Was the output of the C64 SID chip 8 bit sound?

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Was the output of the C64 SID chip 8 bit sound?














2















Was the output of the C64 SID chip 8 bit sound?



I know that SID is not sample based, but instead generate its sound using sine waves and white noise.



So is it correct to say that the audio generation part was really analog, and thus even talking about sound bits really make no sense?



Or maybe what I am really asking is: If I have an 8 bit, 44 kHz sample based audio chip, could that chip generate audio which could not be distinguished from a real SID chip?










share|improve this question









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    2















    Was the output of the C64 SID chip 8 bit sound?



    I know that SID is not sample based, but instead generate its sound using sine waves and white noise.



    So is it correct to say that the audio generation part was really analog, and thus even talking about sound bits really make no sense?



    Or maybe what I am really asking is: If I have an 8 bit, 44 kHz sample based audio chip, could that chip generate audio which could not be distinguished from a real SID chip?










    share|improve this question









    New contributor



    MTilsted is a new contributor to this site. Take care in asking for clarification, commenting, and answering.
    Check out our Code of Conduct.





















      2












      2








      2








      Was the output of the C64 SID chip 8 bit sound?



      I know that SID is not sample based, but instead generate its sound using sine waves and white noise.



      So is it correct to say that the audio generation part was really analog, and thus even talking about sound bits really make no sense?



      Or maybe what I am really asking is: If I have an 8 bit, 44 kHz sample based audio chip, could that chip generate audio which could not be distinguished from a real SID chip?










      share|improve this question









      New contributor



      MTilsted is a new contributor to this site. Take care in asking for clarification, commenting, and answering.
      Check out our Code of Conduct.











      Was the output of the C64 SID chip 8 bit sound?



      I know that SID is not sample based, but instead generate its sound using sine waves and white noise.



      So is it correct to say that the audio generation part was really analog, and thus even talking about sound bits really make no sense?



      Or maybe what I am really asking is: If I have an 8 bit, 44 kHz sample based audio chip, could that chip generate audio which could not be distinguished from a real SID chip?







      commodore-64






      share|improve this question









      New contributor



      MTilsted is a new contributor to this site. Take care in asking for clarification, commenting, and answering.
      Check out our Code of Conduct.










      share|improve this question









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      MTilsted is a new contributor to this site. Take care in asking for clarification, commenting, and answering.
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      share|improve this question




      share|improve this question








      edited 6 hours ago









      Raffzahn

      59.5k6147246




      59.5k6147246






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      asked 8 hours ago









      MTilstedMTilsted

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          3 Answers
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          6














          The output is analog, not 8-bit. Each channel has 8-bit DAC inside to generate the seletable waveforms (sine was not one of them, square/pulse wave, triangle and saw). There were also volume and envelopes, so even at this point the audio can't faithfully be represented at 8 bits. There is also an analog filter through which channels can be sent. The digital DDS oscillators in the chip work at approximately 1 MHz so the DAC output could update at the precision of 1MHz clock. So interesting downsampling or band-limited step interpolation is needed to reproduce or render the output to resemble original audio. 8 bits at 44 kHz would be pushing it.






          share|improve this answer


















          • 2





            If I remember correctly, the person who designed the SID chip had never worked on a sound chip before and presumed that it must work the same way as the analog synthesizers at the time (moog, arp, etc.). Thus, we ended up with the SID chip which was fairly unique in that it had analog sound generation circuitry unlike most others on the market at the time.

            – bjb
            7 hours ago






          • 2





            It does not work like analog synthesizers. The digitally generated waveforms from the oscillator are fed to DAC. This chip was basically meant to be a wavetable oscillator but they were in a hurry so they had to skip the waveform tables and just basically use the phase accumulator oscillator output for DAC which explains the waveforms.

            – Justme
            5 hours ago











          • @Justme: Many analog synths generate waveforms the same way as the SID. What makes them analog are the filters that follow the initial waveform generator. While something like a Juno 106 would have six identical filter circuits to allow six-voice polyphony, the SID only has one, but it's a true analog filter.

            – supercat
            1 hour ago


















          0














          Part 1:




          Was the output of the C64 SID chip 8 bit sound?




          Short Answer: yes



          The term 8-bit sound is generally not related to any sample size or speed, but to describe the sound 'qualities' of the 8-bit generation of consoles/computers.




          Part 2:




          So is it correct to say that the audio generation part was really analog,




          Each and every sound generation is in the end analogue. A (single) speaker can only produce one, on-dimensional output over time.



          The difference between using an DAC with a fixed feeding rate and mixing specialized circuits (like with the SID) is about the amount of data needed to feed either.



          Feeding an 8-bit DAC will need one byte per step, so 44 kByte at 44 kHz. A SID based sound system may need only a single byte for the same duration. That is if only a single symmetric frequency is to be outputted. To produce a more complex output, more data is needed. Notable game sounds can already be generated with less than 100 bytes per second.



