Audio Converters Forum - Recent Threads

SRC4182 / DC level output at mute mode

S.Satoshi — Thu, 25 Apr 2013 15:55:31 GMT

Hi,

Now our customer is evaluating SRC4182 and they have one problem on their test board. They can confirm the expected output from SDOUT but there is DC output from SDOUT at mute mode. SRC4182 is muted at no input from input source because MUTE pin tied to RDY pin in their configuration as attached.

Can you please confirm following question?

1, Is it normal operation the DC level is output from SDOUT at mute condition?
2, Do you have the way to shut off this DC level output?
3, Can they know what voltage is output from SDOUT if it’s default operation? They are asking this because they confirmed various voltage at each checking.

Best Regards,

Sonoki / Japan Disty

Virtex 4 ML403 audio interfacing problem

Muneeb Ziaa — Fri, 24 May 2013 13:35:51 GMT

Hi experts,

I am using Xilnix Virtex 4 ml403 evaluation platform borad, I am using the on board audio processing IC LM4550, I have successfully acquired my audio signal through the on board audio input and output audio jack, but I am facing a problem and that is my audio signal is directly going to the output and I have found from a site that :
"The line-in volume (0x10) and microphone volume (0x0E) registers allow the line-level and microphone inputs to be fed directly into the output mixers, without digitizing these signals".
Link to the site:
http://www-mtl.mit.edu/Courses/6.111/labkit/audio.shtml

Datasheet say that when signal enter from the input jack either from line in or microphone, it first go to the built in18 bit ADC from where the 18 bit digital data goes in to the FPGA pin which follows the AC97 codec serial input protocol. after the digitized data is available at the FPGA pin (refered to SDATA_IN in the datasheet as output of digital from LM4550 ) user can save the data or can send the data back to the LM4550 IC through (SDATA_OUT pin refered in the datasheet for input of digital data) which also follows the AC97 codec serial protocol, this digitized data is send to the built in 18 bit DAC which send an analog signal to either line out jack or headphone jack.

I can control the input and output jack through code but I am still receiving my analog signal even I don't accept the digital data or send it back to the output pin.

My questions are

1. is it true that the line in and microphone is diectly connected to the output?
2. if not then how to stop my input signal to go to the output ports directly, because i am not sending the signal to the output in my code but I am still able to recive it at the output?

please anyone has any information about this please tell me.

Regards,
Muneeb ziaa

Inquiry regarding the amplitude characteristic on TLV320AIC3101

Atsushi Okui — Fri, 24 May 2013 09:10:05 GMT

Hello, all

Now we are evaluating TLV320AIC3101EVM, then would like to ask you regarding amplitude characteristic on TLV320AIC3101.

As I measured this characteristic by following condition, the output amplitude was dropped about 5dB compared with input amplitude.

Input signal level: 20kHz @20dBm

Hardware setting: IN2L(J7)=>A/D=> roop back =>DAC=>Line(LEFT on J10)

Amplitude measurement equipment: Panasonic P-7723A

Script file which is read on GUI: Please see the attachment.

(Please visit the site to view this file)

On these configuration, the output was 5dB dropped than input.

Please let us clarify on which part of AIC3101 this gain reduction was caused.

We thank you in advance for your information.

Best regards,

TLV320AIC3110 codec linux driver

Tommy Hsu — Wed, 14 Dec 2011 08:27:55 GMT

Hi ,

Can you provide me the linux driver of TLV320AIC3110 on linux 2.6.35 ??

PCM3070 and 8MHz clock-in

Raul Atkinson1 — Wed, 08 May 2013 05:05:50 GMT

We are using a HW design that has an 8MHz input clock.

What PLL/clock configuration should we use to enable ADC->DSP->DAC audio flow?

It sounds like we get 1 sec. of audio, and then nothing. Like there is a bubble in the pipeline...

Also, I noticed the output from the PPS writes to page 0, register 254. The manual only describes register 0 through 127.

Is there further documentation on these registers or is it just a nop instruction to delay the programming a bit?

Thanks for your help

-Raul

TLV320AIC3204 - ADC "white noise"

Shlomi Yehezkia — Thu, 16 May 2013 18:45:31 GMT

(Please visit the site to view this file)(Please visit the site to view this file)(Please visit the site to view this audio)

Hi Team,

My customer platform based on TLV320AIC3204 codec.

