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so is there a resistor that can be changed so this display is as bright as other colors?
Is there any reason you've chosen a non standard sampling rate? Instead of 42031.52 (1142 ticks) you could have gone for 44117.65 with 1088 systicks.
In any case 42KHz is just 5% away from 44.1KHz so I expect the internal PCM5102 PLL to lock.
what about pressing the inst button and pressing an encoder down, or inst and menu?
xavier,
with the recent activity in the firmware i think it would be a good time to implement the combo settings page after i4 (i8 respectively)
did you do any thinking in that direction? as i said before i think most crucial settings will be:
midi channel for each instrument
midi offset, transpose +-48 semitones all incoming midi on an instrument
I have a few issues to clarify:
- what is the crystal you are using for the F4? I've put a 12MHz one, it needs to be a divisor of 48MHz because of the USB
- right now I've connected Channel1 Left/Right to one stereo jack AF1 and Channel2 Left/Right to AF2
would it be better to have Channel1 connected to both AF1/AF2 tips and Channel2 connected to both AF1/AF2 rings? This second configuration will allow the use of only one PCM5102 and allow the use on mono cables as before.
- because I'm mixing the 2 lefts and the 2 rights for the headphones amp there is a small (-43dB) signal crosstalk between the before mentioned channels. If one doesn't want this if should replace R30/R38 with 0 ohm resistors and not solder the TDA2822M and its capacitors.
- a very important issue is the output sample frequency. The PCM5102 has its own PLL which locks to the WS signal. If the F4 isn't able to provide a constant stream of data this might be a problem. The F4 can also generate instead the system clock and send it to PCM5102 (just 2 extra pins of F4).
- there is an error between the desired sample rate and what the F4 can provide with the current crystal - for example instead of 44KHz the real audio frequency is 43.4KHz (an error of 1.35%). A 6% error is already a semitone. I haven't studied yet the current firmware - is the software able to compensate for this? what is the sampling rate you are using now?
- 24 and 32 bit samples need to be loaded in the SPI data register twice because that's only 16bit wide in I2S mode so the best way would be to DMA the data from the memory and let the MCU do something else in this time
i could not find a link to a schematic.
find the 8timbres version of 2.08a in the 8timbres thread...
You can have both. The encoder changes the volume globally and the software per channel / instrument. The MCP42010 can jump instantly to any set value. These will have to be bit-banged though because all the SPI/I2S interfaces are used.
I have started changing the schematic for 4 audio channels and I've ordered some more components. I will probably also put a TDA2822M as a headphones stereo amplifier. This is linear rather than class D because I don't want the extra switching noise and 500mW per channel is plenty for headphones.
I've changed all MIDI headers to 3 pins. The new encoder has a separate push switch connected. I will probably have to chose between separating the switches of all the other encoders and the P3 analog input header because I've started to run low on pins. Which one is better? Also is it worth having 4 separate LEDs or more LEDs? If so maybe it's worth dropping a second MCP23S17 to handle the extra switches and LED outputs.
An even better option would be to replace the port expanders with a STM32F103 (only $1.20) running at 8MHz (internal RC clock) which can handle all the encoders, switches and LEDs independently and free a lot of extra pins in the STM32F4 and it can also keep track of the encoder values.
Anyone else thinks that having more audio outputs is a good idea? What other changes would you like to see in PreenFM2?