          Sounds like a compression algorithm, doesn't it? And that's the whole idea here. It's more like playing an instrument(*1) than outputting plain analogue levels. Here as well the output is generated using predefined elements, in effect saving much of the rather slim bandwith 8 bit machines had (*2).



          Viewed from system design having a sound chip where only sound elements have to be set and manipulated is much like having video controllers with programmable functions (CTIA) and/or sprites (VIC). Instead of having the CPU directly manipulating the bitmap data to create an output image, the image is composed from components which in turn need way less bandwith to be manipulates.



          That design view also gives why it got out of fashion - with CPUs fast enough to manipulate bitmap data just in time for output and fast enough to prepare sound data fast enough for straight DAC output, the need for specialized chips vanished.




          Part 3:




          Or maybe what I am really asking is:




          Which is a different question.




          If I have an 8 bit, 44 kHz sample based audio chip, could that chip generate audio which could not be distinguished from a real SID chip?




          Basically yes.



          Quality will be the very same as a SID output recorded at 44 kHz - much like a CD with SID-music.



          The issue here is, as shown, not some 44 kHz DAC, but a CPU able (aka fast enough) to compute (emulate) the various elements of a SID in time - if not going for calculating it ahead of time (aka batch) that is.



          With modern chips/software it's for most parts straight foreward to build a simple SID-alike system. In fact, the web browser your reading this in is already all you need - at least if it supports WebAudio (*3).




          *1 - Which is BTW the idea MIDI is based on. Here sound data is encoded as start stop condition for instruments (simplified). As result a rather narrow 31.25 kBit/s (~8 KiByte/s) can be used to produce a whole orchestras sound.



          *2 - For example the available bandwith for data transport of a 6502 running at 1 MHz is at maximum less 100 KiByte/s. That's for a copy loop, with some processing is gets way lower. Similar for other CPUs of the same time.



          *3 - WebAudio builds its sound generation pipeline from blocks that can be configured to work similar to the SID pipeline. All needed are a few lines JS to configure and link up sound sources, mixers and filters. Of course, this will be only a first iteration. Really remodelling all non linearities/quirks of teh SID will take a lot more.






          share|improve this answer
































            0















            Or maybe what I am really asking is: If I have an 8 bit, 44 kHz
            sample based audio chip, could that chip generate audio which could
            not be distinguished from a real SID chip?




            The short answer: probably. But, given a choice, use standard 44.1 KHz
            16-bit sampling to reproduce the signal.



            PCM audio reproduction has two components: frequency range (what's the
            highest frequency it can reproduce?) and dynamic range (how loud is
            the background noise level in comparison to the highest signal you can
            hear?).



            As described here (in probably far more detail than you care
            about), the frequency range of properly done 44.1 KHz digital sample
            reproduction will be about 0-20 KHz, which covers the entire set of
            frequencies that most people can hear, so there are no issues there.



            The dynamic range of 8-bit PCM reproduction will be significantly more
            than 48 dB (assuming you use proper dithering); that is, the loudest
            signal will be 48 dB higher than the noise floor. This is better than,
            say, a high-quality cassette tape recording.



            As others here have pointed out, SID chip is an analogue sound
            generator with a digital oscillator producing square, triangle and
            sawtooth waves. Because this is not a PCM sampling system, the
            number of bits of resolution of the digital oscillator is nearly
            irrelevant to the noise floor; that will be almost entirely determined
            by the analogue portions of the system.



            I can't find any figures on the noise floor of the C64 audio output,
            but given that it's a very cost-constrained system in so many ways not
            specifically designed for high-quality musical reproduction (think
            about the typical speaker system used with it!), it's very likely that
            the noise level is well above -48 dB.



            However, another thing to consider is the character of the noise; a
            standard PCM reproduction system that produces white noise filtered to
            a certain spectrum may have "background noise" that sounds quite
            different from that produced by a real C64 reproduction system. Your
            digital emulation of a SID chip and any subsequent analogue circuitry
            may or may not emulate this, but if it does that "SID noise" is part
            of the signal and you want it will above the background noise of the
            reproduction system.



            In the end, unless you're particularly resource constrained (to a much
            greater degree than on a modern PC or phone), you should use 44.1 KHz
            16-bit reproduction so that you simply don't have to think about the
            issues in that part of the system, and instead spend your energy
            focusing on the details of the simulated analogue waveform produced by
            your SID emulation. If the output of your emulation is not as noisy as
            the real system and you want that noise, you should add the noise in
            your emulation, ideally in a way that gives it the same character as
            the noise created by the real system.






            share|improve this answer























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              3 Answers
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              3 Answers
              3






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              active

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              active

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              6














              The output is analog, not 8-bit. Each channel has 8-bit DAC inside to generate the seletable waveforms (sine was not one of them, square/pulse wave, triangle and saw). There were also volume and envelopes, so even at this point the audio can't faithfully be represented at 8 bits. There is also an analog filter through which channels can be sent. The digital DDS oscillators in the chip work at approximately 1 MHz so the DAC output could update at the precision of 1MHz clock. So interesting downsampling or band-limited step interpolation is needed to reproduce or render the output to resemble original audio. 8 bits at 44 kHz would be pushing it.






              share|improve this answer


















              • 2





                If I remember correctly, the person who designed the SID chip had never worked on a sound chip before and presumed that it must work the same way as the analog synthesizers at the time (moog, arp, etc.). Thus, we ended up with the SID chip which was fairly unique in that it had analog sound generation circuitry unlike most others on the market at the time.

