonsense and mythology in the pro audio world.
Check out Audiophoolery by Ethan Winer. He describes the REAL characteristics that define "sound quality" - Noise, distortion, frequency response, and time-based errors. Only the first 3 apply to electronics. Time based errors relate to speakers and acoustics.
If you are interest enough to buy a book or two -
The Audio Expert by Ethan Winer is good, or Sound Reproduction by Floyd Toole. Both experts/authors have some YouTube videos and there is an Audio Science Review YouTube channel.
Directional perception is tricky because it's obviously an illusion with the sound actually coming from a pair of speakers or headphones. More (surround) speakers can be a lot more realistic. If you want to hear footsteps coming from behind you, rear speakers are the obvious answer!
The big factors are the recording (or game audio), speakers and room acoustics, and your brain. It's one of the few things that can't be measured.
With headphones, it turns out that most people perceive the sound coming from inside their head!
VERY common!
It looks like you are missing two things - Blind listening (not knowing what you're listening to) and repeatability.
What is a blind ABX test?
That can be "dangerous".
Most "audiophiles" are nuts and most of what you read is nonsense and there's a lot of undefined-nonsense terminology. This is one of the few rational-scientific resources. There's even some nonsense and mythology in the pro audio world.Check out Audiophoolery by Ethan Winer. He describes the REAL characteristics that define "sound quality" - Noise, distortion, frequency response, and time-based errors. Only the first 3 apply to electronics. Time based errors relate to speakers and acoustics.
If you are interest enough to buy a book or two -
The Audio Expert by Ethan Winer is good, or Sound Reproduction by Floyd Toole. Both experts/authors have some YouTube videos and there is an Audio Science Review YouTube channel.
somebodyelse
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And accurate level matching. It looks like you missed three things...
Keith_W
Master Contributor
Guys, you should take a closer look at his device - the Roland Bridge Cast. The OP does not come across to me as a typical subjectivist who is hearing things that do not exist (and welcome to ASR by the way).
It looks as if the Bridge Cast is a mixer / resampler / DSP device. On that page, it boasts that it has DSP capabilities such as a de-esser, low-cut filter, virtual surround sound, and possibly others.
IMO it is totally conceivable that some kind of DSP is interfering with the imaging that the OP is expecting to hear, especially "virtual surround sound". At this point all I can say is to read the manual and make sure all the DSP settings are turned off. Then listen again and see if the problem persists.
The DSP settings are supposed to improve your experience, but if they are not set properly they can certainly negatively affect the sound. You will need to experiment a bit to find the best settings.
You don't need to purchase any expensive measurement devices. Your Bridge Cast has a mic input. So all you need is a loopback cable (in your case, TRS jack on one end, XLR male on the other end) - send the audio output into the mic input. Run a test signal through the right channel, then through the left channel, then compare them. You could use REW to record the left/right channel and then invert one of the channels, then add them together. The result should be a flat line (i.e. it nulls out). If they DON'T null out, then either you have some DSP setting that you haven't turned off, or your device is faulty.
Test equipment like the E1DA are for testing below the limit of audibility. If you think you can hear something, then a mic input will be more than adequate. Provided you can get it to work, of course.
It looks as if the Bridge Cast is a mixer / resampler / DSP device. On that page, it boasts that it has DSP capabilities such as a de-esser, low-cut filter, virtual surround sound, and possibly others.
IMO it is totally conceivable that some kind of DSP is interfering with the imaging that the OP is expecting to hear, especially "virtual surround sound". At this point all I can say is to read the manual and make sure all the DSP settings are turned off. Then listen again and see if the problem persists.
The DSP settings are supposed to improve your experience, but if they are not set properly they can certainly negatively affect the sound. You will need to experiment a bit to find the best settings.
You don't need to purchase any expensive measurement devices. Your Bridge Cast has a mic input. So all you need is a loopback cable (in your case, TRS jack on one end, XLR male on the other end) - send the audio output into the mic input. Run a test signal through the right channel, then through the left channel, then compare them. You could use REW to record the left/right channel and then invert one of the channels, then add them together. The result should be a flat line (i.e. it nulls out). If they DON'T null out, then either you have some DSP setting that you haven't turned off, or your device is faulty.
Test equipment like the E1DA are for testing below the limit of audibility. If you think you can hear something, then a mic input will be more than adequate. Provided you can get it to work, of course.
Good point! Since that has an input, you’re in good shape.
The one nice thing about the E1DA is it’s really poor input impedance which actually can maximize differences. Something subtle like a downstream amp not having ideal input impedance and those DACs not having ideal output impedance may result in something audible but not easily measurable unless you are measuring the downstage element.
Either way, very much can answer the question with some sort of measurements.
- Thread Starter
- #13
From my relatively small time in the audio world, I have indeed seen more snake oil products and nonsense terminology than factual evidence / measurements. However, since I've been in the competitive gaming space for nearly a decade, I like to think that I can mostly sort out real from fake when it comes to products and information. In the competitive gaming space people believe the most unrealistic things that they think will give them less input latency and/or higher framerates. These people never do real tests or show proof of claims but are still able to pass off the fake information to thousands of other people. Sometimes it's just outright lies that do nothing and sometimes things actually hinder performance. It even happens to e-sports professionals and between companies and the people they sponsor.That can be "dangerous".
