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Discussion Groups
Discussion Groups | Freescale DSPs | Use PWM D/A convertor
Technical discussions about Freescale (Motorola) DSPs (including the DSP56000, DSP56300, DSP56600, 56800 DSPs).
Use PWM D/A convertor - haoyeshen - Nov 23 22:34:00 2004
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I am trying to develop a small application for speech signal by using dsp56f801. One of the problem i am having is how to use PWM as a D/A convertor. My understanding of how the PWM work is as follow: - PWM in dsp56f801 has 15 bit resolution - PWM uses a counter to decided the pulse width (duty cycle) - 0 means 0% and 32767 preresents 100% - the IPBus clock is 40MHz - Therefore, in order to keep all 15bits of resolution, the maximum PWM frequency is 40MHz/32767 = 1.2kHz If this is true, then I can't use this system for voice applications since the bandwidth for voice is at least 3kHz. Could some one correct me if i am wrong |
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RE: Use PWM D/A convertor - Ebersman, Howard - Nov 24 14:37:00 2004
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You may use
it for voice, however, you will not be able to use all 15 bits of resolution and product
intelligible voice.
Generally
voice systems run at 8Khz (giving 4kHz BW), which implies that you can get on the order of
40Mhz/8kHz =
5000 steps = log2(5000) bits = 12.3 bits.
This gives a
theoretical max SNR of about 70dB, and is better than standard
PCM which is 8 bits (~48dB SNR max).
PCM has
a better MOS than available to most, if not all, cell phone codecs. It is probably a hair
worse than a typcial CD play.
Howard
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Re: Use PWM D/A convertor - haoyeshen - Nov 25 20:53:00 2004
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Thanks Howard! Two more questions: If I could accept 12bits of resolution which is the resolution for ADC anyway, then I have to re-scale all processed data to 12bits and use this scaled data to control the PWM duty cycle. I am wondering if this is crooect. Also, if 12bits resolution is used, the the PWM frequency is around 8kHz as you mentioned, then this PWM frequency is rather close to the frequencies of the speech signal. To avoid the interference between the two, I have to have very sharp low pass filter to minimize the 8kHz content. Please let me know if the statements above are correct. Thanks again for the help Shen --- In , "Ebersman, Howard" <HEbersman@m...> wrote: > You may use it for voice, however, you will not be able to use all 15 bits > of resolution and product intelligible voice. > > Generally voice systems run at 8Khz (giving 4kHz BW), which implies that you > can get on the order of > 40Mhz/8kHz = 5000 steps = log2(5000) bits = 12.3 bits. > > This gives a theoretical max SNR of about 70dB, and is better than standard > PCM which is 8 bits (~48dB SNR max). > PCM has a better MOS than available to most, if not all, cell phone codecs. > It is probably a hair worse than a typcial CD play. > > Howard > > -----Original Message----- > From: haoyeshen [mailto:haoyeshen@y...] > Sent: Tuesday, November 23, 2004 5:35 PM > To: > Subject: [motoroladsp] Use PWM D/A convertor > > I am trying to develop a small application for speech signal by using > dsp56f801. One of the problem i am having is how to use PWM as a D/A > convertor. My understanding of how the PWM work is as follow: > > - PWM in dsp56f801 has 15 bit resolution > - PWM uses a counter to decided the pulse width (duty cycle) > - 0 means 0% and 32767 preresents 100% > - the IPBus clock is 40MHz > - Therefore, in order to keep all 15bits of resolution, the maximum > PWM frequency is 40MHz/32767 = 1.2kHz > > If this is true, then I can't use this system for voice applications > since the bandwidth for voice is at least 3kHz. Could some one > correct me if i am wrong |
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RE: Re: Use PWM D/A convertor - Ebersman, Howard - Nov 29 17:44:00 2004
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Rescaling of
the signal in a DSP is easy stuff. If tight on processing power, use the barrel shifter
and a power of 2 scaling of the original signal. If you have spare MIPS,
a fractional multiply will do it. The amount of scaling is a function of
You are
correct that to avoid a fairly low sample rate combined with the need for lots of filtering to
average the signal increases the filtering requirements. I'd expect that a 3 poll filter (using
1 op amp) would do the job, although more would reduce noise further. Typically, the cutoff
frequency is somewhere around 3kHz (rather than 4 kHz which is the nyquist
freq).
To make the
work for the analog portion cheaper and/or simpler, you can increase the sampling rate of the
PWM. Doubling that rate doubles the frequency of the sinc products making filtering much
easier, but you lose 1 bit of resolution (6 dBs of theoretical SNR). With a 16 kHz PWM
sample rate, you might even be able to get away with a 1 pole (R-C) filter with a 3-4kHz cutoff
freq.
It it were me
implementing this product, I would simulate the PWM in Matlab to determine the required #
of filter poles, cutoff frequency, and to estimate the SNR of the real system. Typically, doing
this would take a day or two of effort, depending on experience and how hard you are
pushing the PWM. It is not impossible that this PWM will not meet your needs, depending on the
actual requirements, some of which you have omitted. If this is the case, options include using
a different DSP, using a CPLD external to the processor, a DAC chip, etc.
Howard
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