                – bjb
                7 hours ago






              • 2





                It does not work like analog synthesizers. The digitally generated waveforms from the oscillator are fed to DAC. This chip was basically meant to be a wavetable oscillator but they were in a hurry so they had to skip the waveform tables and just basically use the phase accumulator oscillator output for DAC which explains the waveforms.

                – Justme
                5 hours ago











              • @Justme: Many analog synths generate waveforms the same way as the SID. What makes them analog are the filters that follow the initial waveform generator. While something like a Juno 106 would have six identical filter circuits to allow six-voice polyphony, the SID only has one, but it's a true analog filter.

                – supercat
                1 hour ago















              6














              The output is analog, not 8-bit. Each channel has 8-bit DAC inside to generate the seletable waveforms (sine was not one of them, square/pulse wave, triangle and saw). There were also volume and envelopes, so even at this point the audio can't faithfully be represented at 8 bits. There is also an analog filter through which channels can be sent. The digital DDS oscillators in the chip work at approximately 1 MHz so the DAC output could update at the precision of 1MHz clock. So interesting downsampling or band-limited step interpolation is needed to reproduce or render the output to resemble original audio. 8 bits at 44 kHz would be pushing it.






              share|improve this answer


















              • 2





                If I remember correctly, the person who designed the SID chip had never worked on a sound chip before and presumed that it must work the same way as the analog synthesizers at the time (moog, arp, etc.). Thus, we ended up with the SID chip which was fairly unique in that it had analog sound generation circuitry unlike most others on the market at the time.

                – bjb
                7 hours ago






              • 2





                It does not work like analog synthesizers. The digitally generated waveforms from the oscillator are fed to DAC. This chip was basically meant to be a wavetable oscillator but they were in a hurry so they had to skip the waveform tables and just basically use the phase accumulator oscillator output for DAC which explains the waveforms.

                – Justme
                5 hours ago











              • @Justme: Many analog synths generate waveforms the same way as the SID. What makes them analog are the filters that follow the initial waveform generator. While something like a Juno 106 would have six identical filter circuits to allow six-voice polyphony, the SID only has one, but it's a true analog filter.

                – supercat
                1 hour ago













              6












              6








              6







              The output is analog, not 8-bit. Each channel has 8-bit DAC inside to generate the seletable waveforms (sine was not one of them, square/pulse wave, triangle and saw). There were also volume and envelopes, so even at this point the audio can't faithfully be represented at 8 bits. There is also an analog filter through which channels can be sent. The digital DDS oscillators in the chip work at approximately 1 MHz so the DAC output could update at the precision of 1MHz clock. So interesting downsampling or band-limited step interpolation is needed to reproduce or render the output to resemble original audio. 8 bits at 44 kHz would be pushing it.






              share|improve this answer













              The output is analog, not 8-bit. Each channel has 8-bit DAC inside to generate the seletable waveforms (sine was not one of them, square/pulse wave, triangle and saw). There were also volume and envelopes, so even at this point the audio can't faithfully be represented at 8 bits. There is also an analog filter through which channels can be sent. The digital DDS oscillators in the chip work at approximately 1 MHz so the DAC output could update at the precision of 1MHz clock. So interesting downsampling or band-limited step interpolation is needed to reproduce or render the output to resemble original audio. 8 bits at 44 kHz would be pushing it.







              share|improve this answer












              share|improve this answer



              share|improve this answer










              answered 8 hours ago









              JustmeJustme

              87439




              87439







              • 2





                If I remember correctly, the person who designed the SID chip had never worked on a sound chip before and presumed that it must work the same way as the analog synthesizers at the time (moog, arp, etc.). Thus, we ended up with the SID chip which was fairly unique in that it had analog sound generation circuitry unlike most others on the market at the time.

                – bjb
                7 hours ago






              • 2





                It does not work like analog synthesizers. The digitally generated waveforms from the oscillator are fed to DAC. This chip was basically meant to be a wavetable oscillator but they were in a hurry so they had to skip the waveform tables and just basically use the phase accumulator oscillator output for DAC which explains the waveforms.

                – Justme
                5 hours ago











              • @Justme: Many analog synths generate waveforms the same way as the SID. What makes them analog are the filters that follow the initial waveform generator. While something like a Juno 106 would have six identical filter circuits to allow six-voice polyphony, the SID only has one, but it's a true analog filter.

                – supercat
                1 hour ago












              • 2





                If I remember correctly, the person who designed the SID chip had never worked on a sound chip before and presumed that it must work the same way as the analog synthesizers at the time (moog, arp, etc.). Thus, we ended up with the SID chip which was fairly unique in that it had analog sound generation circuitry unlike most others on the market at the time.