It is working correctly but there is ADC "white noise" problem they unable to resolve.

Tried AGC but could not find optimized register settings based on system MIC & HW.

Please find enclosed:

PCB schematics – codec part
Register settings
Audio example attached ( used 12db gain , Reg 0x3C, 0x3B)

BCLK = 256K.

They using codec internal PLL (External MCLK not ued) to generate CODEC_CLKIN = 2.048Mhz.

They using the attached MIC

http://www.puiaudio.com/pdf/TOM-1545P-R.pdf

Can you help?

Thanks,

Shlomi

Mic input noise with AIC3204 and AIC3254

kurt57633 — Mon, 04 Oct 2010 14:35:52 GMT

Hello All,

I need some assistance in reducing the mic input noise we’re hearing with the TLV320AIC3254 and TLV320AIC3204. It sounds like white noise. I’m using a single-ended mic configuration with 10K input resistors and a headset. It’s not a super high level of noise, but it is noticeable. Our gain is set to 35 dB.

To isolate the problem, I put the microphone in ADC bypass using mixer amplifiers mode (5.3.2 in datasheet). This routes the microphone level signals to the headphone output fully bypassing the ADC and DAC. I hear the white noise even if I remove the microphone, or ground out the mic input. Therefore, I don’t think it is the microphone generating the noise. I do get the least amount of noise by selecting the lowest mic bias voltage generated by the LDO.

The DAC performance is excellent. When I output a sine wave from the I2S interface it sounds great. When I output zero to the I2S DAC interface, the headphones are absolutely quiet. Therefore, I’ve narrowed the problem down to the analog input (PGA and mic bias).

So far we’ve used the AGC with the noise threshold (74 dB) to gate the noise. This sounds good if you speak into the mic at high enough volumes to overcome the noise threshold. Some quieter talkers have trouble with this, and if we lower the noise threshold anymore to accommodate them, the noise starts to cut in and out again.

We have all the unused inputs capacitively coupled and grounded.

Any ideas would be appreciated. I’ve been through all the register settings and cannot seem to find anything else to do programmatically. I’m not sure the 3254 DSP functions with pure path studio would help eliminate the noise either.

Thank you,
kurt

Noise Reduction - LMV1089

Andrew Pavei — Thu, 23 May 2013 17:40:33 GMT

Dears

We are researches at the Federal University of Santa Catarina, Brazil (www.linse.ufsc.br), that develops/implements noise reduction and echo cancelling algorithms. Following the recomendation shown in the documents SNAA056B and SNAS441H, we develop a noise reduction kit (http://www.ti.com/lit/ug/snaa056b/snaa056b.pdf.) with the goal to compare the result of it with algorithms developed in our laboratory.

We follow all the calibration procedure shown in SNA441H ( pages 23 to 26 ), but unfortunately, the results of the noise reduction kit were lower than shown on page 13, figure “Far Field Noise Suppression Electrical vs frequency” from document.

We request support to evaluate the performance of the kit, for instance, the signals used to tests “LMV1089 Dual Input, Far Field Noise Suppression Microphone” or the LMV1089 I2C Interface V1.0 software.

Sincerly

LINSE - Circuit and Signal Processing Laboratory

UFSC - Federal University of Santa Catarina

Florianópolis, Santa Catarina, Brazil

PCM1792A

Glenn Fasnacht — Fri, 03 May 2013 15:46:05 GMT

I am using a PCM1792 in a non-audio application, a product which will generate signals up to 80kHz. I am seeing higher noise than I expect. When 0V is commanded from the DAC it generates near zero volts but there is noise on the DAC's output buffer (before the anti-imaging filter). The noise is roughly 80mV p-p in the area of 800kHz to 1.8MHz.

The question is, is this to be expected?

Is it a result of the digital filter?

Is this the source of limit cycles or idle tones?

Or am I doing something wrong?

set sampling rate AIC3204

Albert Torn�� — Thu, 23 May 2013 12:24:06 GMT

Hello,

I would want to know how can I configure the AIC3204 to set the sampling rate to selected sampling rates as 24Khz or 32 KHz.

I am working with BCLK and WCLK as inputs.

