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 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.
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!
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 :)
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 AUDIO_IN_INT_IRQ EXTI15_10_IRQn
/* SAI DMA Stream definitions */#define AUDIO_IN_SAIx_DMAx_CLK_ENABLE() __HAL_RCC_DMA2_CLK_ENABLE()#define AUDIO_IN_SAIx_DMAx_STREAM DMA2_Stream4#define AUDIO_IN_SAIx_DMAx_CHANNEL DMA_CHANNEL_1#define AUDIO_IN_SAIx_DMAx_IRQ DMA2_Stream4_IRQn
#define AUDIO_INx_DMAx_STREAM_IRQHandler DMA2_Stream4_IRQHandler#define AUDIO_IN_SAIx_DMAx_PERIPH_DATA_SIZE DMA_PDATAALIGN_WORD#define AUDIO_IN_SAIx_DMAx_MEM_DATA_SIZE DMA_MDATAALIGN_WORD
#define DMA_MAX_SZE 0xFFFF
/* Select the interrupt preemption priority and subpriority for the DMA interrupt */#define AUDIO_IN_IRQ_PREPRIO ((uint32_t)0x0F)// end SAI AUDIO IN CONFIGURATION
/*------------------------------------------------------------------------------
CONFIGURATION: Audio Driver Configuration parameters
------------------------------------------------------------------------------*/#define AUDIODATA_SIZE 4 /* 16-bits audio data size *//* Audio status definition */#define AUDIO_OK ((uint8_t)0)#define AUDIO_ERROR ((uint8_t)1)#define AUDIO_TIMEOUT ((uint8_t)2)/* Audio In default settings */#define DEFAULT_AUDIO_IN_FREQ BSP_AUDIO_FREQUENCY_48K#define DEFAULT_AUDIO_IN_BIT_RESOLUTION ((uint8_t)24) // can ring buffer load 24bit ? or 32bit ?#define DEFAULT_AUDIO_IN_CHANNEL_NBR ((uint8_t)2) // change to channel count#define DEFAULT_AUDIO_IN_VOLUME ((uint16_t)64)/*------------------------------------------------------------------------------
OUTPUT DEVICES definition
------------------------------------------------------------------------------*//* Alias on existing output devices to adapt for 2 headphones output */#define OUTPUT_DEVICE_HEADPHONE1 OUTPUT_DEVICE_HEADPHONE#define OUTPUT_DEVICE_HEADPHONE2 OUTPUT_DEVICE_SPEAKER /* Headphone2 is connected to Speaker output of the wm8994 *//*------------------------------------------------------------------------------
INPUT DEVICES definition
------------------------------------------------------------------------------*//* MP34DT01TR digital microphone on PCB top side */#define INPUT_DEVICE_DIGITAL_MIC ((uint16_t)0)/* Analog microphone input from 3.5 audio jack connector */#define INPUT_DEVICE_ANALOG_MIC INPUT_DEVICE_INPUT_LINE_1 /**
* @}
*//** @defgroup STM32F769I_DISCOVERY_AUDIO_Exported_Macros STM32F769I_DISCOVERY_AUDIO Exported Macros
* @{
*/#define DMA_MAX(x) (((x) <= DMA_MAX_SZE)? (x):DMA_MAX_SZE)/**
* @}
*//** @defgroup STM32F769I_DISCOVERY_AUDIO_OUT_Exported_Functions STM32F769I_DISCOVERY_AUDIO_OUT Exported Functions
* @{
*/uint8_tBSP_AUDIO_OUT_Init(uint16_t OutputDevice, uint8_t Volume, uint32_t AudioFreq);
voidBSP_AUDIO_OUT_DeInit(void);
uint8_tBSP_AUDIO_OUT_Play(uint16_t* pBuffer, uint32_t Size);
voidBSP_AUDIO_OUT_ChangeBuffer(uint16_t *pData, uint16_t Size);
uint8_tBSP_AUDIO_OUT_Pause(void);
uint8_tBSP_AUDIO_OUT_Resume(void);
uint8_tBSP_AUDIO_OUT_Stop(uint32_t Option);
uint8_tBSP_AUDIO_OUT_SetVolume(uint8_t Volume);
voidBSP_AUDIO_OUT_SetFrequency(uint32_t AudioFreq);
voidBSP_AUDIO_OUT_SetAudioFrameSlot(uint32_t AudioFrameSlot);
uint8_tBSP_AUDIO_OUT_SetMute(uint32_t Cmd);
uint8_tBSP_AUDIO_OUT_SetOutputMode(uint8_t Output);