                – bjb
                7 hours ago






              • 2





                It does not work like analog synthesizers. The digitally generated waveforms from the oscillator are fed to DAC. This chip was basically meant to be a wavetable oscillator but they were in a hurry so they had to skip the waveform tables and just basically use the phase accumulator oscillator output for DAC which explains the waveforms.

                – Justme
                5 hours ago











              • @Justme: Many analog synths generate waveforms the same way as the SID. What makes them analog are the filters that follow the initial waveform generator. While something like a Juno 106 would have six identical filter circuits to allow six-voice polyphony, the SID only has one, but it's a true analog filter.

                – supercat
                1 hour ago







              2




              2





              If I remember correctly, the person who designed the SID chip had never worked on a sound chip before and presumed that it must work the same way as the analog synthesizers at the time (moog, arp, etc.). Thus, we ended up with the SID chip which was fairly unique in that it had analog sound generation circuitry unlike most others on the market at the time.

              – bjb
              7 hours ago





              If I remember correctly, the person who designed the SID chip had never worked on a sound chip before and presumed that it must work the same way as the analog synthesizers at the time (moog, arp, etc.). Thus, we ended up with the SID chip which was fairly unique in that it had analog sound generation circuitry unlike most others on the market at the time.

              – bjb
              7 hours ago




              2




              2





              It does not work like analog synthesizers. The digitally generated waveforms from the oscillator are fed to DAC. This chip was basically meant to be a wavetable oscillator but they were in a hurry so they had to skip the waveform tables and just basically use the phase accumulator oscillator output for DAC which explains the waveforms.

              – Justme
              5 hours ago





              It does not work like analog synthesizers. The digitally generated waveforms from the oscillator are fed to DAC. This chip was basically meant to be a wavetable oscillator but they were in a hurry so they had to skip the waveform tables and just basically use the phase accumulator oscillator output for DAC which explains the waveforms.

              – Justme
              5 hours ago













              @Justme: Many analog synths generate waveforms the same way as the SID. What makes them analog are the filters that follow the initial waveform generator. While something like a Juno 106 would have six identical filter circuits to allow six-voice polyphony, the SID only has one, but it's a true analog filter.

              – supercat
              1 hour ago





              @Justme: Many analog synths generate waveforms the same way as the SID. What makes them analog are the filters that follow the initial waveform generator. While something like a Juno 106 would have six identical filter circuits to allow six-voice polyphony, the SID only has one, but it's a true analog filter.

              – supercat
              1 hour ago











              0














              Part 1:




              Was the output of the C64 SID chip 8 bit sound?




              Short Answer: yes



              The term 8-bit sound is generally not related to any sample size or speed, but to describe the sound 'qualities' of the 8-bit generation of consoles/computers.




              Part 2:




              So is it correct to say that the audio generation part was really analog,




              Each and every sound generation is in the end analogue. A (single) speaker can only produce one, on-dimensional output over time.



              The difference between using an DAC with a fixed feeding rate and mixing specialized circuits (like with the SID) is about the amount of data needed to feed either.



              Feeding an 8-bit DAC will need one byte per step, so 44 kByte at 44 kHz. A SID based sound system may need only a single byte for the same duration. That is if only a single symmetric frequency is to be outputted. To produce a more complex output, more data is needed. Notable game sounds can already be generated with less than 100 bytes per second.



              Sounds like a compression algorithm, doesn't it? And that's the whole idea here. It's more like playing an instrument(*1) than outputting plain analogue levels. Here as well the output is generated using predefined elements, in effect saving much of the rather slim bandwith 8 bit machines had (*2).



              Viewed from system design having a sound chip where only sound elements have to be set and manipulated is much like having video controllers with programmable functions (CTIA) and/or sprites (VIC). Instead of having the CPU directly manipulating the bitmap data to create an output image, the image is composed from components which in turn need way less bandwith to be manipulates.



              That design view also gives why it got out of fashion - with CPUs fast enough to manipulate bitmap data just in time for output and fast enough to prepare sound data fast enough for straight DAC output, the need for specialized chips vanished.




              Part 3:




              Or maybe what I am really asking is:




              Which is a different question.




              If I have an 8 bit, 44 kHz sample based audio chip, could that chip generate audio which could not be distinguished from a real SID chip?




              Basically yes.



              Quality will be the very same as a SID output recorded at 44 kHz - much like a CD with SID-music.



              The issue here is, as shown, not some 44 kHz DAC, but a CPU able (aka fast enough) to compute (emulate) the various elements of a SID in time - if not going for calculating it ahead of time (aka batch) that is.



              With modern chips/software it's for most parts straight foreward to build a simple SID-alike system. In fact, the web browser your reading this in is already all you need - at least if it supports WebAudio (*3).




              *1 - Which is BTW the idea MIDI is based on. Here sound data is encoded as start stop condition for instruments (simplified). As result a rather narrow 31.25 kBit/s (~8 KiByte/s) can be used to produce a whole orchestras sound.



