I am looking for an algorithm that dynamically alters the phase relationships between the spectral components of a music or voice signal in an attempt to reduce the peak-to-average ratio. I recall hearing about this in reference to increasing the coverage range of AM transmitters but I am having trouble digging up the references. Any pointers appreciated! Bob Adams
algorithms for increasing peak-to-average ratio for audio signals
Started by ●July 2, 2008
Reply by ●July 3, 20082008-07-03
On Jul 2, 10:45�pm, Robert Adams <robert.ad...@analog.com> wrote:> I am looking for an algorithm that dynamically alters the phase > relationships between the spectral components of a music or voice > signal in an attempt to reduce the peak-to-average ratio. I recall > hearing about this in reference to increasing the coverage range of AM > transmitters but I am having trouble digging up the references.so a level compressor is not going to do it? i once reviewed (in fact, Bob, this is weird, i had to sorta present it to an IEEE Mohonk because i reviewed it and the author was a no- show) an article about creating test signals (in lieu of swept sines or MLS) that sorta randomized phase and converged on phase combinations of the component sinusoids that looked like a good minimum. now, for audio signals, i think the closer we are to the raw recording of something, the more sharp are the transitions (and higher the peak/ average is), then filtering it with some APF that has a sorta randomized phase response might be expected to reduce the peak-to- average ratio. but if you wanna *in*crease the peak/average, then i dunno of a good way to find the right phase function of the APF. r b-j
Reply by ●July 3, 20082008-07-03
On Jul 2, 11:28�pm, robert bristow-johnson <r...@audioimagination.com> wrote:> On Jul 2, 10:45�pm, Robert Adams <robert.ad...@analog.com> wrote: > > > I am looking for an algorithm that dynamically alters the phase > > relationships between the spectral components of a music or voice > > signal in an attempt to reduce the peak-to-average ratio. I recall > > hearing about this in reference to increasing the coverage range of AM > > transmitters but I am having trouble digging up the references. > > so a level compressor is not going to do it? > > i once reviewed (in fact, Bob, this is weird, i had to sorta present > it to an IEEE Mohonk because i reviewed it and the author was a no- > show) an article about creating test signals (in lieu of swept sines > or MLS) that sorta randomized phase and converged on phase > combinations of the component sinusoids that looked like a good > minimum. > > now, for audio signals, i think the closer we are to the raw recording > of something, the more sharp are the transitions (and higher the peak/ > average is), then filtering it with some APF that has a sorta > randomized phase response might be expected to reduce the peak-to- > average ratio. > > but if you wanna *in*crease the peak/average, then i dunno of a good > way to find the right phase function of the APF. > > r b-jThanks. A compressor won't do it because the signal is already highy compressed. I was looking to eek out a few more db of level without clipping by playing with phase. It seems like there might be 2 approaches; 1) Frequency-domain approach that rotates the phase of each bin to minimize the global peak-to-average ratio exactly how you would do this is unclear to me) 2) Allpass filter approach where each allpass filter is dynamically adjusted in frequency to minimize the local peak-to-average (each allpass might handle a different frequency range; Again, the algorithm for adjustement is unclear but I could see doing a simple gadient search in real time.) Both of the above will have audible artifacts but I'm not too concerned with this since its not an audiophile application. Bob
Reply by ●July 3, 20082008-07-03
On Jul 2, 10:45�pm, Robert Adams <robert.ad...@analog.com> wrote:> I am looking for an algorithm that dynamically alters the phase > relationships between the spectral components of a music or voice > signal in an attempt to reduce the peak-to-average ratio.How much delay can you tolerate? Seems to me that you'll have to look- ahead in the signal at least several periods of the lowest frequency content. Maybe center a window on each temporal peak, FFT that window, look for zero-phase spectral components, and move some of them around. Also; REDUCE crest factor, or INCREASE crest factor? Your message says "reduce" but your title says "increase". Greg
Reply by ●July 3, 20082008-07-03
Robert Adams wrote:> On Jul 2, 11:28 pm, robert bristow-johnson <r...@audioimagination.com> > wrote: >> On Jul 2, 10:45 pm, Robert Adams <robert.ad...@analog.com> wrote: >> >>> I am looking for an algorithm that dynamically alters the phase >>> relationships between the spectral components of a music or voice >>> signal in an attempt to reduce the peak-to-average ratio. I recall >>> hearing about this in reference to increasing the coverage range of AM >>> transmitters but I am having trouble digging up the references. >> so a level compressor is not going to do it? >> >> i once reviewed (in fact, Bob, this is weird, i had to sorta present >> it to an IEEE Mohonk because i reviewed it and the author was a no- >> show) an article about creating test signals (in lieu of swept sines >> or MLS) that sorta randomized phase and converged on phase >> combinations of the component sinusoids that looked like a good >> minimum. >> >> now, for audio signals, i think the closer we are to the raw recording >> of something, the more sharp are the transitions (and higher the peak/ >> average is), then filtering it with some APF that has a sorta >> randomized phase response might be expected to reduce the peak-to- >> average ratio. >> >> but if you wanna *in*crease the peak/average, then i dunno of a good >> way to find the right phase function of the APF. >> >> r b-j > > Thanks. > > A compressor won't do it because the signal is already highy > compressed. I was looking to eek out a few more db of level without > clipping by playing with phase. It seems like there might be 2 > approaches; > > 1) Frequency-domain approach that rotates the phase of each bin to > minimize the global peak-to-average ratio exactly how you would do > this is unclear to me) > > 2) Allpass filter approach where each allpass filter is dynamically > adjusted in frequency to minimize the local peak-to-average (each > allpass might handle a different frequency range; Again, the algorithm > for adjustement is unclear but I could see doing a simple gadient > search in real time.) > > > Both of the above will have audible artifacts but I'm not too > concerned with this since its not an audiophile application.I have no idea how to do it on the fly, but you should be near an optimum when odd harmonics of a strong fundamental cross zero at the same time and in the same direction as the fundamental (think square wave) and the even harmonics are in quadrature. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●July 3, 20082008-07-03
