Old 25th September 2007, 18:43   #1
adam2434
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USB Audio Optimization

I apologize in advance if I'm not providing enough info. I'm well-versed in home A/V, but PC A/V is new to me.

When using Winamp, I use the configuration below to get the audio to my Pioneer Elite home theater receiver, which has an USB input.

Output plug-ins > DirectSound output v2.47(d) > USB Audio CODEC

USB Audio CODEC only shows up as an option when the receiver is connected. I assume that the receiver is essentially taking the place of an external sound card and is primarily just doing digital to analog conversion.

My sound card is a SoundMAX Digital Audio, built-in to the PC. Not sure if that's relevant when using USB for audio

This set-up works, but I'm wondering if it can be optimized for sound quality.

Is the internal sound card being used when audio is being sent through USB?

Is there a way to know what audio format is being sent through USB? In other words, should it be the native resolution of the source file (like a 320 kbps mp3), or is being converted to LPCM or something?

Are there any settings, codecs, drivers, etc. that would optimize sound quality over what I have.

Thanks.
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Old 25th September 2007, 20:06   #2
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What is this USB device that you are connecting? Surely the manual for it will tell you what kind of audio input it accepts.

However I would expect it to be recieving uncompressed PCM audio data as it acts like a sound card. The status tab on the DirectSound config should give you more info. This won't help if the driver for the device then recompresses the data before it is sent over the bus as Winamp cannot know this.

If winamp is using this as the soundcard it will not output to another one unless you are using an output stacker. I doubt you can really control the sound quality of this device from in Winamp. All you can do is maximise Winamp's quality.
  1. Use higher quality files. High bitrate files or lossless, such as FLAC.
  2. Enable 24 bit output if your device supports it.
  3. Make sure that the MP3 decoder is outputting full quality and don't use the Layer 3 EQ.

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Old 25th September 2007, 22:25   #3
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The device is a Pioneer Elite VSX-45TX home theater receiver. I know that it will decode PCM up to 24/96K, Dolby Digital, DTS...the standard home theater and home audio formats.

I was curious to know what is being sent out the USB - basically whether it's uncompressed PCM or the raw mp3 bitstream at xxx kbps. Since the DirectSound output status shows 44100Hz/16bits/2ch, is it safe to assume that's what is coming out of USB? So the decoder would be uncompressing the mp3 and therefor uncompressed PCM would be output from USB. Sound right?

Also, I do have the Nullsoft MPEG Audio Decoder set to full quality and EQ is on Logarithmic (default).

The driver that shows up is USB Audio CODEC, which I assumed was a generic driver already in Windows. There are no drivers to install for the receiver.

Thanks
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Old 25th September 2007, 22:26   #4
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Forgot to add - how do you enable 24 bit?

Thanks.
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Old 26th September 2007, 08:29   #5
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Prefs -> Playback -> Check "Allow 24 bit"

I would guess that it is raw PCM being sent over the USB connection as that is what it is recieving from Winamp/Windows.

About the EQ, you do hane both boxes "Fast Layer 3 EQ" and "Fast Layer 1/2 EQ" unchecked right.

I didn't know that there was a generic driver like this bundeled with Windows as I don't have any USB sound cards

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Old 26th September 2007, 16:20   #6
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Thanks again. Yes, Fast Layer 3 EQ and Fast Layer 1/2 EQ are unchecked.

One interesting thing that I noticed in DirectSound output status is that it shows the 2000 ms buffer to be 352,800 bytes. This would only be 176.4 kb/s in the buffer. However, this particular song was a 320 kb/s mp3. I'm not sure how to interpret this. DirectSound output status shows this as 44100Hz/16bits/2ch.

When I play an uncompressed track with a 48 kHz sampling frequency, the 2000 ms buffer goes to 384,000 bytes, which is higher by the exact ratio of 48 kHz/44.1 kHz. I need to check what happens when I play an uncompressed wav from a CD.

