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57 Threads found on Audio Sampling
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 specification or further d
Hi, SNR means Signal to Noise Ratio. You need an input signal. It should be high in amplitude, but below clipping level. The frequency should be somewhere in near the center of interrest. (often with audio signals 1kHz is used) Then you need to sample a lot of data. sampling rate should be in integer multiple of your input signal frequency t
fs = 1760; % sampling rate You may hear musical tones at this low a sample rate. However sample rates for audio are normally much greater than this. I heard the figure 8kHz for video streaming. CD quality is 44.1 kHz.
Hi, can anyone briefly explain why we need to do autoregressive model analysis for audio files? give some introduction to autoregressive model analysis?
1. The discrete Fs numbers are industry standard audio sampling rates. The converter can work at different rates, the essential point is the master clock to fs ratio. 2. The internal fs divider is switched according to the actual BCLK rate. 3. A sigma-delta converter is continuously sampling the input signal and generating the output (...)
I know that I can use an oscillator or perhaps a PLL. A PLL appears to be a more complex component and I am not sure how I shall use it right now. Anyway, this question is related to Texas Instrument's PCM1808 audio ADC. It needs a clock input. The table in its data shows the the relationship between sampling rate and the clock input. It says:
Hi, I want to design a controller which capable for capturing the analog signal (from a piezo) and want to process the signal and audio out with assigned sound. Its like music instrument. I tried with pic16f877a to make the comparison, but not able to finish with the audio out part. with help of ADC I did, and also there was problem i
Hi everyone, I'm going to realize an audio player with a PIC32, using PWM as a DAC for audio output. I have not write any code yet, because I want to understand deeply the theory behind it. I stored a .WAV file, 44.1 Khz sample rate, 16 bit, in a SD card, in a previous part of code. Now i want to reproduce this audio trought PWM. Because (...)
I am using a VNC2 (in the 32-bit VDIP board and the full Vinculum tool chain) to develop a USB audio host for a design I'm working on. My 2 test audio devices are a Logitech headset (headphones that can be mono or stereo and 8KHz to 48KHz sampling; mono microphone with the same sampling range) and a USB Soundblaster device (...)
The main achievement of circuit simulators is the ability to predict the working and give transient current/voltage analysis of each device. Generally, for the most applications, such as audio and telephony, sampling on order of mili seconds would suffice. However, if you work with circuits having components working on scale of micro seconds suc
Hai Can anyone help me to calculate the bandwidth required for transmitting a digital audio with 44.1khz sampling rate and 16 bit depth ? regards Aneesh T V
Hi, all when doing the sampling for the audio, do we really care about the sampling rate? I mean as long as we satsify the nyquist conditoin, we will always get back the original audio? The process is done on a PC, I tired several times, it seems this is true. If so, does this mean the reconstruction filter I (...)
Not sure about specifically what kind of resampling topic you are looking at. Resample can refer to aligning signals that need to be syncronized. For example if you have two independent async digitized audio signal that need to be combined into a single data stream you need to resample one or both to get symbol/bit alignment to allow combining
dsPIC30F4013 ADC configuration help looking for a single channel, maximum sampling rate , minimum conversion time 12 bit ADC configuration. Thanks in advance...
I want to demodulate and decode FSK Modulated audio Data in C Language. Data is modulated using similar algorithm as of Caller Id. Can anyone suggest me any good and efficient algorithm for this. I tried zero crossing algorithm but it was not efficient as no. of samples per bit were less to get an accurate result sampling Rate is 8Khz C
In a test cell for diesel engines I am sampling audio from a high-quality industrial microphone and performing an FFT, from which I hope to extract a sound pressure level in dB comparable to a commercial dosimeter to measure how loud the engine is. At this point I am not concerned about absolute references starting with the microphone's sensitivit
Hi ripitup, I suppose you want your 6 inputs to be good quality audio so you'll probably have a sampling rate of something like 44,100Hz per channel. Using one ADC and some multiplexing you can probably get it working, the main factor here is how fast your ADC can go. If you use a microcontroller as the interface between USB and the channels,
Where i can download wav files?. Thank you.
Hey PPL! I'm having a problem with sampling a audio signal using 18F452 Microchip.I've written the mikroc program to sample the audio signal & send it to the computer as soon as PIC reads the sample value(It's been attached with this problem is I tried simple sinusoidal wave of 50Hz to sample but the number of samples are very less As I
I know ADC are used in audio/video/data acquisition/cellphone/HD TV ... but what are the specifications of different applications like sampling rate and resolution? and how are they used?

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