Showing posts with label Audio. Show all posts
Showing posts with label Audio. Show all posts

Sunday, 23 June 2019

The Motorola DSP56L307EVM signal processing board

Some information to get a Motorola dsp56L307evm digital signal processing evaluation board from 2001 working on a linux computer in 2019.

The dsp56002evm (top) and dsp56l307evm (bottom)
This is a kind of a next step after getting my old dsp56002evm working under linux.

Introduction


The dsp56L307 is a 24bit (dsp) processor and can run at 150MHz. Together with the Enhanced Filter Coprocessor (efcop) it can generate 270MIPS. I will need to do some real benchmarking, but it can probably be compared with something like a STM32F407 (arm-M4 core at 168MHz) for the calculation speed.However the 24bit Harvard architecture with 3 memory spaces makes it still different and not fully comparable with an arm-M4, which is probably much better in quite some tasks..

I have bought my board from "littlediode" on ebay. While not advertised as such, it did came as the full original EVM kit in original box; that is with documentation, software, power adapter and cable.


The first challenge is to find, roughly 15 years after discontinuation and changing the company from Motorola to Freescale to NXP, all the documentation and tools back in the archives. My board did come with a CD with software like assembler and simulator, but that was not the latest version. The box did not include DSP specific test software for the board.

The wayback machine, does help a bit to find old pages, manuals(pdf) and software back. There are Motorola related pages and Freescale related pages that can be found back. It is advisable to browse a bit between the different snapshots to try to try to find what you are looking for. Another source are old university pages; this dsp was used for some courses.

 Then are all the tools that can be found only for dos / windows available, which gives inherently inconveniences for using those tools on Linux.


Most of the dos/windows tools can be used under a wine console, this shares at least the filesystem. For the dsp56002EVM was it partly needed to use dosbox / dosemu to get the debugger working, which is way more cumbersome in usage.

My current status is:
  • I can use the official (windows) tools and IDE for writing and compiling code.
  • I can use openocd to upload the code and inspect memory and registers to validate the working. (It is not real debuging, for that I would need gdb)
The next sections give a bit more information for this.

Setting up the (Windows) toolchain

The latest version of the required tools are part of the Freescale Symphony studio. This package contains the assembler, linker, c-compiler and tools to change the COFF files in a more universal format.
To get the eclipse based IDE working is it necessary to install an old  Java runtime. I was able to get one from oracle. This does need an account / login.

After installing this with "wine" then we get three sub directories in the install directory:

├── [4.0K]  dsp56720-devtools 
│   ├── [4.0K]  dist
│   │   ├── [4.0K]  dsp56720
│   │   ├── [4.0K]  gcc
│   │   │   ├── [4.0K]  bin  <-- directory with all the tools !
│   │   │   ├── [4.0K]  dsp
│   │   │   ├── [4.0K]  etc
│   │   │   └── [4.0K]  lib
│   │   ├── [4.0K]  gdb
│   │   └── [4.0K]  openocd
│   │       ├── [4.0K]  bin
│   │       └── [4.0K]  driver
│   ├── [4.0K]  doc
│   └── [4.0K]  licenses
├── [4.0K]  eclipse <-- directory with "startsymphony.bat" to start eclips based IDE
│   ├── [4.0K]  configuration
│   ├── [4.0K]  features
│   ├── [ 12K]  plugins
│   └── [4.0K]  readme
└── [4.0K]  sample-projects
    ├── [4.0K]  ASM-Tutorial
    │   └── [4.0K]  Debug
    └── [4.0K]  C-tutorial
        └── [4.0K]  Debug
The batch file to start the eclips based IDE is setting some paths which are necessary for the tool chain to work correctly.  These paths have to be set as well when we want to use the tools, for instance the gcc based c compiler, outside of the IDE. The contents of the batch file is:

~/.wine/drive_c/Symphony-Studio/eclipse$ cat startsymphony.bat 
@echo off

set DDT_HOME=C:\Symphony-Studio\dsp56720-devtools
set G563_EXEC_PREFIX=%DDT_HOME%\dist\gcc\lib\
set PATH=%DDT_HOME%\dist\gcc\bin\;%PATH%
set DSPLOC=%DDT_HOME%\dist\gcc

start C:\Symphony-Studio\eclipse\symphony-studio.exe

exit

The Symphony-studio program can also be started from the start menu. It does look to have sometimes problems to start properly; the windows stay empty. A bit clicking and loading some files seem to help.
The Symphony-studio IDE with assembler output.
With some well chosen command line options is it also possible to get the C-compiler working from the wine console.

