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• new digi board in the box :)

  ben biles • 01/02/2019 at 14:21 • 0 comments

  under half an amp @ 12v ! nice :)

  I'd like to keep the final mixer under 12v 1amp ( 12 watts )

  this is with bluetooth control module running

  4x preamps

  2x line drivers

  2x HP drivers

  32 channel IO ADC DAC

  DSP

  ARM stm32f7

  Phantom power 48v on 4 channels

• quick video of board plugged in a being debugged

  ben biles • 12/21/2018 at 10:26 • 0 comments

  just a quick look at the board powered on and in debugging mode.

  the bluetooth module is plugged in and working.

  I'm nearly done porting over the basic mixer code for the DSP ADAU1467 from the old ADAU1446 and i'm

  starting to use safeload code loaded to RAM for the DSP coefficient on the fly changes.

  I think I will make a digital AES3 IO board for bus B and just have 16 channels of differntial analouge IO using Bus A.

  I sort of regret not putting on a 1" oled screen on there just so I can print some basic status info although its not really essential.

  Would have looked cool and there so cheap ! maybe its for the best , could be just a waste of precious power since it will

  eventually be a battery device !

  The Debugger might stay on there for another revision. It might be useful for just updating firmware if nothing else. 

  Its just an STM32F10xx . Obviously I could'nt re-sell the board with the stm debugger software installed , but maybe I could

  leave that to the end user :) I think my worst error on this board revision was to not connect the stm32f10xx to breakouts for

  programming. For some reason I though I could just program it through the JTAG header connected to the STM32F769xx !!

  Obviously that was not the case , so I ended up getting some really thin gauge wire and tacking onto stupid places to flash the

  debugger firmware.  I sought I'll do that again on this board !

  these people are doing some nice Analouge Devices DSP boards but like me i'm not including the AD sigmastudio debugger on the main PCB :) http://www.freedsp.cc/
  

  The next board will have a nice TC2030-IDC 6-Pin footprint on it as the massive 20pin header for JTAG /SWD is a gross waste of space :)

  I ordered one from here

  http://www.tag-connect.com/TC2030-IDC

  unfortunately I had to pay a massive postage fee to get one to Japan , they seam to have run out here !
  

  Anyway , it will save a ton of board space in the long run.. 

• new 4 layer mainboard

  ben biles • 11/20/2018 at 17:18 • 0 comments

  Version 2 parts soldered :)

  more powerful DSP and ARM + memory ( underside )

  ADC and DAC cards are now plugin BUSA & BUSB

  On board ARM debugger on underside of board.

  only 1 x uSD card because I ran out of DMA capable pins on the STM32F7

  but have since discovered if I use a different revision of the chip without DSI

  there are more pins available for second uSD card for recording redundancy.

  JTAG

  SPI is for analogue pre-amp control

  BT for BT module

  I am now working on DSP mixer design for new DSP chip & firmware

  to be done:

  redesign analogue back plane for pre amps and driver cards

  design the AES3 IO card option

  think about what kind of file system I should use for recording. currently FAT

• new digital board arived :)

  ben biles • 09/04/2018 at 08:03 • 0 comments

  Just got my new 4 layer board and 2 layer daughter cards. I need to make a new analogue IO board to but will have to use

  my old one for now.

  I decided to be brave and add the stm32f7 micro and programmer on the underside. I also included DRAM !

  I'm not sure I'll use the JTAG or not. I have a micro usb port hooked up to an STM style serial programmer.

  There should be 2 x uSD cards slots but I didn't appear to have DMA capable pins left on the micro to accommodate that.

  I have broken all the spare micro pins out so maybe i will find a way to make the redundant file storage work.

  I didn't bother with a usb host controller to simplify the design.

  I kept all the parts to 0603 and avoided BGA
  

  The DSP and QSPI flash are QFN style though.

  I'm using a 10 channel fan out clock buffer U5 for 24.576mhz. the ARM micro can generate this but using the external clock

  I can hopefully keep clock jitter to a minimum.

  As you can probably see in pic there a ton of vias ! The DRAM chip is under the micro and there seams to be no good

  way to wire up RAM to a stm32f7 !

  Each trace has only 1 via from micro to dram. Its not running to fast so hopefully it will work.

  The Daughter boards are spit plane and analogue power / GND will come from the analogue connector IO side

  I don't really need 32 channels of balanced analogue audio IO !

