Jump to content


  • Content Count

  • Joined

  • Last visited

About yamamoto2002

  • Rank
    Freshman Member

Personal Information

  • Location

Recent Profile Visitors

3424 profile views
  1. I think 10 to 30 years old Goldmund Alize DACs do not have digital oversampling filter. Instead, it has super steep brickwall analog low pass filter of 20 stage or something to cut all noise (aliasing noise which is inevitably generated by non-oversampling digital circuit) above 20kHz. IIRC there is an analog all pass filter circuit cascaded in order to compensate phase distortion caused by analog brickwall filter. It seems output analog signal is clean, similar to the conventional oversampling dac.
  2. WASAPI shared mode do nothing to PCM data when There is only one audio session coming into the mixer Original PCM bit depth is 25bit or smaller (theoretically, I verified up to 24bit using S/PDIF digital out and digital PCM recorder) All Sound enhancements are turned off Session volume and endpoint volume are both 100% Original PCM magnitude does not exceed 98.5% of full swing Shared mode sample format bit depth is set to 24bit or 32bit (not 16bit, where dither is applied) Even bit perfect playback (original PCM values are outputted unaltered) is possible in WASAPI shared mode when those condition is met. Limiter APO temporally reduces gain when incoming signal exceeds 98.5% but some times after incoming signal becomes smaller 98.5%, gain returns 1 (I don't remember time scale but it is smaller than 1second) Sound distortion occurs when Limiter APO is activated and its distortion level is up to 0.13dB, some people in Japan reports they noticed sound difference compared to original signal but they said difference is subtle. I cannot hear the difference Resampling artifact may become larger problem than mixer (Limiter APO) distortion, it is important to adjust shared mode sample rate to match to original PCM sample rate on shared mode playback for critical listening (this is true when resampling algorithm is poor). I don't know implementation details but limiter APO behavior should be the same across all type of CPUs CPU difference do cause sound difference when CPU computing performance is relatively low and other tasks run while playback: other tasks take CPU time and playback thread fails to deliver PCM signal on time: this causes sound stuttering like vinyl needle jump. This is always very obviously heard as annoying playback failure when it happens. This happens more frequent in Windows 2000 era and Windows Vista era of computer and things gradually improved
  3. I'm curious what Tsimané people do when a man and a woman sing the same song in unison 🤔💭
  4. Windows 10 privides several different sample rate converters to resample PCM signal for apps, with different conversion quality. Also apps may have own sample rate converter. I don't know Amazon HD uses which API to play sound, but SRC can be bypassed when source sample rate and shared mode sample rate is matched. Shared mode's sound altering filters can be disabled on Enhancement tab of Sound Control Panel (Fig. 2). When SRC is bypassed, remaining problem of altering sound of Windows shared mode is Limiter APO, which reduces gain temporary when incoming PCM is too hot. It starts to work when incoming sound magnitude exceeds 98.5% of full swing. Windows provides these sample rate converters: MME Resampler: Used when App play sound using old MME API. I don't check its conversion quality of Windows 10 versions. Direct Sound Resampler: Short-delay (not linear-phase) resampler. Conversion quality is low. Media Foundation Resampler: Latest converter of Windows. Linear-phase. Conversion quality can be controlled by app. App's own resampler: Music player apps may have their own resamplers for their needs. I believe all SRCs of Windows 10 provides use CPU. Most app's own SRC use CPU, small number of apps use GPU to perform SRC. Also there are audio hardwares such as Lynx AES16e-SRC that contain hardware sample rate converter chip. Fig.1 Fig.2 Enhancement tab of Sound Control Panel
  5. These OS setting is used on “shared mode audio session” such as You Tube, Spotify, Skype etc. It is not used on “exclusive mode audio session”, apps (not OS) can change DAC sample format, as Mr. Rubinson said. 16bit settings: Limiter is applied when mixed signal magnitude exceeds 98.5 %. Dither is applied (16bit 2 LSB TPDF dither, even when input data is 16bit full scale signal and dither is not necessary) and 24bit PCM is truncated to 16bit. 24bit settings: Limiter is applied when mixed signal magnitude exceeds 98.5 %. Dither is not applied. So 24bit settings output signal is slightly more cleaner than 16bit. And 24bit is recommended even when playing 16bit PCM. I recommend 24bit 48000 Hz for You Tube watching, 24bit 44100 Hz for local music file playback. ---- If app uses Windows Audio Session API directly to play sound using shared mode audio, Surprisingly Window Audio Engine does not provide automatic resampling feature: resampling to shared mode sample rate should be performed by app. If app uses Media Foundation API to play shared mode audio, OS automatically resamples to target rate using Media Foundation resampler with quality=30. its conversion quality is reasonable (not best) and it is sharp roll-off linear phase. If app uses DirectSound API to play shared mode audio, OS uses slow roll-off linear phase resampler (conversion quality is lower than Media Foundation resampler, I think this is to improve throughput, reduce delay for better user experience of game sound effect). If app uses MM API to play shared mode audio, OS uses sharp roll-off linear phase resampler, and its conversion quality improvement patch is released on 2013 or 2015 ? I don't remember
  6. Dither can be applied to 1bit SDM processing pipeline and it is sometimes used. Dither worsens SNR so other methods are often used for reducing idle tone problem.
  7. Sound quality is not so different, I like Supra for cable thickness and Zonotone for its vivid color cable skin. My friend preferred Wireworld. ifi Galvanic 3.0 works properly when I'm lucky Hi-Speed 5m cable is not suitable for audio use, once in a while stuttering happens. I use generic Hi-Speed certified USB cable 2m for reference (pictured below). It always works. My other USB cables.
  8. In Japan, people uses Roland UA-4FXII for this purpose (Mix PC audio and microphone with voice alteration effects). If you are old man and you'd like to get female voice, 恋声 can be used https://youtu.be/HCJEtdOrI4Q?t=329 https://www.youtube.com/watch?v=0Pu2EwMCv9c
  9. Thank you, I understand what people said. 1bit audio SDM digital signal processing output signal is more analog-circuit friendly (it is what people emphasis) and analog parts count of SDM DAC becomes more small.
  10. IIRC the problem 2 is called `idle tones problem', which happens on normal SDM, when input signal amplitude is very low and it decreases SNR. Trellis SDM, for example, mitigates problem 2, 3 and 5. It is mathematical, digital computation method to produce 1 bit digital output. Some people say 1 bit audio is more analog than PCM. I disagree, 1bit is more digital from the very first step when Inose added a digital circuit to improve SNR of his ADC in 1962.
  11. I agree, it is needed to have “métier” to produce an art.
  12. There is more appropriate forums, such as gearslutz for this kind of question. Maybe enabling pad switch to reduce input signal magnitude by 20dB and cracking problem will be solved. Or reverting driver to version 1.0.4, and check if your hypothesis is correct or not.
  13. Onboard audio codec chips are connected to south bridge with PCIe. On X570 Aorus Xtreme motherboard, Realtek ALC 1220VB audio codec is used. This audio codec connects to ESS Sabre DAC. ALC 1220 uses "High Definition Audio" protocol. This High Definition Audio protocol for PCIe audio is equivalent to USB Audio Class protocol of USB DACs. HDMI audio also uses High Definition Audio protocol. https://www.intel.com/content/dam/www/public/us/en/documents/product-specifications/high-definition-audio-specification.pdf Studio PCIe/PCI cards such as RME Hammerfall, Echo Layla 3G, Lynx AES16e etc has proprietary protocol.
  14. You can choose camera position, camera direction, lens focal length (field of view), focus and depth of field after taking the picture!
  • Create New...