And what part of the OPs question did you answer tony1954?
Merry Christmas
Regards
Tracing the signal from source to speaker
Hi all,
I'd like to get a better understanding of what happens between a flac file and the sound coming out of a speaker in a digital environment. If any of you can provide insight or point to references for the following, I'd appreciate it. I want to be able to explain this signal path in semi technical terms so that any of us can understand what is happening. To wit:
1, What does a player do? It must acquire (via index) a file of various type, consume it, and output a digital signal. What is the transformation, and what is the type of signal that emerges?
2. What does a DAC do? It takes the digital signal and converts it to an analog signal. What is this conversion? What is the difference between the digital and analog signals?
3. What does the preamp do? What are the differences between preamp processing that create a different type or quality of signal out?
4. What does the amp do? What is the signal out made up of?
5. How do passive or active crossovers recognize the difference in incoming signals?
6. How does the speaker respond to the incoming amplified analog signal?
At some point I would like to include a high level explanation of how the player sends a signal through a switch and router to a NAS (or even a remote service), and how the NAS responds.
I see this as an 10-15 page ppt explaining the functional processes of the digital audio chain. Any help would be appreciated.
Thanks
Chuck
djones51, millercarbon, and oldhvymec, thanks for your responses, especially the recommendation for Audio Science Review. I hadn't visited that forum before. My objective here is to establish a common language and understanding with a few friends so we can agree, more or less, on what we are discussing. Nonoise and tony1954, thanks for posting your selfies on this thread. C |
I'm retired so I no longer make ppt files but I can try to summarize the range of magic necessary to coax music out of baked silicone. I have a music server, DAC, Preamp, amp(s) and a pair of speakers. So that makes me eminently qualified to explain the whole process. (Not really- I own a cell phone too and everything about what a cell phone does is complete magic. For all I know I'm carrying the soul of Lucifer around in my pocket.) Starting with the FLAC file on my music server which is a good place to start since I barely have any idea how it got there in the first place, I use ROON/ROCK to process the digits and transport them to my DAC via a special audiophile USB cable that costs 1100 times more than a basic computer USB cable. Surprisingly, it has a USB A connector on one end and a USB B connector on the other end with a wire in-between just like the standard computer USB cable. So here's what we know based on the cool diagram shown in the ROON program. We have a 44.1 kHz/16bit FLAC file with a particular song title. I use DSP in order to transport the music data in DSD128 format to my DAC. This diagram shows that ROON first does bit depth conversion from 16 bit to 64 bit. Then it makes a sample rate conversion from 44.1 kHz to 5.645 MHz. Next is a Sigma-Delta modulator that converts the data to DSD128 and that is sent into my DAC. Now, I can turn off the DSP and send the music file data to my DAC as PCM in the actual 44.1 kHz/16 bit format as well. ROON is happy to send the data in any format to my DAC. PCM files can be sent in 24 bit formats as well and at higher frequencies than just 44.1 kHz I think it is a matter of preference how someone wants to transmit the music data to the DAC. I have tried both ways, PCM and DSD and did not hear a noticeable difference in my case. I will add that I have not spent much time or effort comparing the two methods of transport. Of course many other imperfect ways to transport the music file data from source to the DAC exist such as AES, Optical, Coaxial. In the end it is still a bit word being transmitted to the DAC. The DAC takes these bit words and converts them into an analog signal. I liken the digital flow of data to the early days of Morse code for communication. The sender and receiver have to start with a handshake to prepare for the information about to be sent. The Morse code operator would send a rapid sequence of taps to get the receiver's attention. If the receiver misses a single tap imagine how garbled the message will be. If the sender and receiver get out of sync the message will be lost, it will be gibberish. Digital code has the same problem. The transfer of the code between sender and receiver has to be synced and the signal clear enough that no bit or digit is missed. This is very basic but more detailed information and even a Bachelors degree can be gathered by searching the web for the specifics about DAC chipsets. The analog output of the DAC chipset is conditioned and amplified by a typical analog amplifier circuit. The DAC chipset is part of the amplifier circuit with it's varying voltage generating the musical signal. The maximum voltage is usually 2 Vrms. So if the digital word for a 1kHz sine wave at maximum signal level was input to the DAC Chipset then it will output an analog sine wave of 1 kHz at 2 Vrms or 5.65 V peak to peak. Comparing this to a turntable and phono cartridge output, it is the phono cartridge generating the voltage based on movement of the stylus exciting a coil as it moves across a magnet. The phono preamplifier circuit takes this small, say 5mV signal and boosts it to 2Vrms. You can get the same 1 kHz signal at 2 Vrms but the 1 kHz will likely vary from 995-1005 Hz due to Wow and Flutter of the record. The digital signal does not vary but it can drop out momentarily if the bit stream gets out of sync or drops a digit or two. (That normally doesn't happen due to error correction routines and algorithms that smooth out the missing data.) Our 2 Vrms output from the DAC can drive a power amplifier but another preamp in the chain helps the sound out mainly by adding more current to the signal which makes the signal less susceptible to either low or high impedances. Solid state preamps can generate 15 Vrms and older tube preamps generated even more voltage. Impedance is like resistance in DC circuits. As the analog signal moves from one box (as in CD or DAC to preamp and then preamp to amp) to the next the inputs on each box has an impedance to control the level of current flow in the cables. If the impedance is lower looking into the next box in the chain then the output voltage can drop unless the amplifier circuit has a large enough power supply to compensate. That means the sound can be affected. When you are selecting a preamp and amp combo it is always wise to look at the output impedance of the preamp and input impedance of the amp. Amps with high input impedance are easier on preamps. Expensive preamps usually feature larger more powerful power supplies. This improves robustness and also helps to maintain good sound when paired with low input impedance amplifiers. Now we are at the amplifier. It has just one job to do- drive the speakers. Well, that can get tricky because speakers are electric motors. Motors have back EMF. That means as a motor moves it is also generating voltage. The faster we move a voice coil the more voltage the voice coil is generating. Our amplifier has to have a big enough power supply to generate enough voltage to overcome back EMF in order to keep the voice coil moving with the music and sounding good. If it is a solid state amp then the transistors go from high resistance to nearly zero resistance depending on the music signal and are basically passing power from the wall plug directly into your speakers. If you have nice speaker cables but are using a cheap power cord- well you only did half the job. And we all know the value of a job half done... |
@tonywinga This is very useful. It'll take me a while to consume it all, and I may have questions. Thanks for your time and attention. Chuck |