  I may well make BUSIO_B a digital IO board !

  I was thinking of AES3 or maybe even something like DANTE

  There's a bluetooth module header and SPI header that runs to the analogue board for digital gain etc.

  I've ordered the parts. the NOR flash qspi chip was near impossible to get , I got some samples on way. 20 week lead time!

  Might need to try and change that part to something less difficult to obtain on the next revision.

  I should have enough power for all the IC's including the digital side of the ADC's and DAC's.

  Lets hope I haven't made to many mistakes and its usable :) I'll be happy to bin the old proto board that didn't even have

  power or ground planes :)
  

• 8 channels looking good

  ben biles • 07/24/2018 at 10:51 • 0 comments

  Its just all the same microphone signal coming through the DSP at slightly different mix levels.

  Its not perfect , a couple of really small glitches near the beginning of the recording.

  Going for perfect here so will have to find out what that is ! The TDM8 clk 12.288mhz and FS are just going down

  jumper wires and are patched into a scope , so probably not helping at all!

  I scrapped the TS screen now and start recording with the bluetooth app.

  after i finish 24bit & 32bit  I will move onto sending the realtime recording data back to the app including nearly real time audio meters from the DSP to the app.

  I'm going to unscrew / disconnect the screen now, waste of space :)

• 8 channel wav poly recording working

  ben biles • 07/24/2018 at 00:38 • 0 comments

  8 channel wav poly 16bit recording is working in TDM8 mode !!!!!!!

  no 24bit or 32bit yet but I can see how to make that work now. Its a question of making the SAI port 32bit , then the buffer and DMA channel 32bit ( WORD )
  
  Its been interesting to work on. There is no timing as such in the application. the data just comes in over DMA and is dumped into a file.

  Its the PCM wav record format header that describes channel count , bit rate , sample rate.

  I thin the player might be more complex as it has to read the WAV file header and then playback in appropiate sample rate bitrate etc
  
  There is the odd click and glitch in the recording , so I'm going to play about with buffer sizes and DMA priority etc until I

  smooth out those problems then I'll add 24 and 32bit capability.
  

  
  I'll need to go to TDM16 also to record the mix track as well as the independent channels.
  

  I'm not sure where the transfer speed limit of usb host is but stm32f7 supports high speed , so the limit might the usb stick itself.
  

  8 channels of 16bit audio

  8 channels 16bit mono
  48000 x 16 bytes
  768kBsec x 30 seconds =  23040000 bytes , ADD 44 bytes for the WAV header = 23040044 bytes 30 sec
  

  the minimum i need will be around double that in 24bit mode and including a 2 channel mix track
  Should be ok , its under 3MByte/sec write speed.

  I haven't really thought about a playback ( review ) and how to do that . I suppose could just pipe the 8 channels back into the DSP mixer so you can preview any recorded channel or the mix. not really sure yet. I could have a kind of mix playback mode..
  
  The recorder / player commands just about work in the bluetooth app. I should disable the TS/screen soon though as it could be the cause of the small glitches in the recording. really I need to send some data back to the bluetooth app suck as record time player recorder app state etc.
  

• 8 channel wav poly recording

  ben biles • 06/26/2018 at 06:07 • 0 comments

  The recorder app now records 8 channel WavPoly with a perfect mono of my microphone input on one channel.

  There's is some nasty digital noise on two other channels so I don't think I'm there yet !

  I'm getting the correct  12.288mhz SK clock and 48khz FS.

  The 12.288mhz / 48khz should clock in the data at the correct speed for the recording.
  

  It could be that although I've selected 16bit slots on the DSP that the adau1446 is sending it in 24bit slots anyway.

  I'm going to figure out TDM8 then try TDM16. I need to record 8 ISO tracks and 2 stereo mix tracks simultaneously.

  Then I need to work on unlimited recording time. It could be as simple as changing the record time variable in the app to a massive for now !  If I hit stop on the app the play closes the file properly.

  then its look at 24 bit mode.

  playback does not support 24 or 32 bit yet. I might need to get the DMA to split 32bit into 2 x 16bit for DMA. not sure DMA supports 32bit buffer samples data or not. more reading to do on stm32f7 DMA!

• Its Alive !!!!!!!!! :):)

  ben biles • 06/21/2018 at 13:58 • 0 comments

  https://soundcloud.com/benbiles/bbbox-proto1-recording-48khz16bitwav

  The 1st test recording is here ! 16bits 2 channel recording from the master mix output. Sorry about the tired voice !