From a configuration working at 48Khz, I tried to change values of divisors to get this sampling frequencies and I only get loud noise.

Thanks!

TLV320AIC3111 "standalone" after init?

Thomas Gruber — Thu, 23 May 2013 07:43:45 GMT

Dear TI Forum,

I am working on an architecture study for a customer of ours and I want to implement the TLV320AIC3111.

My concern is the host system that this IC would be connected to. Latency in communication to it cannot be guaranteed so we would need to set up the TLV320AIC3111 to work as much in standalone mode, as possible.

We would use the I2S Input of the TLV320AIC3111 and would MIX the MIC1LP, MIC1RP signal to the output signal. But we would want to MIX it depending on the status of the VOL/MICDET Pin. So if there is a signal on e.g. the MIC1LP (which is connected to the VOL/MICDET Pin then) the I2S Input signal should be Soft-Muted automatically. If the signal on the VOL/MICDET Pin is gone again, the signal coming from the I2S Input shall fade in again - all automatically. Of course initialise the IC in the beginning, but then this audio fade out / fade in would work automatically.

Could you please let me know if this is possible with this platform.

Thanks very much!

tom

PCM1795 / I/V and filter for 192kHz sampling

S.Satoshi — Tue, 21 May 2013 15:13:48 GMT

Hi,

Now our customer is designing their board by using PCM1795. And they are designing I/V and filter for 192kHz sampling with Vout=2Vrms for both PCM and DSD mode.

Can you prepare reference circuit of the three pair opamp for I/V and filter and external passive components ?

Best Regards,

Sonoki

PCM5102EVM-U : No function

Tim Pantel — Thu, 14 Feb 2013 13:56:27 GMT

Dear community,

i am experimenting with PCM5102EVM-U.

The probleme is that the board has no function. No signal comes through the board. It does not start.

If i connect the board with the pc and i start the CodecControll Software i will get the following message: "'149.99998' is not a valid floating point value". Do you have a solution for this problem?

Do you have a step by step tutorial to start the board?

PCM9211 (S-PQFP-G48 package) IC Leg bend up spec

Tze Khong Yap — Tue, 21 May 2013 02:12:19 GMT

Hi,

My customer is using PCM9211PTR (S-PQFP-G48 package) for their project.

And they would like to request the IC leg bend up spec for this device?

Understand from PCM9211PTR datasheet, the Gage Plane is 0^o - 7^o.

This means the IC leg can allow to bend up to 7^o?

Can we know how many millimeter can the IC leg to be bend up?

Thanks.

Yap.

set sampling frequency with AIC3204

Albert Torn�� — Fri, 17 May 2013 11:41:09 GMT

Hello,

I would want to know how can i set the sampling frequency at 24KHz or 32 KHz. I have an example from a code working at 48KHz:

/* Configure AIC3204 */

AIC3204_rset( 0, 0x00 );

// Select page 0

AIC3204_rset( 1, 0x01 );

// Reset codec

AIC3204_rset( 0, 0x01 );

// Point to page 1

AIC3204_rset( 1, 0x08 );

// Disable crude AVDD generation from DVDD

AIC3204_rset( 2, 0x00 );

// Enable Analog Blocks

// PLL and Clocks config and Power Up

AIC3204_rset( 0, 0x00 );

// Select page 0

AIC3204_rset( 27, 0x00 );

// BCLK and WCLK is set as i/p to AIC3204(Slave)

AIC3204_rset( 4, 0x07 );

// PLL setting: PLLCLK <- BCLK and CODEC_CLKIN <-PLL CLK

//AIC3204_rset( 6, 0x08 ); // PLL setting: J=8

AIC3204_rset( 6, 0x20 );

// PLL setting: J=32

AIC3204_rset( 7, 0 );

// PLL setting: HI_BYTE(D = 0)

AIC3204_rset( 8, 0 );

// PLL setting: LO_BYTE(D) = 0

// For 48 KHz sampling

AIC3204_rset( 5, 0x92 );

// PLL setting: Power up PLL, P=1 and R=2

AIC3204_rset( 13, 0x00 );

// Hi_Byte(DOSR) for DOSR = 128 decimal or 0x0080 DAC oversamppling

AIC3204_rset( 14, 0x80 );