              *2 - For example the available bandwith for data transport of a 6502 running at 1 MHz is at maximum less 100 KiByte/s. That's for a copy loop, with some processing is gets way lower. Similar for other CPUs of the same time.



              *3 - WebAudio builds its sound generation pipeline from blocks that can be configured to work similar to the SID pipeline. All needed are a few lines JS to configure and link up sound sources, mixers and filters. Of course, this will be only a first iteration. Really remodelling all non linearities/quirks of teh SID will take a lot more.






              share|improve this answer





























                0














                Part 1:




                Was the output of the C64 SID chip 8 bit sound?




                Short Answer: yes



                The term 8-bit sound is generally not related to any sample size or speed, but to describe the sound 'qualities' of the 8-bit generation of consoles/computers.




                Part 2:




                So is it correct to say that the audio generation part was really analog,




                Each and every sound generation is in the end analogue. A (single) speaker can only produce one, on-dimensional output over time.



                The difference between using an DAC with a fixed feeding rate and mixing specialized circuits (like with the SID) is about the amount of data needed to feed either.



                Feeding an 8-bit DAC will need one byte per step, so 44 kByte at 44 kHz. A SID based sound system may need only a single byte for the same duration. That is if only a single symmetric frequency is to be outputted. To produce a more complex output, more data is needed. Notable game sounds can already be generated with less than 100 bytes per second.



                Sounds like a compression algorithm, doesn't it? And that's the whole idea here. It's more like playing an instrument(*1) than outputting plain analogue levels. Here as well the output is generated using predefined elements, in effect saving much of the rather slim bandwith 8 bit machines had (*2).



                Viewed from system design having a sound chip where only sound elements have to be set and manipulated is much like having video controllers with programmable functions (CTIA) and/or sprites (VIC). Instead of having the CPU directly manipulating the bitmap data to create an output image, the image is composed from components which in turn need way less bandwith to be manipulates.



                That design view also gives why it got out of fashion - with CPUs fast enough to manipulate bitmap data just in time for output and fast enough to prepare sound data fast enough for straight DAC output, the need for specialized chips vanished.




                Part 3:




                Or maybe what I am really asking is:




                Which is a different question.




                If I have an 8 bit, 44 kHz sample based audio chip, could that chip generate audio which could not be distinguished from a real SID chip?




                Basically yes.



                Quality will be the very same as a SID output recorded at 44 kHz - much like a CD with SID-music.



                The issue here is, as shown, not some 44 kHz DAC, but a CPU able (aka fast enough) to compute (emulate) the various elements of a SID in time - if not going for calculating it ahead of time (aka batch) that is.



                With modern chips/software it's for most parts straight foreward to build a simple SID-alike system. In fact, the web browser your reading this in is already all you need - at least if it supports WebAudio (*3).




                *1 - Which is BTW the idea MIDI is based on. Here sound data is encoded as start stop condition for instruments (simplified). As result a rather narrow 31.25 kBit/s (~8 KiByte/s) can be used to produce a whole orchestras sound.



                *2 - For example the available bandwith for data transport of a 6502 running at 1 MHz is at maximum less 100 KiByte/s. That's for a copy loop, with some processing is gets way lower. Similar for other CPUs of the same time.



                *3 - WebAudio builds its sound generation pipeline from blocks that can be configured to work similar to the SID pipeline. All needed are a few lines JS to configure and link up sound sources, mixers and filters. Of course, this will be only a first iteration. Really remodelling all non linearities/quirks of teh SID will take a lot more.






                share|improve this answer



























                  0












                  0








                  0







                  Part 1:




                  Was the output of the C64 SID chip 8 bit sound?




                  Short Answer: yes



                  The term 8-bit sound is generally not related to any sample size or speed, but to describe the sound 'qualities' of the 8-bit generation of consoles/computers.




                  Part 2:




                  So is it correct to say that the audio generation part was really analog,




                  Each and every sound generation is in the end analogue. A (single) speaker can only produce one, on-dimensional output over time.



                  The difference between using an DAC with a fixed feeding rate and mixing specialized circuits (like with the SID) is about the amount of data needed to feed either.



                  Feeding an 8-bit DAC will need one byte per step, so 44 kByte at 44 kHz. A SID based sound system may need only a single byte for the same duration. That is if only a single symmetric frequency is to be outputted. To produce a more complex output, more data is needed. Notable game sounds can already be generated with less than 100 bytes per second.



                  Sounds like a compression algorithm, doesn't it? And that's the whole idea here. It's more like playing an instrument(*1) than outputting plain analogue levels. Here as well the output is generated using predefined elements, in effect saving much of the rather slim bandwith 8 bit machines had (*2).



                  Viewed from system design having a sound chip where only sound elements have to be set and manipulated is much like having video controllers with programmable functions (CTIA) and/or sprites (VIC). Instead of having the CPU directly manipulating the bitmap data to create an output image, the image is composed from components which in turn need way less bandwith to be manipulates.



                  That design view also gives why it got out of fashion - with CPUs fast enough to manipulate bitmap data just in time for output and fast enough to prepare sound data fast enough for straight DAC output, the need for specialized chips vanished.




