>jadhav_rahul wrote: >>> jadhav_rahul wrote: >>> >>> ... >>> >>>> Thanks jerry, >>>> but which other design decisions you are talking about? >>>> Rahul. >>> You can dynamically adjust the scaling to allow any sum. (Dynamic >>> compression.) You can allow enough headroom for two people to shout at>>> each other, but no more. Clipping overloads would be better than >>> wrapping around, but either works. There is not much reason to allowfor>> >>> everyone talking at once because it wouldn't be intelligible anyway. >>> >>> >>> Jerry >>> -- >>> Engineering is the art of making what you want from things you canget.>>>???????????????????????????????????????????????????????????????????????>> >>> Thanks jerry >> what should be the level of buffering (of decompressed voice data)before>> adding them? >> i mean in case of y[n]=a.x1[n]+b.x2[n]+c.x3[n] >> we have to buffer all the inputs or we can add first two and bufferthe>> result to add with third one? >> which one will be better? >> is there any standard algorithm availabe for entire processing? >> Thanks in advance, > >Getting the result with minimum delay is the only important point. >Assuming that the data are already in linear twos-complement form (what >I think you mean by "PCM"), I would add first and scale the result back >to the allowed bit count. The computer can add only two numbers in one >operation. That may be obscured by a high-level language, but the >processor isn't fooled by fancy notation. :-) > >Jerry >-- >Engineering is the art of making what you want from things you can get. >??????????????????????????????????????????????????????????????????????? >thanks jerry
Audio Mixing using DSP
Started by ●July 7, 2008
Reply by ●July 14, 20082008-07-14
Reply by ●July 16, 20082008-07-16
rahul.jadhav@spectross.com wrote: > Rahul Jadhav visited DSPRelated.com and clicked on your name from this page: > http://www.dsprelated.com/showmessage/99660.php > to contact you. His message follows: > > Hi Jerry, > Thanks for your reply on my Topic of Audio mixing. > I have one doubt, suppose we have conference between 3 > users (PCM companded voice channels) then we add the streams together > with scaling but data which a user can receive will include his own > voice information also or i think we should substract his info. from > the combined data, also as the total sum of scaling factors should be > 1 how we decide these scaling factors becoz these factors decides audio gain of each channel? > Can you plz suggest me steps to follow to implenent voice conference using DSP(I am using Fixed point DSP TMS320c55x) and Components to use from DSP and level of buffering for incoming data. > Thanks in advance. > Rahul jadhav. I don't know how DSPRelated works, but you have reached me through my home email. Please put your questions so that they appear in the newsgroup comp.dsp where there can be a general discussion rather than a private conversation. There, others can correct any errors I might make and contribute ideas I might overlook. Ordinary telephones (and aircraft and military tank intercoms) feed the speaker's voice back to his ears, usually at reduced volume. That is called "sidetone". The system sounds "dead" without it. Do a web search for details. If you are in a position to keep the voices separate until they are mixed for each user, you can generate reduced-loudness sidetone easily. If not, you have two choices: reduce the sidetone locally with a cancellation arrangement that can be difficult to make stable, or let it ride through at full loudness. Techniques for locally reducing (suppressing, in that case) sidetone are used in speaker phones. Remember that the individual must be converted to linear before being added and re-encoded for distribution. Jerry -- "The rights of the best of men are secured only as the rights of the vilest and most abhorrent are protected." - Chief Justice Charles Evans Hughes, 1927 ���������������������������������������������������������������������
Reply by ●July 16, 20082008-07-16
On Jul 16, 10:05�am, Jerry Avins <j...@ieee.org> wrote:> I don't know how DSPRelated works, but you have reached me through my > home email. Please put your questions so that they appear in the > newsgroup comp.dsp where there can be a general discussion rather than a > private conversation.I received the same message through DSPRelated, Jerry, so I think that he just shotgunned everyone who responded on comp.dsp. Is school back in session? We seem to be getting a lot of homework questions lately. Greg
Reply by ●July 25, 20082008-07-25
>jadhav_rahul wrote: >>> jadhav_rahul wrote: >>> >>> ... >>> >>>> Thanks jerry, >>>> but which other design decisions you are talking about? >>>> Rahul. >>> You can dynamically adjust the scaling to allow any sum. (Dynamic >>> compression.) You can allow enough headroom for two people to shout at>>> each other, but no more. Clipping overloads would be better than >>> wrapping around, but either works. There is not much reason to allowfor>> >>> everyone talking at once because it wouldn't be intelligible anyway. >>> >>> >>> Jerry >>> -- >>> Engineering is the art of making what you want from things you canget.>>>???????????????????????????????????????????????????????????????????????>> >>> Thanks jerry >> what should be the level of buffering (of decompressed voice data)before>> adding them? >> i mean in case of y[n]=a.x1[n]+b.x2[n]+c.x3[n] >> we have to buffer all the inputs or we can add first two and bufferthe>> result to add with third one? >> which one will be better? >> is there any standard algorithm availabe for entire processing? >> Thanks in advance, > >Getting the result with minimum delay is the only important point. >Assuming that the data are already in linear twos-complement form (what >I think you mean by "PCM"), I would add first and scale the result back >to the allowed bit count. The computer can add only two numbers in one >operation. That may be obscured by a high-level language, but the >processor isn't fooled by fancy notation. :-) > >Jerry >-- >Engineering is the art of making what you want from things you can get. >??????????????????????????????????????????????????????????????????????? >Thanks jerry, Does any one have any link or document for voice conference using fixed point DSP? Rahul.
Reply by ●July 26, 20082008-07-26
jadhav_rahul wrote: ...> Thanks jerry,You're welcome.> Does any one have any link or document for voice conference using fixed > point DSP? > Rahul.Sadly, not me. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