$ wineconsole
C:\Symphony-Studio\dsp56720-devtools\dist\gcc\bin>g563c -BC:\Symphony-Studio\dsp56720-devtools\dist\gcc\lib hola.c -o hola.cld

The assembler can also be used stand-alone:

$ wineconsole
C:\Symphony-Studio\dsp56720-devtools\dist\gcc\bin>asm56300 -A -lpass.lst -B pass.asm

Setting up OpenOCD

The current OpenOCD (version 0.10.0) as some issues and is not fully supporting the dsp563xx anymore. The register structure is asking for an "exist" parameter which was missing. And the read and write functions for the memory have an incorrect word size definition. I have submitted two patches to openocd to solve this. The patches can also be found here.  Depending when this is read can this be already included in the latest openocd version.

Further more is a configuration file for openocd needed to make the jtag interface, target and board work together.  This  is still work in progress. The contents for the target is for now as follows:

$ cat dsp5gl307.cfg
if { [info exists CHIPNAME] } { 
   set _CHIPNAME $CHIPNAME
} else {
   set _CHIPNAME dsp56L307
}

if { [info exists ENDIAN] } { 
   set _ENDIAN $ENDIAN
} else {
  # this defaults to a big endian
   set _ENDIAN big
}

if { [info exists CPUTAPID] } {
   set _CPUTAPID $CPUTAPID
} else {
   set _CPUTAPID 0x0181101d
}
#jtag scan chain
jtag newtap $_CHIPNAME cpu -irlen 4 -ircapture 1 -irmask 0x03 -expected-id $_CPUTAPID
#target configuration
set _TARGETNAME $_CHIPNAME.cpu
target create $_TARGETNAME dsp563xx -endian $_ENDIAN -chain-position $_TARGETNAME
#working area at base of ram
$_TARGETNAME configure -work-area-virt 0

And for the EVM board I have made the following configuration file:
$ cat dsp56l307evm.cfg
# Script for freescale DSP56L307EVM
#
# the build in wiggler Jtag interface
source [find interface/parport.cfg]
#jtag speed
adapter_khz 100
reset_config trst_and_srst
# Some additional delays to improve the initialisation.
adapter_nsrst_delay 100
jtag_ntrst_delay   20

source dsp56l307.cfg

#setup flash 
#missing for now; did not work yet

proc enable_fast_clock {} {
    # Programm a factor of 5 into the PLL 
    # 12.288MHz * 3 = 36.8MHz
    mwwx 0xFFFFFD 0x060002
    # enable a timer to see a LED flashing
    mwwx 0xFFFF8F 0x000021
}
# initialize only board specifics - reset, clock, adapter frequency
proc init_board {} {
    global _TARGETNAME
    
    $_TARGETNAME configure -event reset-start {
        # Removed the system reset to be able to have the platform run
        reset_config trst_only
        adapter_khz 500
    }

    $_TARGETNAME configure -event reset-init {
        # this will be done only once!
        # perform a tap and system reset to be sure that the memory is cleared
        reset_config trst_and_srst
        enable_fast_clock
        adapter_khz 500
    }
}

I will upload the final version (when the flash is also working) to git for inclusion into the openocd source.

When starting and working with openocd you should get something as on the pictures below:
Starting openocd with the dsp56l307evm.cfg config file
Connecting with telnet to the debug session

The dsp653xx target of openocd uses a special parameter "memspace" in which of the 4 memory spaces the data is put:
Set memspace #
  • 1 gives access to X data memory  (MEM_X = 0)
  • 2 gives acces to Y data memory (MEM_Y = 1)
  • 3 gives access to the combined X and Y memory called L memory (MEM_L = 3).
  • all other values result in access to P memory or program memory (MEM_P = 2)
This can be used to load data in all memory spaces as follows:

> set memspace 0  # set P memory as default
> load_image program.p 0x0 s19

> set memspace 1   # set X memory as default
> load_image program.x 0x0 s19

> set memspace 2   # set Y memory as default
> load_image program.y 0x0 s19


Checking if open ocd is working by programming one of the timers:
> mwwx 0xFFFF8F 0x21
> mdwx 0xFFFF8F

To change the PLL we can do this:

> mwwx 0xFFFFFD 0x060002    <--  PLL (2+1)=3 x 12.288MHz = 36.864MHz
> mwwx 0xFFFFFD 0x060008    <--  PLL (8+1) =9 x 12.288MHz = 110.592MHz
> mwwx 0xFFFFFD 0x06000B    <--  PLL (11+1) =12 x 12.288MHz = 147.456MHz

Which makes the timer go faster as well.

