Hi, Im new on DSPs, so i have a few questions, not a bluetooth one.
1) I want BT audio input at 44.1Khz on ADAU1701 I2S Input and at Outputs 48Khz, i will need an IC sample converter? Sample converters looses quality?
2) Analog Devices haves an Adau Family DSP that supports Internal Sample rate converter?
3) Which ADC/TDM 4/8 Channels ICs is recomended for better compatibility with ADAUs
4) Which DAC/TDM 4/8 Channels ICs is recomended for better compatibility with ADAUs
About Sigma Studio Design:
1) Is possible to have an 32 bands Grafic Equalizer? if yes, how to?
2) What is better way to bypass an object, like disable/bypass filter or disable/bypass dynamic bass, Analog Devices can put this option on all modules to keep design easy? ( I have found that option on General 2nd Order Filter, at to left corner, blue circle, and its is perfect! )
3) Can i have an auto gain on Input, so audio from any inputs have same level inside DSP?
You are asking many questions so it is difficult for me to answer all of them. Many of my answers will bring up more questions so I guess I just need to start with some answers to keep this response from being the new novel on the best sellers list! :)
Question #1 and #2) If you use a Bluetooth receiver you will need to derive the master clock from the Bluetooth unless it is able to sync to a common master clock which I do not think it will do. I am not an expert on Bluetooth hardware. So then you will need to switch the 1701 over to a new master clock which means the PLL needs to lock to the new source. In that case then you will not need a sample rate converter. However, this is not easy to implement and it can be a risk for issues down the road. If Bluetooth signals get weak or are lost what is the behavior of the clocks etc.? Using an ASRC is better. If size and power are not an issue, then I think you should use the ADAU1452 which does have sample rate converters. If size and power are an issue then look at the ADAU1787. It will be a great solution for you in many ways. There are sample rate converters. There are built in converters and two serial ports and two DSP cores. It is VERY small since it is in a wafer level package which can be great for some applications and bad for others. It also can only support up to 1.8V for analog and digital power.
Our sample rate converters do sound pretty good. I come from a Pro Audio recording background so I know how some purists can be. However, it is a Bluetooth signal... not exactly a super high fidelity high bit depth and high sample rate source. :)
Question # 3, I would recommend you use the ADAU1372 codec. It has ASRCs on the serial ports so it will simply follow the sample rate being fed to it. This simplifies the clocking a great deal. This is good for use with Bluetooth but in this case it will not help you get into the 1701 unless you will want to go through another AtoD conversion and I think that you would not want to do that. With the ADAU1452 I would recommend using the AD1938, AD1937, AD1939 family of codecs. The ADAU1977/78/79 family of ADCs are also really good.
If you need many DACs then look at the ADAU1962 or 1966 family of DACs. ( The "A" versions, ADAU196xA, are optimized for single ended use and for 3.3V) Otherwise, the AD1938 codecs are good to use.
For your SigmaStudio design questions:
1) Yes, you can have a 32 band graphic EQ. There are many ways to do it. If you are using the 1701 then you are limited by the small number of instructions (MIPS) available. I will attach an example file that would use the AUX ADCs for input. Since there are only four AUX ADC inputs this would limit it to four bands. The example uses one since there is one pot on the evaluation board and it shows DC sources for the other three controls. These can be changed to the other three available AUX ADCs. However, if you have a system controller you can simply change the DC setting or directly select the filter. Then you can do more than four bands per DSP. You can probably go as high as 16 bands or at least 10 if you want a lot of curves for each filter. If you want to calculate the filter coefficients in a system controller then you might be able to get it up to 32 but I am not certain. Like I said, there are so many ways to do it. You can find some examples on this forum.
ADAU1701 Graphic EQ Example.zip
Yes, a lot of the filters have a bypass button. Some are hidden in the pop up window like in the above example. Sometimes you want to bypass more than one thing so in that case you simply use the library cells to build a bypass circuit like you would do with a relay. Here is an example.
Your third question is a little more difficult to explain and implement.
You can do level sensing and compression. You can heavily compress a signal to keep them all at a similar level. You can implement and very slow moving gain control that would adapt the level of the audio but not squash it to oblivion but then you have to be able to tolerate some clipping while the AGC adapts. This all becomes how the time constants are setup and your application and how you want it to sound.
So this can be another book written here...