// Lo_Byte(DOSR) for DOSR = 128 decimal or 0x0080

AIC3204_rset( 20, 0x80 );

// AOSR for AOSR = 128 decimal or 0x0080 for decimation filters 1 to 6

AIC3204_rset( 11, 0x88 );

// Power up NDAC and set NDAC value to 8

AIC3204_rset( 12, 0x82 );

// Power up MDAC and set MDAC value to 2

AIC3204_rset( 18, 0x88 );

// Power up NADC and set NADC value to 8

AIC3204_rset( 19, 0x82 );

// Power up MADC and set MADC value to 2

I have read documentation about AIC3204, and I thought thay by changing values of MADC and NADC, I could change the sampling frequency. However, It is not the case.

I would want to know which register should I change, and with which values in order to set the sampling frequency at 24 KHz for example.

Thank you!!

PCM2900 Linux ALSA driver information?

takishin — Tue, 21 May 2013 01:42:55 GMT

Our customer has a question.
I'm posting this question for a customer:

++++++++++++++++++++++++++++++++++++++++++++++
If you know, please let me know.

Does Linux have an ALSA driver for PCM2900 ?
If Linux has it, which distribution is with it ?

++++++++++++++++++++++++++++++++++++++++++++++

thanks in advance

PurePath Studio (Home Audio) v5.95 Build 1 Rev. 20094 Problem

Andrew Gillham — Wed, 24 Oct 2012 15:47:18 GMT

Hi I have just been granted access to the PurePath Studio platform. I am hoping to use it with a PCM5141. Unfortunately when I install and run the program I get an error citing some missing xml files. It appears that the program is failing to start up due to some missing files in the component cache. I have tried installing the "portable" version of PurePath Studio. This starts up correctly but the framework for the PCM5141 chip is not available in this version.

Has the problem with this build been noticed before? When can a new build that starts up correctly be expected?

Many thanks for looking into this for me.

Andrew.

How flat is frequency response of PCM5102 ?

Debasish Deka — Fri, 17 May 2013 13:18:18 GMT

Dear All,

On behalf of our organization I am posting this information request. We are in requirement of a DAC having a flat frequency response of upto +/-0.01dB in the range 30Hz to 120kHz. I have gone through the documentation but could not find any information regarding this. Could any one shed some light on it ?

Regards,

Debasish Deka

audio chip

shani wolf — Sun, 19 May 2013 11:20:11 GMT

Hello,

I'm looking for a HW config audio chip with I2S as an input and headphone as output.

Is there somthing u can offer me?

I have found an audio chip with I2S input to line out so a line out to headphone convertor would also be good.

Thanks in advance

shani

Linux device drivers for AIC31xx/DAC31xx/AIC325x/AIC320x/AIC326x/AIC321x

Hari Rajagopala — Tue, 21 May 2013 05:12:20 GMT

Reference ALSA SoC codec drivers for the above mentioned devices are now available from gitorious.org. Please see this wiki article for more details. These device drivers have undergone limited testing on select OMAP3- and OMAP4-based platforms. While every attempt has been made to adhere to guidelines for the development of ALSA SoC codec drivers, use of these drivers on other application processor platforms may require significant additional plumbing.

Device variants that include the miniDSP also require a binary firmware to be made available to the driver by the system. The firmware may be built from PurePath™ Studio GDE process flows using firmware tools attached to this post. Additional information is available in this wiki article and in documents contained in the attached package.

These drivers are recommend for all use with Linux kernels 2.6.37 and newer. Select drivers for use on older kernels (as listed here) are available on request. Please send a mail to codecdrivers-sw@list.ti.com mentioning the kernel version and application processor you plan to use to enable us to send you the closest available match.

AIC3254: Not reflect in the Flag register of "HPL powered up" even though the HPL PGA register was powered-down

Kato Motoki — Fri, 19 Apr 2013 08:15:10 GMT

Hi Community member,

Please let me confirm my concern as below.

[My concern]

I have summarized my concern to the subject as this post . AIC3254 did not reflect in the Flag register of "HPL powered up" even though the HPL PGA register was powered-down.

So, would you please double check whether this problem will occur in your side?

I summarized the reproduced method with using AIC3254EVM-U as below.