Robert Adams wrote:> I am looking for an algorithm that dynamically alters the phase > relationships between the spectral components of a music or voice > signal in an attempt to reduce the peak-to-average ratio.Why would you need it in our days? It is much more efficient to convert the audio into something like mp3 and transmit it in the digital form.> I recall > hearing about this in reference to increasing the coverage range of AM > transmitters but I am having trouble digging up the references.There are the approaches with the multiband compressors or nonlinear limiters. For better results, the signal can be represented as the Hilbert transform, and then the module of the signal is limited while keeping the phase. Vladimir Vassilevsky DSP and Mixed Signal Design Consultant http://www.abvolt.com
Reply by ●July 3, 20082008-07-03
robert bristow-johnson wrote:> On Jul 2, 10:45 pm, Robert Adams <robert.ad...@analog.com> wrote: > >>I am looking for an algorithm that dynamically alters the phase >>relationships between the spectral components of a music or voice >>signal in an attempt to reduce the peak-to-average ratio. I recall >>hearing about this in reference to increasing the coverage range of AM >>transmitters but I am having trouble digging up the references. > > > so a level compressor is not going to do it?Audio crest factor is essentially a short term parameter. A level compressor can adapt for the slow variation of the volume. Actually, the fast compressor is perceived worse then the memoryless nonlinear limiter. But now, with all digital, I wonder who and why would have a need for that. Vladimir Vassilevsky DSP and Mixed Signal Design Consultant http://www.abvolt.com
Reply by ●July 3, 20082008-07-03
Jerry Avins wrote:> Robert Adams wrote: >> On Jul 2, 11:28 pm, robert bristow-johnson <r...@audioimagination.com> >> wrote: >>> On Jul 2, 10:45 pm, Robert Adams <robert.ad...@analog.com> wrote: >>> >>>> I am looking for an algorithm that dynamically alters the phase >>>> relationships between the spectral components of a music or voice >>>> signal in an attempt to reduce the peak-to-average ratio. I recall >>>> hearing about this in reference to increasing the coverage range of AM >>>> transmitters but I am having trouble digging up the references. >>> so a level compressor is not going to do it? >>> >>> i once reviewed (in fact, Bob, this is weird, i had to sorta present >>> it to an IEEE Mohonk because i reviewed it and the author was a no- >>> show) an article about creating test signals (in lieu of swept sines >>> or MLS) that sorta randomized phase and converged on phase >>> combinations of the component sinusoids that looked like a good >>> minimum. >>> >>> now, for audio signals, i think the closer we are to the raw recording >>> of something, the more sharp are the transitions (and higher the peak/ >>> average is), then filtering it with some APF that has a sorta >>> randomized phase response might be expected to reduce the peak-to- >>> average ratio. >>> >>> but if you wanna *in*crease the peak/average, then i dunno of a good >>> way to find the right phase function of the APF. >>> >>> r b-j >> >> Thanks. >> >> A compressor won't do it because the signal is already highy >> compressed. I was looking to eek out a few more db of level without >> clipping by playing with phase. It seems like there might be 2 >> approaches; >> >> 1) Frequency-domain approach that rotates the phase of each bin to >> minimize the global peak-to-average ratio exactly how you would do >> this is unclear to me) >> >> 2) Allpass filter approach where each allpass filter is dynamically >> adjusted in frequency to minimize the local peak-to-average (each >> allpass might handle a different frequency range; Again, the algorithm >> for adjustement is unclear but I could see doing a simple gadient >> search in real time.) >> >> >> Both of the above will have audible artifacts but I'm not too >> concerned with this since its not an audiophile application. > > I have no idea how to do it on the fly, but you should be near an > optimum when odd harmonics of a strong fundamental cross zero at the > same time and in the same direction as the fundamental (think square > wave) and the even harmonics are in quadrature.Oops! Not quadrature: they should cross in the opposite direction. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●July 3, 20082008-07-03
Robert Adams wrote:> A compressor won't do it because the signal is already highy > compressed. I was looking to eek out a few more db of level without > clipping by playing with phase. It seems like there might be 2 > approaches; > > 1) Frequency-domain approach that rotates the phase of each bin to > minimize the global peak-to-average ratio exactly how you would do > this is unclear to me) > > 2) Allpass filter approach where each allpass filter is dynamically > adjusted in frequency to minimize the local peak-to-average (each > allpass might handle a different frequency range; Again, the algorithm > for adjustement is unclear but I could see doing a simple gadient > search in real time.) > > Both of the above will have audible artifacts but I'm not too > concerned with this since its not an audiophile application. >Robert, What kind of output do you expect from the algorithm if the input is the sum of the two unrelated sine waves? Vladimir Vassilevsky DSP and Mixed Signal Design Consultant http://www.abvolt.com
Reply by ●July 3, 20082008-07-03
On 2008-07-03, Vladimir Vassilevsky <antispam_bogus@hotmail.com> wrote:> Robert Adams wrote: > >> I am looking for an algorithm that dynamically alters the phase >> relationships between the spectral components of a music or voice >> signal in an attempt to reduce the peak-to-average ratio. > > Why would you need it in our days? It is much more efficient to convert > the audio into something like mp3 and transmit it in the digital form.The first step of mp3 (and other audio) compression is to use "psychoacoustics" to figure out which parts of the signal are important to a human listener. There may be models that could be applied and have some advantage back in the analog domain. -- Ben Jackson AD7GD <ben@ben.com> http://www.ben.com/
