Audio - Increasing output sample rate increases frequency ?

Started by ABM 200 April 2, 2015
Suppose, i record audio at say, 10000 samples/sec for some amount of time.
I store these samples in a file(WAV file). When i play back this audio, at
a different rate, say 30000 samples/ sec. The audio will definitely be
fast. But, theoretically more samples/ sec implies higher frequency. So,
does this mean that the play back audio will be at 3 times the original
(recorded) audio frequency or more ?

	 

_____________________________		
Posted through www.DSPRelated.com
On 2.4.15 23:02, ABM 200 wrote:
> Suppose, i record audio at say, 10000 samples/sec for some amount of time. > I store these samples in a file(WAV file). When i play back this audio, at > a different rate, say 30000 samples/ sec. The audio will definitely be > fast. But, theoretically more samples/ sec implies higher frequency. So, > does this mean that the play back audio will be at 3 times the original > (recorded) audio frequency or more ?
The pitch will change in the same ratio as the signal speedup. In your case, the pitch change will be 3 times (an octave and a quint higher). -- -TV
ABM 200 wrote:
> Suppose, i record audio at say, 10000 samples/sec for some amount of time. > I store these samples in a file(WAV file). When i play back this audio, at > a different rate, say 30000 samples/ sec. The audio will definitely be > fast. But, theoretically more samples/ sec implies higher frequency. So, > does this mean that the play back audio will be at 3 times the original > (recorded) audio frequency or more ? > > > > _____________________________ > Posted through www.DSPRelated.com >
The D/A converter would have to use a reconstruction filter with a different upper bandlimit for this to work. -- Les Cargill
On 4/2/15 7:05 PM, Les Cargill wrote:
> ABM 200 wrote: >> Suppose, i record audio at say, 10000 samples/sec for some amount of >> time. >> I store these samples in a file(WAV file). When i play back this >> audio, at >> a different rate, say 30000 samples/ sec. The audio will definitely be >> fast. But, theoretically more samples/ sec implies higher frequency. So, >> does this mean that the play back audio will be at 3 times the original >> (recorded) audio frequency or more ? >> > > The D/A converter would have to use a reconstruction filter with a > different upper bandlimit for this to work. >
to work *well*. sometimes we be cheap SOBs and our reconstruction filter is nothing other than what is inherent to the D/A. maybe there is a simple RC LPF after the D/A, maybe more or maybe less. for a conventional D/A, there is the zero-order hold which *would* naturally adjust to the higher sample rate. for a sigma-delta D/A, i dunno if the internal arithmetic in the sigma-delta modulator gets adjusted or not. i never designed a sigma-delta D/A, but i imagine that before the modulator there is an upsampler, say 64x (so it looks like a 3 MHz sampled signal bandlimited to 24 kHz going into the sigma-delta modulator with +1 and -1 coming out at 3 MHz). but i don't think increasing the sample rate changes any numerical parameters in that upsampler (if it stays at 64x). i dunno. -- r b-j rbj@audioimagination.com "Imagination is more important than knowledge."
robert bristow-johnson wrote:
> On 4/2/15 7:05 PM, Les Cargill wrote: >> ABM 200 wrote: >>> Suppose, i record audio at say, 10000 samples/sec for some amount of >>> time. >>> I store these samples in a file(WAV file). When i play back this >>> audio, at >>> a different rate, say 30000 samples/ sec. The audio will definitely be >>> fast. But, theoretically more samples/ sec implies higher frequency. So, >>> does this mean that the play back audio will be at 3 times the original >>> (recorded) audio frequency or more ? >>> >> >> The D/A converter would have to use a reconstruction filter with a >> different upper bandlimit for this to work. >> > > to work *well*. sometimes we be cheap SOBs and our reconstruction > filter is nothing other than what is inherent to the D/A. maybe there > is a simple RC LPF after the D/A, maybe more or maybe less. > > for a conventional D/A, there is the zero-order hold which *would* > naturally adjust to the higher sample rate. > > for a sigma-delta D/A, i dunno if the internal arithmetic in the > sigma-delta modulator gets adjusted or not. i never designed a > sigma-delta D/A, but i imagine that before the modulator there is an > upsampler, say 64x (so it looks like a 3 MHz sampled signal bandlimited > to 24 kHz going into the sigma-delta modulator with +1 and -1 coming out > at 3 MHz). but i don't think increasing the sample rate changes any > numerical parameters in that upsampler (if it stays at 64x). > > i dunno. > >
This looks like it's darn near just PWM into a lowpass filter: http://www.cirrus.com/en/pubs/proDatasheet/CS4398_F2.pdf Sez it has a "switched capacitor" DAC . Looks complicated: http://www.seas.ucla.edu/brweb/teaching/AIC_Ch12.pdf To abuse a phrase; a "class D DAC". -- Les Cargill
On 4/2/15 9:40 PM, Les Cargill wrote:
> robert bristow-johnson wrote: >> On 4/2/15 7:05 PM, Les Cargill wrote: >>> ABM 200 wrote: >>>> Suppose, i record audio at say, 10000 samples/sec for some amount of >>>> time. >>>> I store these samples in a file(WAV file). When i play back this >>>> audio, at >>>> a different rate, say 30000 samples/ sec. The audio will definitely be >>>> fast. But, theoretically more samples/ sec implies higher frequency. >>>> So, >>>> does this mean that the play back audio will be at 3 times the original >>>> (recorded) audio frequency or more ? >>>> >>> >>> The D/A converter would have to use a reconstruction filter with a >>> different upper bandlimit for this to work. >>> >> >> to work *well*. sometimes we be cheap SOBs and our reconstruction >> filter is nothing other than what is inherent to the D/A. maybe there >> is a simple RC LPF after the D/A, maybe more or maybe less. >> >> for a conventional D/A, there is the zero-order hold which *would* >> naturally adjust to the higher sample rate. >> >> for a sigma-delta D/A, i dunno if the internal arithmetic in the >> sigma-delta modulator gets adjusted or not. i never designed a >> sigma-delta D/A, but i imagine that before the modulator there is an >> upsampler, say 64x (so it looks like a 3 MHz sampled signal bandlimited >> to 24 kHz going into the sigma-delta modulator with +1 and -1 coming out >> at 3 MHz). but i don't think increasing the sample rate changes any >> numerical parameters in that upsampler (if it stays at 64x). >> >> i dunno. >> >> > > This looks like it's darn near just PWM into a lowpass > filter: > http://www.cirrus.com/en/pubs/proDatasheet/CS4398_F2.pdf >