                  Part 3:




                  Or maybe what I am really asking is:




                  Which is a different question.




                  If I have an 8 bit, 44 kHz sample based audio chip, could that chip generate audio which could not be distinguished from a real SID chip?




                  Basically yes.



                  Quality will be the very same as a SID output recorded at 44 kHz - much like a CD with SID-music.



                  The issue here is, as shown, not some 44 kHz DAC, but a CPU able (aka fast enough) to compute (emulate) the various elements of a SID in time - if not going for calculating it ahead of time (aka batch) that is.



                  With modern chips/software it's for most parts straight foreward to build a simple SID-alike system. In fact, the web browser your reading this in is already all you need - at least if it supports WebAudio (*3).




                  *1 - Which is BTW the idea MIDI is based on. Here sound data is encoded as start stop condition for instruments (simplified). As result a rather narrow 31.25 kBit/s (~8 KiByte/s) can be used to produce a whole orchestras sound.



                  *2 - For example the available bandwith for data transport of a 6502 running at 1 MHz is at maximum less 100 KiByte/s. That's for a copy loop, with some processing is gets way lower. Similar for other CPUs of the same time.



                  *3 - WebAudio builds its sound generation pipeline from blocks that can be configured to work similar to the SID pipeline. All needed are a few lines JS to configure and link up sound sources, mixers and filters. Of course, this will be only a first iteration. Really remodelling all non linearities/quirks of teh SID will take a lot more.






                  share|improve this answer















                  Part 1:




                  Was the output of the C64 SID chip 8 bit sound?




                  Short Answer: yes



                  The term 8-bit sound is generally not related to any sample size or speed, but to describe the sound 'qualities' of the 8-bit generation of consoles/computers.




                  Part 2:




                  So is it correct to say that the audio generation part was really analog,




                  Each and every sound generation is in the end analogue. A (single) speaker can only produce one, on-dimensional output over time.



                  The difference between using an DAC with a fixed feeding rate and mixing specialized circuits (like with the SID) is about the amount of data needed to feed either.



                  Feeding an 8-bit DAC will need one byte per step, so 44 kByte at 44 kHz. A SID based sound system may need only a single byte for the same duration. That is if only a single symmetric frequency is to be outputted. To produce a more complex output, more data is needed. Notable game sounds can already be generated with less than 100 bytes per second.



                  Sounds like a compression algorithm, doesn't it? And that's the whole idea here. It's more like playing an instrument(*1) than outputting plain analogue levels. Here as well the output is generated using predefined elements, in effect saving much of the rather slim bandwith 8 bit machines had (*2).



                  Viewed from system design having a sound chip where only sound elements have to be set and manipulated is much like having video controllers with programmable functions (CTIA) and/or sprites (VIC). Instead of having the CPU directly manipulating the bitmap data to create an output image, the image is composed from components which in turn need way less bandwith to be manipulates.



                  That design view also gives why it got out of fashion - with CPUs fast enough to manipulate bitmap data just in time for output and fast enough to prepare sound data fast enough for straight DAC output, the need for specialized chips vanished.




                  Part 3:




                  Or maybe what I am really asking is:




                  Which is a different question.




                  If I have an 8 bit, 44 kHz sample based audio chip, could that chip generate audio which could not be distinguished from a real SID chip?




                  Basically yes.



                  Quality will be the very same as a SID output recorded at 44 kHz - much like a CD with SID-music.



                  The issue here is, as shown, not some 44 kHz DAC, but a CPU able (aka fast enough) to compute (emulate) the various elements of a SID in time - if not going for calculating it ahead of time (aka batch) that is.



                  With modern chips/software it's for most parts straight foreward to build a simple SID-alike system. In fact, the web browser your reading this in is already all you need - at least if it supports WebAudio (*3).




                  *1 - Which is BTW the idea MIDI is based on. Here sound data is encoded as start stop condition for instruments (simplified). As result a rather narrow 31.25 kBit/s (~8 KiByte/s) can be used to produce a whole orchestras sound.



                  *2 - For example the available bandwith for data transport of a 6502 running at 1 MHz is at maximum less 100 KiByte/s. That's for a copy loop, with some processing is gets way lower. Similar for other CPUs of the same time.



                  *3 - WebAudio builds its sound generation pipeline from blocks that can be configured to work similar to the SID pipeline. All needed are a few lines JS to configure and link up sound sources, mixers and filters. Of course, this will be only a first iteration. Really remodelling all non linearities/quirks of teh SID will take a lot more.







                  share|improve this answer














                  share|improve this answer



                  share|improve this answer








                  edited 6 hours ago

























                  answered 6 hours ago









                  RaffzahnRaffzahn

                  59.5k6147246




                  59.5k6147246





















                      0















                      Or maybe what I am really asking is: If I have an 8 bit, 44 kHz
                      sample based audio chip, could that chip generate audio which could
                      not be distinguished from a real SID chip?