What happens to the audio bitstream in the decoder and USB is still a mystery to me.
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Old 26th September 2007, 16:40   #7
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pcm at 2ch, 44.1khz in 16 bit is 172kb/s

so thats all good.

what happens after that is upto your drivers and your amp.

if your not happy then I suggest using optical output of your current soundcard. quality will be the same if not better as it won't use the a/d converters on chip.
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Old 26th September 2007, 16:47   #8
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There is a difference between b and B, b is bits, B is bytes.
352 800 bytes = 2 822 400 bits
for a 2000 ms bufffer this is 1 411 200 bits/s which is the bitrate of uncompressed stereo 44.1/16 PCM data

The output plugin is just that, an output plugin, the ausio is sent through this to the driver. Most soundcards cannot handle anything but uncompressed PCM, some have an AC3 passthrough.

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Old 26th September 2007, 17:08   #9
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Guys, thanks for clearing this up. Yep, I was comparing bits to bytes. This makes sense now.

I don't have an external sound card per se. However, my Pioneer receiver has a USB input. Therefore, the receiver is doing the D-A conversion.

It's not that I have any issues with the sound. My goal was to determine if there was anything I could adjust or tweak for optimal sound.

One last question - what component/software converts the mp3 to PCM before outputing to USB?
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Old 26th September 2007, 20:36   #10
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in_mp3, the MPEG Audio Decoder plugin of Winamp.

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Old 28th September 2007, 01:51   #11
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OK, I guess that wasn't my last question.

So, the Nullsoft MPEG audio decoder plug-in decodes the mp3 into PCM. Is that the standard, or does the internal or external soundcard do this in some set-ups? I ask this because I have read that some external soundcards will convert mp3 to PCM. If so, what determines whether your system will output the mp3 bitstream or PCM from USB? Also, is the internal soundcard being used at all when the audio is sent out USB?

Also, I'm confused about the relationship of the MPEG audio decoder plug-in to the ffdshow audio decoder. In ffdshow, I can choose from 3 codecs for mp3, as well as "disabled". As an experiment, I chose "disabled" and I was still able to play an mp3. Maybe there is no relationship?

Lastly, what is the codec used in the Nullsoft MPEG audio decoder? Are there better codecs or decoder plugin-ins for this?

Being new to this, I trying to get a grip on how the audio is handled by Winamp and the PC and it's all kinda black magic to me right now. Home audio and D-A conversion... that's another story.
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Old 28th September 2007, 08:24   #12
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The soundcard cannot usually decode anything from inside the PC. There may be some special ones that can decode/encode from/to compressed formats if they need to do this. Having said this, I am not aware of any. If you specifically want one you will have to ask around to find one.

ffdshow has no effect on Winamp ability to play mp3 files as Winamp does not use DirectShow for decoding (thank god!). Winamp only uses DirectShow for video decoding and any other extensions you have associated with in_dshow.

All standard compliant MPEG audio decoders should produce nearly identical output (the standard defines how much it can vary I believe). The differences come from rounding errors and slight differences in the inverse transforms that they use.

To see information about in_mp3, hit the about button in the plugin list. It either uses Nullsoft's own code or uses code licenced from Thompson & Fraunhofer.

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Old 29th September 2007, 00:34   #13
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J_Darnley, thanks for the continued info.

The specific USB DAC I was reading about was the Harmon Kardon DAL-150. It does seem to be a unique product, in that it can decode mp3. I assume that this thing's driver/software must be able to tell the PC to send raw mp3 through USB. Supposedly, it's a very limited product and doesn't even work with Winamp. I have no plans to buy it...just doing some random research.

What are folks opinions on ASIO and the Winamp ASIO plug-in?

I've also read that audio through USB is more jitter-prone than SPDIF due to clocking issues. Would a USB-SPDIF adapter offer any improvements, or would it still being using USB clocking? My current PC does not have SPDIF out.
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