SREC

The (windows) SREC tool can be used to convert the special CLD COFF files to something more universal that can be loaded by openocd to program the dsp.
This tool has to be run with the following options:
$ srec -r -t 4 infile.cld 
The two necessary options are:
  • The -r option is needed to flip the direction of the words to fix the Endianness difference between host and the dsp. 
  • With the -t 4 option are the words expanded from 24 bit to 32bit.
this image can be loaded by openocd onto a target:
> load_image infile.p 0x0 s19

This srec tool can be called from the Symphony IDE to convert the files after compiling.

Saturday, 3 November 2018

Firewire audio interfaces (hardware)



After my slightly disappointing experience with the Soundblaster Audigy 2 ZS I decided to step up a bit with my "project". In the past was or maybe still is an external firewire audio interface considered the best performance. The clock for audio stream is supposedly more stable than with USB and external is there less noise than inside of computer case. Therefore are all the external devices considered semi professional and come with at least 24bit and 96kHz sample rates in combination with balanced and unbalanced signal inputs and outputs to reduce noise.

The support for Firewire devices under linux is a bit sketchy:
  1. As usual are manufactures of such devices not releasing drivers and or documentation to make (open source) drivers.
  2. From 2007 and on went firewire the way of the dodo in favorite of USB; less computers got firewire buses (Apple stopped supporting it) and more equipment (audio, camera etc.) switched to USB as interface.
  3. The ffado project is providing out of kernel drivers.
  4. The Alsa sound system started to supports also a few firewire interfaces.

The second point makes that it is now a days not so "meaningful" anymore to spend a lot of time in making new or improved drivers for Firewire equipment. At the other hand this makes that it it is possible to buy fairly cheaply old firewire equipment.

I looked a bit on the ffado supported hardware and I found the following three interfaces relative cheaply (<<50 euro) on ebay.
  • Terratec Producer Phase 24 FW
  • Edirol FA-66
  • M-Audio Firewire 410

The front and back of the three boxes are visible in the following picture:
Three (semi professional) Firewire audio interfaces

The following table gives an overview of the analogue interface functionality of the three boxes:
Brand Type Input Output
Mic Line Bits Sample rate Headphone Line Bits Sample rate
Terratec Producer Phase 24 0 2 24 192kHz 2 2 24 192kHz
Edirol (Roland) FA-66 2 2 24 192kHz [2] 4 24 192kHz
M-audio Firewire 410 2 [2] 24 96kHz 2x2 8 24 96kHz

Some notes to the above shown table:
  • The signal level of the headphone output of the Terratec Producer Phase 24 can be changed to line level which results in 4 line outputs.
  • The headphone output of the Edirol FA-66 interface is not independent but always connected with the first two analogue line outputs.
  • The M-audo Firewire 410 interface has not really 4 inputs it can use the 2 microphone inputs or the two line inputs at the back but not both at the same time.   And the two headphone outputs have independent volume control but have both the same signal.
The microphone inputs have the option to enable a 48V phantom power supply for microphones. The have a signal gain control and/or limiter. Both balanced (differential) and unbalanced (single ended) signals can be used as XLR connector or 1/4" jack plug.
At the digital side are there also some differences between the three boxes:
Brand Type Input Output
Midi SPDIF Optical MIDI SPDIF Optical
Terratec Producer Phase 24 [1] [1] 0 [1] [1] 0
Edirol (Roland) FA-66 1 0 1 1 0 1
M-audio Firewire 410 1 1 1 1 1 1
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Some notes to the above shown table:
  • For the Phase 24 interface box are the signals available on the 9 pin sub-d connector. Unfortunately I did not get this breakout cable, but I could find the connections by some measuring. See this blog post for more information.
  • The SPDIF and Optical input of the Firewire 410 cannot be used at the same time. They both are used as one data stream.
  • The Firewire 410 box can use ADAT signals on the digital input which would result in 8 channels more. However I have not tried this yet.
The difference between the three audio interfaces is also very much visible when looked at the mixer possibilities. The next figure shows the three "mixer" views of ffado-mixer.
The ffado mixer interface for the Phase 24 (left), Edirol FA66 (middle) and Firewire 410 (right).
  The Phase 24 interface offers a simple mixer for the 2 inputs and the 3 output signals. This mixer can be mapped to one of the output channels. Furthermore is it possible to select the signal level of the headphone output (line or headphone), select the synchronisation source (internal or spdif) and a kind of 5 level signal gain on the analogue input.
The FA66 interface has an even simpler interface; all the options have buttons at the box itself and cannot configured over software. Only the mixer for the 4 analogue and spdif signal is left in software.
The Firewire 410 interface has the most complex mixing function to match the 2 analogue input, 2 digital input and 8 audio stream signals to the 8 analogue, 2 digital and 2 headphone output signals.
This makes it clear that the M-audio Firewire 410 box is the more "professional" one of the three interfaces. It is a pity that this specific box has only 2 analogue inputs, other, more expensive, versions have more analogue inputs but are in general not well supported by the linux software.
However with firewire and ffado is this not a real issue; as long as the interfaces are on the same bus and the clock is synchronised is it possible to stitch them together as one big audio interface. This give the possibility to sample 8 analogue inputs at the same time. More about this in the software part.