if it were a 1-bit D/A, then what it has in common with PWM is that it's greatly oversampled and the low-frequency signal you want is in the "duty-cycle" of the clocking +1 and -1 states. but in sigma-delta they clock back-and-forth more in a random pattern, so that the spectrum of the error is not a deterministic line-spectra, but more like HPF noise. but that device uses a multi-bit D/A (maybe 4 bits), which means a tradeoff of sorts between the noise power of the sigma-delta modulator (which is reduced) and the requirements of linearity of the multibit D/A. except for issues of capacitors charging and discharging symmetrically, a 1-bit D/A doesn't have linearity issues because there is only 1 step (it's when there are multiple steps in the staircase function of a conventional D/A that there are issues of the steps not being precisely equal-sized and the staircase not going up straight and linear).
> Sez it has a "switched capacitor" DAC . Looks > complicated: > http://www.seas.ucla.edu/brweb/teaching/AIC_Ch12.pdf > > To abuse a phrase; a "class D DAC".
something like that. it's a combination of a sigma-delta modulator and a class D D/A. -- r b-j rbj@audioimagination.com "Imagination is more important than knowledge."
robert bristow-johnson wrote:
> On 4/2/15 9:40 PM, Les Cargill wrote: >> robert bristow-johnson wrote: >>> On 4/2/15 7:05 PM, Les Cargill wrote: >>>> ABM 200 wrote: >>>>> Suppose, i record audio at say, 10000 samples/sec for some amount of >>>>> time. >>>>> I store these samples in a file(WAV file). When i play back this >>>>> audio, at >>>>> a different rate, say 30000 samples/ sec. The audio will definitely be >>>>> fast. But, theoretically more samples/ sec implies higher frequency. >>>>> So, >>>>> does this mean that the play back audio will be at 3 times the >>>>> original >>>>> (recorded) audio frequency or more ? >>>>> >>>> >>>> The D/A converter would have to use a reconstruction filter with a >>>> different upper bandlimit for this to work. >>>> >>> >>> to work *well*. sometimes we be cheap SOBs and our reconstruction >>> filter is nothing other than what is inherent to the D/A. maybe there >>> is a simple RC LPF after the D/A, maybe more or maybe less. >>> >>> for a conventional D/A, there is the zero-order hold which *would* >>> naturally adjust to the higher sample rate. >>> >>> for a sigma-delta D/A, i dunno if the internal arithmetic in the >>> sigma-delta modulator gets adjusted or not. i never designed a >>> sigma-delta D/A, but i imagine that before the modulator there is an >>> upsampler, say 64x (so it looks like a 3 MHz sampled signal bandlimited >>> to 24 kHz going into the sigma-delta modulator with +1 and -1 coming out >>> at 3 MHz). but i don't think increasing the sample rate changes any >>> numerical parameters in that upsampler (if it stays at 64x). >>> >>> i dunno. >>> >>> >> >> This looks like it's darn near just PWM into a lowpass >> filter: >> http://www.cirrus.com/en/pubs/proDatasheet/CS4398_F2.pdf >> > > if it were a 1-bit D/A, then what it has in common with PWM is that it's > greatly oversampled and the low-frequency signal you want is in the > "duty-cycle" of the clocking +1 and -1 states. but in sigma-delta they > clock back-and-forth more in a random pattern, so that the spectrum of > the error is not a deterministic line-spectra, but more like HPF noise. >
Yep - although using PWM for classic motor control, the HPF moise is way down there. For audio, "more like HPF noise" is a very good thing.
> but that device uses a multi-bit D/A (maybe 4 bits), which means a > tradeoff of sorts between the noise power of the sigma-delta modulator > (which is reduced) and the requirements of linearity of the multibit > D/A. except for issues of capacitors charging and discharging > symmetrically, a 1-bit D/A doesn't have linearity issues because there > is only 1 step (it's when there are multiple steps in the staircase > function of a conventional D/A that there are issues of the steps not > being precisely equal-sized and the staircase not going up straight and > linear). >
Right. With one unit, there's no unit to unit variation. "Fold" the many-units to be the one unit, and presto.
>> Sez it has a "switched capacitor" DAC . Looks >> complicated: >> http://www.seas.ucla.edu/brweb/teaching/AIC_Ch12.pdf >> >> To abuse a phrase; a "class D DAC". > > something like that. it's a combination of a sigma-delta modulator and > a class D D/A. > >
So it is actually class D? It seemed like that but I was diving into details too soon. We're seeing a lot of things go more like class D these days. I suppose you can keep all the fiddly parts controlled enough to replicate it. Class D looks like a giant suite of offsetting .. hacks to me. I have a class D guitar amp now - and it sounds somewhere between a MOSFET amp and tubes in ... color, while being as "fast" and detailed as MOSFET. Hard to explain. Size weight and power. Maybe it's just me but that seems to have happened pretty fast*. It's really driven by shipping costs, which is fascinating if you think about it. *slightly ludicrous - switchmode power supplies for PC have been the norm since the '80s. -- Les Cargill
robert bristow-johnson <rbj@audioimagination.com> writes:

> but in sigma-delta they clock back-and-forth more in a random pattern, > so that the spectrum of the error is not a deterministic line-spectra, > but more like HPF noise.
Robert, What is "HPF?" I would normally think "High Pass Filter," but in this context that doesn't make sense. -- Randy Yates Digital Signal Labs http://www.digitalsignallabs.com
robert bristow-johnson <rbj@audioimagination.com> writes:

> On 4/2/15 7:05 PM, Les Cargill wrote: >> ABM 200 wrote: >>> Suppose, i record audio at say, 10000 samples/sec for some amount of >>> time. >>> I store these samples in a file(WAV file). When i play back this >>> audio, at >>> a different rate, say 30000 samples/ sec. The audio will definitely be >>> fast. But, theoretically more samples/ sec implies higher frequency. So, >>> does this mean that the play back audio will be at 3 times the original >>> (recorded) audio frequency or more ? >>> >> >> The D/A converter would have to use a reconstruction filter with a >> different upper bandlimit for this to work. >> > > to work *well*. sometimes we be cheap SOBs and our reconstruction > filter is nothing other than what is inherent to the D/A. maybe there > is a simple RC LPF after the D/A, maybe more or maybe less. > > for a conventional D/A, there is the zero-order hold which *would* > naturally adjust to the higher sample rate. > > for a sigma-delta D/A,
You mean delta sigma?
> i dunno if the internal arithmetic in the sigma-delta modulator gets > adjusted or not.
No, it doesn't.
> i never designed a sigma-delta D/A, but i imagine that before the > modulator there is an upsampler, say 64x (so it looks like a 3 MHz > sampled signal bandlimited to 24 kHz going into the sigma-delta > modulator with +1 and -1 coming out at 3 MHz). but i don't think > increasing the sample rate changes any numerical parameters in that > upsampler (if it stays at 64x).
Nope. At least it wouldn't have in my design I did way back when. Of course you have to be careful to stay within operating parameters of the device's hardware. -- Randy Yates Digital Signal Labs http://www.digitalsignallabs.com
"ABM 200" <104798@dsprelated> writes:

> Suppose, i record audio at say, 10000 samples/sec for some amount of time. > I store these samples in a file(WAV file). When i play back this audio, at > a different rate, say 30000 samples/ sec. The audio will definitely be > fast. But, theoretically more samples/ sec implies higher frequency. So, > does this mean that the play back audio will be at 3 times the original > (recorded) audio frequency or more ?
Yes, at exactly 3 times the original. Except that the reconstruction filter won't change, so you may get some filtering of those higher frequencies, as others have pointed out. -- Randy Yates Digital Signal Labs http://www.digitalsignallabs.com