[ Re-produce Procedure]

1. Connect the AIC3254EVM-U to PC

2. Run the Typical Configuration of Playback.

* There is not relationship of Typical Playback Configuration mode for this issue. Please choose one configuration by yourse3. lf.

3. Open the Status Flags and select the "DAC Flags". And push the "poll" button on this window.

4. Open the "Analog Output" window, and select the "Headphone Outputs" tab.

5. Power down the HPL PGA with checking the Status Flags.

For your information, when powered down the HPR PGA, this status Flags worked well.

If you have any question, please let me know.

Best regards,

Kaka

TAS3204 radiated noise at 135mhz (clock)

David Alexandrou — Tue, 05 Mar 2013 02:04:38 GMT

Hey there,

We have a design that uses two TAS3204 chips, and it's having trouble passing C.E. in the emitted RFI test, specifically with an RCA attached, at 135mhz we're between 5 and 10 dB over the acceptable limit.

The board is designed well, we have ferrites isolating all power supply lines, also between analog and digital grounds. There aren't any ground loops.

Shielded case has little effect, the closest we can come is by putting a ferrite choke over the power supply cable, and also the RCA cables. It gets the noise down to 4db over the limit. We tried ferrite pads on top of the chip and under the board where the chip is, it had almost no effect.

The unit has potentially 10 rca cables connected at a time, so putting ferrite chokes on all of these cables is not feasible in a consumer level product.

Any thoughts / tips / things we should look for? The circuit design itself was based around the EVM schematics, though I don't know if the EVM would pass CE either.

Any tips or recommendations, or "gotcha" mistakes to look out for would be most appreciated.

~David

PCM9211 Configuration issue

Chandra Bhanu Vats — Tue, 30 Apr 2013 09:49:00 GMT

Hello Sir,

I want to make a ADC with the help of PCM9211 .Here the input would be an Analog Stereo pair and the output would be S/PDIF signal which we will collect from MPO 0/1. I have configured the PCM 9211 registers as

0x42 ---------------- 0x02

0x31 ---------------- 0x16

0x48 ---------------- 0x00

0x49 ---------------- 0x00

0x60 ---------------- 0x22

0x78 ---------------- 0xFE

0x61 ---------------- 0x10

0x35 ---------------- 0x0C

0x6b ---------------- 0x22

The strategy is to activate the ADC in I2S Mode and we have selected 48KHz as our sampling rate.The I2S output of the ADC would be given as the Source signals(Data,LR Clock,Bit Clock) to DIT.The output of the DIT would be connected to RECOUT0 and RECOUT1 and then it would be routed to MPO 0/1.
I am using PCM9211 EVM-U as my initial development board and controlling the registers with the help of Codec Control Software.
The problem is that,I am not able to perform this task at all.Please suggest me the way by which we can perform this task with the help of PCM9211 IC.

Regards,

CB Vats

Application Engineer

Comcon Technologies Limited

AIC3262 EVM, digital mic's and ANC algorithm from PurePath - is it possible?

Dmitriy Issaev — Mon, 20 May 2013 12:59:51 GMT

Hello!

I'm interested in realization of dual mic noise canceler. I learned that some of TI's codecs have DSP onboard and there is some IDE for that DSP, caled PurePath. That IDE have some NC algorithms based on adaptive filters. Moreover 2 PDM ( with sigma-delta output) digital mics could be connected somewhere in the middle of codec's ADS, between its own sigma-delta modulator, or instead of it, and decimating CIC filter. I'd like to purchase AIC3262 EVM to test provided ANC abilities. after registration I'v downloaded latest PPS.

questions:

1. how to configure codec's ADC to deal with digital mic's? I mean where to do that? it's seems like PPS only for miniDSP code developing.

2. with PPS trying build ANC code I have "resources exceed 100%" error. what's wrong? is it tested option for 3262?

Regards,

Dmitriy.

(Please visit the site to view this file)

Is there any driver, as for, TLV320AIC1110 carried out Linux correspondence?

Satoshi — Wed, 15 May 2013 01:19:04 GMT

Although you examine TLV320AIC1110 for Headset, isn't there any driver who carried out Linux correspondence?

OS is Linux in order that a customer may control by i.MX25 of Freescale.

And the driver who carried out linux correspondence was not found within web.

Thank you for your consideration.