Wednesday, 31 October 2018

Soundblaster Audigy 2 ZS and Linux

I started recently to look a bit around for a good audio card to get to a "studio" quality recordings under Linux. I didn't want to invest much money so I decided to buy something on ebay.

A short investigation pointed me in the direction of the Audigy 2 ZS sound card. A Creative product with interesting specifications:
  • 24bit 96kHz sampling (in and out)
  • 108dB signal noise ratio
  • EMU10K1/2 processor
  • Hardware wavetable synthesizer (4x 16 channel polyphony)
  • Firewire interface (which became the best feature of this card)
And that all in an attractive package for only 10.50 Euro:
A2zs002.jpg
Creative SoundBlaster Audigy 2 ZS
By The original uploader was Swaaye at English Wikipedia. - Transferred from en.wikipedia to Commons by Lockal using CommonsHelper., CC BY-SA 3.0, Link

The mixer options look great:
Alsamixer for Audigy 2 ZS

Reality


While a quick scan suggested that this card is supported under linux is the reality rather disappointing.  The Alsa page gives some information.

It looks that this card is in practice only supported for  44.1kHz or 48kHz sample rate. And the playback and capturing is mainly 16bits with some support for 24bits. This is an alsa limitation and in that sense is the card not really an upgrade from the "crapy" audio of a normal main-board.

This forces me to look a  further for a new solution that does deliver 24bit at a high 96kHz or 192kHz sample rate.

Wavetable synthesizer


This part has some hardware limitations.  According to alsa information page is the address bus only 31bits, which gives problems with a 64bit operating system and more than 4 Gigabyte memory. When this card was released was this probably a none issue, but now with only 64bit Operating systems and 16Gbyte of memory is this a bit of pity.
A work arround is saving some bigger memory block by using the following kernel parameter within the grub2 boot menu:
memmap=2048M\$6144M
It can be unfortunately necessary to play a bit with single (\), double (\\) or triple (\\\) to escape the $ sign within grub2 config file to make this work correctly automatically on boot time.
Furthermore can it be necessary to increase the normal 128MByte memory for the wave table to 200MByte to make the FluidR3_GM.sf2 sound font fit. This can be done by using the following kernel module parameter:
options snd-emu10k1 max_buffer_size=<size_in_MB>
This parameter can be put in the /etc/modprobe.d/alsa-base.conf file.