Check out Audiophoolery by Ethan Winer. He describes the REAL characteristics that define "sound quality" - Noise, distortion, frequency response, and time-based errors. Only the first 3 apply to electronics. Time based errors relate to speakers and acoustics.
I also appreciate the link to the Audiophoolery article, as it basically confirms what I've been thinking about a lot of the information that I've been gathering over the months.
Yeah most likely, but I enjoy learning about how and why things work. I'll probably replace it regardless though for peace of mind and because some of the features of the device aren't really done well.
Appreciate it.
The de-esser, low-cut filter, compressor, and noise suppression are a part of the mic input's DSP capabilities. However, the de-esser just muffles everything, and the noise suppression is just awful. I dislike virtual surround sound anyway, but their implementation of that is also bad. I don't use any of them though, so it doesn't bother me... The gain slider on the mic input also causes white noise on certain values for some reason. Seeing as they seemed to barely put any effort into the features, and the fact that they don't want to share any details regarding the internal components kind of leads me to think that the dac and amp portions aren't very great either.
Thank you for teaching me something new.
I also double checked that I have every form of DSP disabled on both the device and within Windows 11. I don't know for sure if this is what you meant and if I did it correctly, but here's what I measured in REW. (L&R Channels are overlapped and the Left channel is inverted)
Keith_W
Master Contributor
Doesn't sound like a high quality device.
Thank you for teaching me something new.
If you inverted one channel, then the graphs would diverge. Those graphs look exactly the same. Regardless, it looks as if there isn't much difference between the two channels. Can I confirm that you used an actual cable to connect the output from the jack to the XLR mic input?
Also, is the problem still audible?
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Sokel
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You can use this silly guide and a loopback as @Keith_W is suggesting to dive a little deeper to it:
www.audiosciencereview.com
Sometimes its not only the FR that tells the whole story.
Horrible Crackling from Topping E1x2
Hi! I lurked on this forum a lot when deciding on which interface to buy, and I ended up deciding on "upgrading" from my rudimentary Behringer UM-2 to a Topping E1x2. The first day, it worked great! The next day however, I noticed a horrible crackling noise when I was playing a game and...
www.audiosciencereview.com
Sometimes its not only the FR that tells the whole story.
- Thread Starter
- #16
The only thing I could find relating to inverting in REW was the 'Invert Polarity' button in the Actions tab, which is what I did for one channel. How do I actually invert it the way you intended me to do?
Well... it's a cable and some adapters, as I don't currently own a 3.5mm to XLR cable (going to buy one though). I cleaned all the connections on every cable and adapter before testing, but the current setup of it is HP Out -> 3.5mm Aux -> 3.5mm to 6.35mm adapter -> 6.35mm to XLR adapter -> XLR Mic Input. I can't verify for sure if the adapters cause any measurable issues, but they work without problems or noise on the devices I was using them with before.
Running the 1khz tone at the highest output volume before the mic input starts clipping, I get this:You can use this silly guide and a loopback as @Keith_W is suggesting to dive a little deeper to it:
Horrible Crackling from Topping E1x2
Hi! I lurked on this forum a lot when deciding on which interface to buy, and I ended up deciding on "upgrading" from my rudimentary Behringer UM-2 to a Topping E1x2. The first day, it worked great! The next day however, I noticed a horrible crackling noise when I was playing a game and...
Sometimes its not only the FR that tells the whole story.
And running it at the volume that I regularly listen at I get this:
Sokel
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Considering the levels at play results are decent.
It would be much better i suspect using WASAPI exclusive from the soundcard options (for both I/O) and testing different gain structure amongst the devices.
Also add some more dither at the generator (24-bit is fine) .
But even as is, no problem at all.
- Thread Starter
- #18
Is it supposed to have more of a curve in the lower and higher frequency areas when using exclusive & with the dithering vs the previous test? I thought it was supposed to be mostly flat in the other areas since only a 1khz tone is playing.
Sokel
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Yes,this looks a lot better, the curve is probably caused by low input impedance, it's common with mixers and interfaces.
But it draws a truer image.
You can also limit the view by the arrows to 0-160dB or even 150dB for the y-axis for better visuals and to align better with what we use to see.
Keith_W
Master Contributor
How to do a null test in REW ... I am embarrassed to say that I do not know how. I know how to do it in Acourate (which is what I normally use). In REW I thought you do the inversion with Trace-Arithmetic 1/A, and then the convolution with Trace-Arithmetic A*B. I tried it with some spare measurements that I have. But for some reason it gives me a different result to Acourate. In Acourate, it nulls out. In REW, it doesn't. I'll have to ask a REW expert for help on this, very sorry.
In any case, the cable connection you made is fine.
The THD+N measurements are not great. We don't know if it's the output or the input since it was measured with a loopback cable, but it should still be below audible limits. I am starting to think that we won't be able to get to the bottom of why you are hearing a difference with this testing. You might have to send the device to Amir for testing.
In any case, the cable connection you made is fine.
The THD+N measurements are not great. We don't know if it's the output or the input since it was measured with a loopback cable, but it should still be below audible limits. I am starting to think that we won't be able to get to the bottom of why you are hearing a difference with this testing. You might have to send the device to Amir for testing.
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