Creative AudioPCI ES1370 - AK4531 tweaking

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Tweaking Creative AudioPCI sound card
(Ensoniq ES1370, AK4531)

Tweaking the best results out of whatever u got is a fun thing to do, despite the risk of destroying it. Improving the quality by your own effort is more exiting than just spending some money for buying a new better one. OK... thats enough advice part. And I wanted to hear a little cleaner sound from my poor sound card... bravely taking the risk of loosing it... although i had no money to buy a new one. I took a philips head screwdriver and walk (in slow motion) towards my computer :) pull the card out... as expected.. i found a cheap opamp; JRC4565 on board along with Asahi Kasei AK4531 (codec) and Ensoniq ES1370 PCI audio controller. With the help of a multimeter I confirmed that the opamp JRC4565 is exactly in the output path.. and it must be replaced with a good one to hear the DAC better. Took an AD8599 which i've recently 'free sample'ed from the Analog Devices and soldered to the board replacing JRC4565. First stage is over.

Fig.1. Original JRC4565 opamp - Fig.2. Replaced with AD8599
Note: The soldering process was a little brutal...
not any standard procedure for SMDs. so not shown here to the public. :)

Fig.3. The DAC connection.
Going through the datasheet of AK4531.. it is found that the chip also have pins for audio output straight from its 16bit DAC's, bypassing the internal gain, mute, mixer and attenuation stages. ie. less distorted sound.. but no volume control by software and unable to hear other sources like mic, line-in, audiocd..etc. I don't want neither of them... So cut the two tracks carrying 'normal' right and left outputs (pin 27, 28) from the codec to the opamp. soldered two thin wires to the right and left DAC outputs (pin 23, 24). Then solder the other ends of the wires to the cut tracks going to the opamp (not to the lines coming from AK4531). In simple words connect the opamp input to the codec's 'DAC output' pins 23 and 24 with wires... and leave the mixer output pins 27 and 28 cut open. huhhh...

Fig.4. New opamp and direct DAC output connections.
Note: THIS TWEAK WORKS ONLY WITH LINUX ALSA DRIVER. Doing this will make your sound card silent with Windows operating system. This might be because the windows driver using the alternate 18bit stereo DAC inside the codec.

Fig.5. Two 0.47uF polyester coupling capacitors.
The next and last thing to replace is the two SMD ceramic coupling capacitors in audio channels between AK4531 and the opamp (the same lines which are previously diverted to the codec's DAC output pins). I tried bypassing them.. but there is more than 3V DC comes at the output. so remove the ceramic caps and place 0.47uF polyester caps instead. This improves bass and audio quality. Now there is frequencies as low as 5Hz coming out from the card moving the woofer cone back and forth. ie. the low frequency path is free. ceramic caps are the worst choice for audio coupling and they are piezo-electric, (i think) they produces noise from vibrations coming from cooling fans and such other sources.

And there is a problem... the sound card uses a strange samplerate of 47040Hz to feed it's 16bit dac. There is a 22.5792MHz crystal on board which is usable for both the samplerates... 44.1kHz (22579200/512) and 47040Hz (22579200/480). so the 18bit DAC will be working at 44100Hz... the chip supports independent samplerate for each DAC.

using a good software sample rate converter solved the problem:

$ sox -V "louis armstrong - what a wonderful world.mp3" -r 47040 -w -s -t alsa default polyphase

The Results: believe me... there is big difference... the improvement in the audio quality is very much audible... it is a lot cleaner now. You can trust my ears.. since i've worked as a sound recordist in a studio for television programs.. the most important part in my job was to listen to the sound being recorded and find imperfections in it... and did it continuously for 2 long years...

Sox's sample rate conversion gives pretty good results... it produces no considerable distortions. You can hack the ALSA driver for the card as follows if u really wanna get rid of the 47040Hz samplerate problem:

Download the latest linux kernel source code from www.kernel.org , apply the following bug, compile and install.

In the kernel source tree, find the following structure in the file : ./sound/pci/ens1370.c

static struct snd_ratnum es1370_clock = {
	.num = ES_1370_SRCLOCK,
	.den_min = 29, 
	.den_max = 353,
	.den_step = 1,
};

Set '.den_min = 32', instead of 29. which in turn limits the maximum samplerate to 1411200/32 = 44100. (ES_1370_SRCLOCK = 1411200). ES1370 have an internal programmable clock divider which can be set through the PCLKDIV bits of its control register. So there is no fixed sample rates... and the ALSA driver default went up to 47040Hz..!

Here is the modified version:

static struct snd_ratnum es1370_clock = {
	.num = ES_1370_SRCLOCK,
	.den_min = 32,  /* orig. = 29 */
	.den_max = 353,
	.den_step = 1,
};

Thats all... save the file and compile and install the kernel as usual. When u boot with the fresh kernel... ur soundcard will be ready for 44.1kHz music..!

To see the current samplerate, do:

$ sox -V file.wav -t alsa default

$ cat /proc/asound/card0/pcm0p/sub0/hw_params

Hence... there is the cleanest sound ever produced by a Creative AudioPCI soundcard..! :)
bye... take care of ur soundcard..!

Bottom Note: All the photos (except the first one) are taken with a cheap Vivitar 3315 point-n-shoot digital camera, through a biconvex lens (f=10cm) attached for macro focusing... :) see more fotos of the same thing and such other things and also some different weird things at http://www.flickr.com/photos/ronybc

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