/* User Callbacks: user has to implement these functions in his code if they are needed. *//* This function is called when the requested data has been completely transferred.*/voidBSP_AUDIO_OUT_TransferComplete_CallBack(void);
/* This function is called when half of the requested buffer has been transferred. */voidBSP_AUDIO_OUT_HalfTransfer_CallBack(void);
/* This function is called when an Interrupt due to transfer error on or peripheral
error occurs. */voidBSP_AUDIO_OUT_Error_CallBack(void);
/* These function can be modified in case the current settings (e.g. DMA stream)
need to be changed for specific application needs */voidBSP_AUDIO_OUT_ClockConfig(SAI_HandleTypeDef *hsai, uint32_t AudioFreq, void *Params);
voidBSP_AUDIO_OUT_MspInit(SAI_HandleTypeDef *hsai, void *Params);
voidBSP_AUDIO_OUT_MspDeInit(SAI_HandleTypeDef *hsai, void *Params);
/**
* @}
*//** @defgroup STM32F769I_DISCOVERY_AUDIO_IN_Exported_Functions STM32F769I_DISCOVERY_AUDIO_IN Exported Functions
* @{
*/uint8_tBSP_AUDIO_IN_Init(uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr);
uint8_tBSP_AUDIO_IN_InitEx(uint16_t InputDevice, uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr);
uint8_tBSP_AUDIO_IN_AllocScratch(int32_t *pScratch, uint32_t size);
uint8_tBSP_AUDIO_IN_GetChannelNumber(void);
voidBSP_AUDIO_IN_DeInit(void);
uint8_tBSP_AUDIO_IN_Record(uint16_t *pData, uint32_t Size);
uint8_tBSP_AUDIO_IN_Stop(void);
uint8_tBSP_AUDIO_IN_Pause(void);
uint8_tBSP_AUDIO_IN_Resume(void);
/* User Callbacks: user has to implement these functions in his code if they are needed. *//* This function should be implemented by the user application.
It is called into this driver when the current buffer is filled to prepare the next
buffer pointer and its size. */voidBSP_AUDIO_IN_TransferComplete_CallBack(void);
voidBSP_AUDIO_IN_HalfTransfer_CallBack(void);
/* This function is called when an Interrupt due to transfer error on or peripheral
error occurs. */voidBSP_AUDIO_IN_Error_CallBack(void);
/* These function can be modified in case the current settings (e.g. DMA stream)
need to be changed for specific application needs */voidBSP_AUDIO_IN_ClockConfig(DFSDM_Filter_HandleTypeDef *hdfsdm_filter, uint32_t AudioFreq, void *Params);
voidBSP_AUDIO_IN_MspInit(void);
voidBSP_AUDIO_IN_MspDeInit(void);
/**
* @}
*//**
* @}
*//**
* @}
*//**
* @}
*/#ifdef __cplusplus
}
#endif#endif /* __STM32F769I_DISCOVERY_AUDIO_H */
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.
I've started to route the main board. It doesn't seam impossible but I'm just hoping I can get away with 4 layers and not have to go to 6 layer. The jump in cost from 4 to 6 is a bit much !
at the moment its 10cm x 6cm which is'nt to bad. No ADC or DAC buffers though and I'm starting to thing I should have included them on the main board rather than put them on the HP + PreAamp + LineDriver cards. The cost would have gone sky high though since the main board has to be 4 layer. All the other cards are 2 layer including the analouge back plane.
Here's a pic of the routing progress with the ARM STM32F7 144pin MC and new DSP. I have never wired up an SD card socket to a micro and no idea what protocol I should use yet ?
I'm thinking it will be 4 or 8bit SPI for speed? I'll get to learning that soon I suppose !
still trying to decide if I should have the 3.3v LDO off this board ! probably a bad idea to leach off the analogue +5v rail !
ben biles
