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Sigma Delta Audio Dac

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21 Threads found on Sigma Delta Audio Dac
I presume you understand that most of the filtering of this delta-sigma dac is done in the digital domain by an oversdampling digital filter. According to datasheet, I would assume that the output filter is not required because the dac already includes an analog output filter to remove sampling residuals. There's no (...)
i have similar problem .. i am using pic32mz micro controller and uda1334 dac. (both supports i2s) i have shifted my data with 0x7fff. but output shows no change.. currently i am feeding a saw tooth wave to device.. i can see digital pins are toggling on CRO. but no output
Hi All I am having a problem with a dac and I need a little help. I am attempting to stream audio from an iPhone via Bluetooth. I have an FPGA connected to the Bluetooth chip via an I2S interface, so as the audio comes in I am passing it along to the delta sigma dac and from there of (...)
To achieve 16 bit resolution with 44 kHz sampling rate using regular PWM, you would need about 3 GHz pwm clock. That's obviously not feasible. Some kind of sigma-delta modulation is necessary.
I think they don't spec the settling time because those are probably sigma-delta dacs, not 'normal' ladder types. You could just contact the manufacturer and ask them...
Hello everyone......... I am working on sigma delta Modulator For audio Range of 0 to 20KHz..... I designed the two stage class A/AB amplifier, internal positive feedback hysteresis comparator and 1 bit dac.... My supply voltages are 0 to 1.8V in 180nm of technology in CADENCE.... But for both amplifier and (...)
I just want to design a audio dac, architecture: sigma delta dac + class D driver I don't know the equivalent model of the headphone if no LC filter is used on pcb,is class D still work? i mean if we can hear the voice normally?:-o
1 bit ( it is called sigma-delta encoding for ADC) OTherwise dac's use R/2R switches to convert D 2 A. The other method is logarithmic like A law using in American Telephony or ? Law for the ROTW but that is based on high dynamic range companding with 7 or 8 bits. Compression on short bursts works poorly since you need a library of sounds (...)
Hi, This is my first post on here, wasn't sure if this was better suited in this forum or the FPGA forum, please move if appropriate. I'm in the progress of developing a S/PDIF input delta sigma audio dac on a Spartan 3E 500k FPGA. I have completed the S/PDIF receiver and works well with a very simple 8-bit first order (...)
Dear Scorpeio Inaddition to MP3 Decoder, You have forgot the audio dac too. You should have an sterio audio dac like AD1852 | Stereo, 24-Bit, 192KHz, Multibit sigma delta dac | audio D/A Converters | audio/Video Products | Analog Devices to (...)
Dear All : No I work in delta-sigma dac design , I need a good opamp (single-end ) for the low pass filter design, I read some paper , It need High DC gain about 90dB , Unit-Gain Bw is large 75MHz, The Noise is about 2~3 uVrms (20~20KHz). Does anyone work in this filed, And good OP Structure can share it , Thanks
delta-sigma converter may be the best solution for audio dac.
The book "Understanding delta sigma Data converter" has one chapter to explain the DS dac, and there are many useful papers on JSSC,
I will try to answer some: 1. sigma delta is suitable for small bandwidth and high resolution applications (typical for audio). It can be used for dc. There are techniques used to overcome the idle tone problem + higher order loops suffer less from this problem. 2. Latency is a function of the resolution needed. You can roughly (very (...)
160 MHz to 1.25 MHz is a factor of 128 (7 bits). Why do you have only 4 bits? audio outputs commonly use delta-sigma dac techniques (not simple PWM) to get high amplitude resolution from a modest clock rate. Here's an introduction:
although i don't think that sigma-delta ADC is easy, but it is a good choice to achieve 16-bit accuracy low speed ADC.
I meet a confuse question. When I design the filter, how to decide phase margin. For example, if the passband is 24k (that main are used in audio), becasue of the no-ideal tansform function, the phase will shift from 0 after the low pass filter. So what is the spec for this design, is it 20 or 40 degree shift. Which number is reasonable for aud
Absolutely sigma-delta dacs.
There is always difference between theoretical & pratical limit in analog IC design. Could someone give me some figure or estimation about the degradation of sigma delta ADC/dac from the ideal (theoretical) SNR calculation using different architecture as shown below? I found the SNR estimation in text book as: [Modulator (MOD) , (...)
What is the new development of the audio codec? I mean such as sigma delta ADC/dac ....,etc... Thanks for answering

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