If this is correctly working can be checked with the following commands:
hansan@Desk-computer:~/Music/mid$ cat /proc/asound/cards
0 [FW             ]: BeBoB - PHASE 24 FW
                      TerraTec Electronic Gmb PHASE 24 FW (id:4, rev:1), GUID 000aac0400239b18 at fw2 
1 [PCH            ]: HDA-Intel - HDA Intel PCH
                      HDA Intel PCH at 0xef410000 irq 30 
2 [HDMI           ]: HDA-Intel - HDA ATI HDMI
                      HDA ATI HDMI at 0xef360000 irq 31 
3 [Audigy2        ]: Audigy2 - SB Audigy 2 ZS [SB0350]
                      SB Audigy 2 ZS [SB0350] (rev.4, serial:0x20021102) at 0xc000, irq 16 
4 [U0x46d0x9a2    ]: USB-Audio - USB Device 0x46d:0x9a2
                      USB Device 0x46d:0x9a2 at usb-0000:00:14.0-3, high speed
And to check the status of the synthesizer:
hansan@Desk-computer:~/Music/mid$ cat /proc/asound/card3/wavetableD1
Device: Emu10k1
Ports: 4
Addresses: 29:0 29:1 29:2 29:3
Use Counter: 0
Max Voices: 64
Allocated Voices: 0
Memory Size: 134217728
Memory Available: 103062476
Allocated Blocks: 866
SoundFonts: 1
Instruments: 14491
Samples: 864
Locked Instruments: 14491
Locked Samples: 864
A new sound font can be loaded with:
hansan@Desk-computer:~/work/sf2$ asfxload  "GeneralUser GS Live-Audigy v1.44.sf2"
hansan@Desk-computer:~/work/sf2$ asfxload -M
DRAM memory left = 100714 kB
And a midi file can be played with:
hansan@Desk-computer:~/work/sf2$ aplaymidi -l
 Port    Client name                      Port name
 14:0    Midi Through                     Midi Through Port-0
 16:0    PHASE 24 FW                      PHASE 24 FW MIDI 1
 28:0    SB Audigy 2 ZS [SB0350]          Audigy MPU-401 (UART)
 28:32   SB Audigy 2 ZS [SB0350]          Audigy MPU-401 #2
 29:0    Emu10k1 WaveTable                Emu10k1 Port 0
 29:1    Emu10k1 WaveTable                Emu10k1 Port 1
 29:2    Emu10k1 WaveTable                Emu10k1 Port 2
 29:3    Emu10k1 WaveTable                Emu10k1 Port 3
hansan@Desk-computer:~/work/sf2$ aplaymidi -p 29.0  test.mid
This does works at least.... But I have say that fluidsynth is maybe an even better solution, given the performance of modern processors.

Conclusion


All with all this was not really a good very invested 10.50 Euro. The selling features like the high bit rate and resolution are not really delivered on the linux platform.
The only good item is a firewire interface, which enables me to look into a different class of audio cards.


(updated and corrected a few items on 3/11/2018)

Saturday, 27 October 2018

Terratec Producer Phase 24FW digital / midi connector pin-out.

I started recently to look into "professional" audio interfaces for Linux. As part of this I have picked up from ebay a Terratec Producer Phase 24 FW in my quest for decent equipment . This is a fairly simple interface with 2 analogue inputs and 4 analogue outputs of which 2 are used for the headpones output at the front of the box.  There are two nice things of this box:
  • Fully supported by the FFADO project.
  • 24bit resolution with 192kHz sample rate.
There is now-a-days not much information to find about this box. There are some old reviews, but the normal website of Terratec is not showing much anymore. However there is a kind of manual archive with some good information and software. This can be found here. The software is of course MAC and windows only and therefore not so important.

Unfortunately the box came without the adapter cable for the digital connector.  This connector contains S/PDIF signals and MIDI signals in and out.  I thought I was fine; I don't have MIDI equipment and I did not want to experiment with SPIDF. However at the time I didn't realize that the S/PDIF signal is mandatory to synchronize multiple interfaces together on one master clock.  Some audio interfaces can even use a world clock signal, but most use the optical or coax S/PDIF or ADAT signal for this purpose.

The missing digital and MIDI signal cable as shown on the advertisement material of the manufacturer
Therefore It was necessary to find out the pin-out of the digital 9 pin sub-d connector.  It was fairly easy to guess the function of the pins on the connector by looking at the more or less standard circuits for S/PDIF and MIDI interfaces and by some poking around with a multimeter.

Based on the measurements and the standard schematics I was able to get to the following schematic / cable diagram of a part of the inside of the Phase 24 FW box and the resulting breakout cable:
Terratec Phase 24 FW breakout cable
Schematic of the digital / midi connector and the breakout cable

The inside of the Phase 24 FW part is not very accurate but simplified to understand the way the breakout cable should be connected. There are much more components on the PCB than showed in the above schematic. Interestingly not only the Midi-in is isolated, but also the SPDIF-out is more or less floating. This is probably to prevent ground loops. The suppressor diode tries to keep the excessive voltages under control.

The following table gives an quick overview of the pinning of the 9 pin female sub-d connector on the box:

Signal group
Pin
Description
MID out
1
+5V out through 220Ohm
2
Out signal
3
Gnd
S/PDIF out
4
Signal out
5
Suppressor diode to GND
Midi in
6
LED (diode in parallel with LED) through 220Ohm
7
Return
S/PDIF in
8
82Ohm input
9
Gnd

Looking at the website and other old data of Terratec I expect that the Phase 22 has the same adapter cable for the digital signals.  And it is very well possible that more products use the same pin out for this 9pin sub-d connector.

Now I have to solder a cable and synchronise my audio interfaces.