  I had a few hours tough programing to make this work and was completely tired by the time I got it working!

  You can here some audio click / jumps. This is because the ARM is master and generating its own clock with

  the DSP in slave mode to it. The ADAU1446 doesn't have any ASRCs to sync the 2 clocks and doesn't like the clock mismatch :)
  

  I had to battle with re-assigning SAI ( serial audio interface) pins on the stm32f7 demo board. I only have one SD pin available

  on SAI_block_B and have to switch between player and recorder mode using the one SD pin and sharing the rest for FS and CLK.

  I think I can can get 16 channels IO this way in TDM though so it should be enough for testing. Although it will be 16 channels in or out and not both at the same time ! or maybe they can be clocked in and out at the same time , not sure.
  

  The next job is to get the SAI interface working in Slave RX TX Synchronious to the DSP MCLK then the whole system will be running off

  the one high quality clock / fan out buffer. I already have the DSP setup for ISO recording.

  I think after that my next coding headache will come from writing WAV poly. 8 channel WAV audio PCM files.

  I will need to look up the spec of them and see how they are made. hopfully I can just write the TDM in slots and define in the header

  in some kind of PCM descriptor ? anyone already know the answer? would be really helpful :)

• C++ STM32f7 audio recorder application demo code

  ben biles • 05/11/2018 at 14:31 • 0 comments

  I started to modify ARM stm32f7 demo audio recorder / playback application.

  I'm going to connect TDM8 channels to the SAI on the ARM demo board.

  The demo code is horrible for me. its a forest of board support packages BSP's and HAL's. I'm sure in a way its over complicating the basic idea that the SAI stream will be copied using DMA to the circular buffer. then when the buffer is half full the data is written to the USB stick ?! 

  I'm thinking there does't need to be any timing as the data is just dumped onto the stick as a file as it comes in. I'm hoping there is a way to append data to the file on the stick rather than determining the audio record time as in the demo!

  /*------------------------------------------------------------------------------
                            USER SAI defines parameters
   -----------------------------------------------------------------------------*/
  /** CODEC_AudioFrame_SLOT_TDMMode In W8994 codec the Audio frame contains 4 slots : TDM Mode
    * TDM format :
    * +------------------|------------------|--------------------|-------------------+ 
    * | CODEC_SLOT0 Left | CODEC_SLOT1 Left | CODEC_SLOT0 Right  | CODEC_SLOT1 Right |
    * +------------------------------------------------------------------------------+
    */
  /* To have 2 separate audio stream in Both headphone and speaker the 4 slot must be activated */
  #define CODEC_AUDIOFRAME_SLOT_0123                   SAI_SLOTACTIVE_0 | SAI_SLOTACTIVE_1 | SAI_SLOTACTIVE_2 | SAI_SLOTACTIVE_3
  /* To have an audio stream in headphone only SAI Slot 0 and Slot 2 must be activated */ 
  #define CODEC_AUDIOFRAME_SLOT_02                     SAI_SLOTACTIVE_0 | SAI_SLOTACTIVE_2 
  /* To have an audio stream in speaker only SAI Slot 1 and Slot 3 must be activated */ 
  #define CODEC_AUDIOFRAME_SLOT_13                     SAI_SLOTACTIVE_1 | SAI_SLOTACTIVE_3
  
                                                         
  /////////    ben tries to add arduino header SAI1_Block_B , no idea if this will work !
  /////////    arduino header example I found used same defines as the onboard DAC
  
  //configure SAI1 for Arduino header
  
  #define AUDIO_OUT_SAIx                           SAI1_Block_A
  #define AUDIO_OUT_SAIx_CLK_ENABLE()              __HAL_RCC_SAI1_CLK_ENABLE()
  #define AUDIO_OUT_SAIx_CLK_DISABLE()             __HAL_RCC_SAI1_CLK_DISABLE()
  #define AUDIO_OUT_SAIx_AF                        GPIO_AF6_SAI1
  #define AUDIO_OUT_SAIx_MCLK_ENABLE()             __HAL_RCC_GPIOF_CLK_ENABLE()
  #define AUDIO_OUT_SAIx_MCLK_GPIO_PORT            GPIOF
  #define AUDIO_OUT_SAIx_MCLK_PIN                  GPIO_PIN_7 //PF7  MCLK
  #define AUDIO_OUT_SAIx_SD_FS_CLK_ENABLE()        __HAL_RCC_GPIOF_CLK_ENABLE()
  #define AUDIO_OUT_SAIx_SD_FS_SCK_GPIO_PORT       GPIOF
  #define AUDIO_OUT_SAIx_FS_PIN                    GPIO_PIN_9 //  PF9  LRCK
  #define AUDIO_OUT_SAIx_SCK_PIN                   GPIO_PIN_8  // PF8  BCK
  #define AUDIO_OUT_SAIx_SD_PIN                    GPIO_PIN_6  // PF6  DATA
  