                      The short answer: probably. But, given a choice, use standard 44.1 KHz
                      16-bit sampling to reproduce the signal.



                      PCM audio reproduction has two components: frequency range (what's the
                      highest frequency it can reproduce?) and dynamic range (how loud is
                      the background noise level in comparison to the highest signal you can
                      hear?).



                      As described here (in probably far more detail than you care
                      about), the frequency range of properly done 44.1 KHz digital sample
                      reproduction will be about 0-20 KHz, which covers the entire set of
                      frequencies that most people can hear, so there are no issues there.



                      The dynamic range of 8-bit PCM reproduction will be significantly more
                      than 48 dB (assuming you use proper dithering); that is, the loudest
                      signal will be 48 dB higher than the noise floor. This is better than,
                      say, a high-quality cassette tape recording.



                      As others here have pointed out, SID chip is an analogue sound
                      generator with a digital oscillator producing square, triangle and
                      sawtooth waves. Because this is not a PCM sampling system, the
                      number of bits of resolution of the digital oscillator is nearly
                      irrelevant to the noise floor; that will be almost entirely determined
                      by the analogue portions of the system.



                      I can't find any figures on the noise floor of the C64 audio output,
                      but given that it's a very cost-constrained system in so many ways not
                      specifically designed for high-quality musical reproduction (think
                      about the typical speaker system used with it!), it's very likely that
                      the noise level is well above -48 dB.



                      However, another thing to consider is the character of the noise; a
                      standard PCM reproduction system that produces white noise filtered to
                      a certain spectrum may have "background noise" that sounds quite
                      different from that produced by a real C64 reproduction system. Your
                      digital emulation of a SID chip and any subsequent analogue circuitry
                      may or may not emulate this, but if it does that "SID noise" is part
                      of the signal and you want it will above the background noise of the
                      reproduction system.



                      In the end, unless you're particularly resource constrained (to a much
                      greater degree than on a modern PC or phone), you should use 44.1 KHz
                      16-bit reproduction so that you simply don't have to think about the
                      issues in that part of the system, and instead spend your energy
                      focusing on the details of the simulated analogue waveform produced by
                      your SID emulation. If the output of your emulation is not as noisy as
                      the real system and you want that noise, you should add the noise in
                      your emulation, ideally in a way that gives it the same character as
                      the noise created by the real system.






                      share|improve this answer



























                        0















                        Or maybe what I am really asking is: If I have an 8 bit, 44 kHz
                        sample based audio chip, could that chip generate audio which could
                        not be distinguished from a real SID chip?




                        The short answer: probably. But, given a choice, use standard 44.1 KHz
                        16-bit sampling to reproduce the signal.



                        PCM audio reproduction has two components: frequency range (what's the
                        highest frequency it can reproduce?) and dynamic range (how loud is
                        the background noise level in comparison to the highest signal you can
                        hear?).



                        As described here (in probably far more detail than you care
                        about), the frequency range of properly done 44.1 KHz digital sample
                        reproduction will be about 0-20 KHz, which covers the entire set of
                        frequencies that most people can hear, so there are no issues there.



                        The dynamic range of 8-bit PCM reproduction will be significantly more
                        than 48 dB (assuming you use proper dithering); that is, the loudest
                        signal will be 48 dB higher than the noise floor. This is better than,
                        say, a high-quality cassette tape recording.



                        As others here have pointed out, SID chip is an analogue sound
                        generator with a digital oscillator producing square, triangle and
                        sawtooth waves. Because this is not a PCM sampling system, the
                        number of bits of resolution of the digital oscillator is nearly
                        irrelevant to the noise floor; that will be almost entirely determined
                        by the analogue portions of the system.



                        I can't find any figures on the noise floor of the C64 audio output,
                        but given that it's a very cost-constrained system in so many ways not
                        specifically designed for high-quality musical reproduction (think
                        about the typical speaker system used with it!), it's very likely that
                        the noise level is well above -48 dB.



                        However, another thing to consider is the character of the noise; a
                        standard PCM reproduction system that produces white noise filtered to
                        a certain spectrum may have "background noise" that sounds quite
                        different from that produced by a real C64 reproduction system. Your
                        digital emulation of a SID chip and any subsequent analogue circuitry
                        may or may not emulate this, but if it does that "SID noise" is part
                        of the signal and you want it will above the background noise of the
                        reproduction system.



                        In the end, unless you're particularly resource constrained (to a much
                        greater degree than on a modern PC or phone), you should use 44.1 KHz
                        16-bit reproduction so that you simply don't have to think about the
                        issues in that part of the system, and instead spend your energy
                        focusing on the details of the simulated analogue waveform produced by
                        your SID emulation. If the output of your emulation is not as noisy as
                        the real system and you want that noise, you should add the noise in
                        your emulation, ideally in a way that gives it the same character as
                        the noise created by the real system.






                        share|improve this answer

























                          0












                          0








                          0








                          Or maybe what I am really asking is: If I have an 8 bit, 44 kHz
                          sample based audio chip, could that chip generate audio which could
                          not be distinguished from a real SID chip?