  /* SAI DMA Stream definitions */
  #define AUDIO_OUT_SAIx_DMAx_CLK_ENABLE()         __HAL_RCC_DMA2_CLK_ENABLE()
  #define AUDIO_OUT_SAIx_DMAx_STREAM               DMA2_Stream1
  #define AUDIO_OUT_SAIx_DMAx_CHANNEL              DMA_CHANNEL_0
  #define AUDIO_OUT_SAIx_DMAx_IRQ                  DMA2_Stream1_IRQn
  #define AUDIO_OUT_SAIx_DMAx_PERIPH_DATA_SIZE     DMA_PDATAALIGN_HALFWORD
  #define AUDIO_OUT_SAIx_DMAx_MEM_DATA_SIZE        DMA_MDATAALIGN_HALFWORD
  #define DMA_MAX_SZE                              0xFFFF
     
  #define AUDIO_OUT_SAIx_DMAx_IRQHandler           DMA2_Stream1_IRQHandler
  
  /* Select the interrupt preemption priority and subpriority for the DMA interrupt */
  #define AUDIO_OUT_IRQ_PREPRIO                    ((uint32_t)0x0E) 
  
  
  ///////  end ben tried to add SAI1_Block_A
  
  /*------------------------------------------------------------------------------
                          AUDIO IN CONFIGURATION
  ------------------------------------------------------------------------------*/
  
  /* SAI peripheral configuration defines */
  #define AUDIO_IN_SAIx                           SAI1_Block_B
  #define AUDIO_IN_SAIx_CLK_ENABLE()              __HAL_RCC_SAI1_CLK_ENABLE()
  #define AUDIO_IN_SAIx_CLK_DISABLE()             __HAL_RCC_SAI1_CLK_DISABLE()
  #define AUDIO_IN_SAIx_AF                        GPIO_AF6_SAI1
  #define AUDIO_IN_SAIx_SD_ENABLE()               __HAL_RCC_GPIOE_CLK_ENABLE()
  #define AUDIO_IN_SAIx_SD_GPIO_PORT              GPIOE
  #define AUDIO_IN_SAIx_SD_PIN                    GPIO_PIN_3
  #define AUDIO_IN_INT_GPIO_ENABLE()               __HAL_RCC_GPIOJ_CLK_ENABLE()
  #define AUDIO_IN_INT_GPIO_PORT                   GPIOJ
  #define AUDIO_IN_INT_GPIO_PIN                    GPIO_PIN_12
  #define...

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• backplane for prototype 1

  ben biles • 04/27/2018 at 17:21 • 0 comments

  The backplane is in and after adding 2 bodge wires ( had 2 serial traces the wrong away around ! ) it works
  
  The next version will have the upgraded ADAU1467 DSP and the design will be even more modular.
  the backplane will have an the STM32f7 and ADAU1467 and all other cards will be slotted in !
  Hopefully all the sockets will be PCB mounted to! I really want to loose this box full of wires and
  connectors :)

  This back-plane has 4 differential preamp channels and 4 differential transformer style outputs with 1 high quality

  headphone driver plugged in.

  The ADC DSP DAC 8 chan board is hidden under the ARM dev board and has a nest of jumper wires connecting up the I2C TDM audio UART Bluetooth 4.0 etc. the Bluetooth module is gaffer taped in there somewhere !

  It sounds very noise free and better than my sound devices 633 mixer ! BUT i'm not even going to compare them until

  revision 2 since these wires are all over the place :)

  Also if you read that far back rev 1 of the ADC DSP DAC still doesn't have ground planes since I did'nt know how to make them

  back then!

  I'm still working on the TDM8 ARM recorder/playback app and will post some test WAV poly files here when I get that working properly.
  

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