                          The short answer: probably. But, given a choice, use standard 44.1 KHz
                          16-bit sampling to reproduce the signal.



                          PCM audio reproduction has two components: frequency range (what's the
                          highest frequency it can reproduce?) and dynamic range (how loud is
                          the background noise level in comparison to the highest signal you can
                          hear?).



                          As described here (in probably far more detail than you care
                          about), the frequency range of properly done 44.1 KHz digital sample
                          reproduction will be about 0-20 KHz, which covers the entire set of
                          frequencies that most people can hear, so there are no issues there.



                          The dynamic range of 8-bit PCM reproduction will be significantly more
                          than 48 dB (assuming you use proper dithering); that is, the loudest
                          signal will be 48 dB higher than the noise floor. This is better than,
                          say, a high-quality cassette tape recording.



                          As others here have pointed out, SID chip is an analogue sound
                          generator with a digital oscillator producing square, triangle and
                          sawtooth waves. Because this is not a PCM sampling system, the
                          number of bits of resolution of the digital oscillator is nearly
                          irrelevant to the noise floor; that will be almost entirely determined
                          by the analogue portions of the system.



                          I can't find any figures on the noise floor of the C64 audio output,
                          but given that it's a very cost-constrained system in so many ways not
                          specifically designed for high-quality musical reproduction (think
                          about the typical speaker system used with it!), it's very likely that
                          the noise level is well above -48 dB.



                          However, another thing to consider is the character of the noise; a
                          standard PCM reproduction system that produces white noise filtered to
                          a certain spectrum may have "background noise" that sounds quite
                          different from that produced by a real C64 reproduction system. Your
                          digital emulation of a SID chip and any subsequent analogue circuitry
                          may or may not emulate this, but if it does that "SID noise" is part
                          of the signal and you want it will above the background noise of the
                          reproduction system.



                          In the end, unless you're particularly resource constrained (to a much
                          greater degree than on a modern PC or phone), you should use 44.1 KHz
                          16-bit reproduction so that you simply don't have to think about the
                          issues in that part of the system, and instead spend your energy
                          focusing on the details of the simulated analogue waveform produced by
                          your SID emulation. If the output of your emulation is not as noisy as
                          the real system and you want that noise, you should add the noise in
                          your emulation, ideally in a way that gives it the same character as
                          the noise created by the real system.






                          share|improve this answer














                          Or maybe what I am really asking is: If I have an 8 bit, 44 kHz
                          sample based audio chip, could that chip generate audio which could
                          not be distinguished from a real SID chip?




                          The short answer: probably. But, given a choice, use standard 44.1 KHz
                          16-bit sampling to reproduce the signal.



                          PCM audio reproduction has two components: frequency range (what's the
                          highest frequency it can reproduce?) and dynamic range (how loud is
                          the background noise level in comparison to the highest signal you can
                          hear?).



                          As described here (in probably far more detail than you care
                          about), the frequency range of properly done 44.1 KHz digital sample
                          reproduction will be about 0-20 KHz, which covers the entire set of
                          frequencies that most people can hear, so there are no issues there.



                          The dynamic range of 8-bit PCM reproduction will be significantly more
                          than 48 dB (assuming you use proper dithering); that is, the loudest
                          signal will be 48 dB higher than the noise floor. This is better than,
                          say, a high-quality cassette tape recording.



                          As others here have pointed out, SID chip is an analogue sound
                          generator with a digital oscillator producing square, triangle and
                          sawtooth waves. Because this is not a PCM sampling system, the
                          number of bits of resolution of the digital oscillator is nearly
                          irrelevant to the noise floor; that will be almost entirely determined
                          by the analogue portions of the system.



                          I can't find any figures on the noise floor of the C64 audio output,
                          but given that it's a very cost-constrained system in so many ways not
                          specifically designed for high-quality musical reproduction (think
                          about the typical speaker system used with it!), it's very likely that
                          the noise level is well above -48 dB.



                          However, another thing to consider is the character of the noise; a
                          standard PCM reproduction system that produces white noise filtered to
                          a certain spectrum may have "background noise" that sounds quite
                          different from that produced by a real C64 reproduction system. Your
                          digital emulation of a SID chip and any subsequent analogue circuitry
                          may or may not emulate this, but if it does that "SID noise" is part
                          of the signal and you want it will above the background noise of the
                          reproduction system.



                          In the end, unless you're particularly resource constrained (to a much
                          greater degree than on a modern PC or phone), you should use 44.1 KHz
                          16-bit reproduction so that you simply don't have to think about the
                          issues in that part of the system, and instead spend your energy
                          focusing on the details of the simulated analogue waveform produced by
                          your SID emulation. If the output of your emulation is not as noisy as
                          the real system and you want that noise, you should add the noise in
                          your emulation, ideally in a way that gives it the same character as
                          the noise created by the real system.







                          share|improve this answer












                          share|improve this answer



                          share|improve this answer










                          answered 1 hour ago









                          Curt J. SampsonCurt J. Sampson

                          346110




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