Add support for Vivado's simulator. By default the run script is using
the Icarus simulator.
If the user want to switch to another simulator, it can be explicitly
specify the required simulator tool in the SIMULATOR variable.
Currently, beside Icarus, Modelsim (SIMULATOR="modelsim") and Vivado's
xsim (SIMULATOR="xsim") is supported.
For consistent simulation behavior it is recommended to annotate all source
files with a timescale. Add it to those where it is currently missing.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
By default inferred output reset signals have an active low polarity. The
axi_ad9361 rst output signal is active high though. Currently when
connecting it to a input reset with active high polarity will generate an
error in IPI.
Fix this by explicitly marking the polarity of the rst signal as active
high.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Replace the open-coded instances of a perfect shuffle in the DAC framer with
the new helper module.
Using the helper module gives well defined semantics and hopefully makes
the code easier to understand.
There are no changes in behavior.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
The perfect shuffle is a common operation in data processing. Add a shared
module that implements this operation.
Having this in a shared module rather than open-coding every instance makes
sure that there are clear and well defined semantics associated with the
operation that are the same each time. This should ease review, maintenance and
understanding of the code.
The perfect shuffle splits the input vector into NUM_GROUPS groups and then
each group in WORDS_PER_GROUP. The output vector consists of
WORDS_PER_GROUP groups and each group has NUM_GROUPS words. The data is
remapped, so that the i-th word of the j-th word in the output vector is
the j-th word of the i-th group of the input vector.
The inverse operation of the perfect shuffle is the perfect shuffle with
both parameters swapped.
I.e. [perfect_suffle B A [perfect_shuffle A B data]] == data
Examples:
NUM_GROUPS = 2, WORDS_PER_GROUP = 4
[A B C D a b c d] => [A a B b C c D d]
NUM_GROUPS = 4, WORDS_PER_GROUP = 2
[A a B b C c D d] => [A B C D a b c d]
NUM_GROUPS = 3, WORDS_PER_GROUP = 2
[A B a b 1 2] => [A a 1 B b 2]
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
The write logic (DMA side) has to be independent from the read logic (DAC side).
In general the FIFO is always ready for the DMA, and every DMA transaction will
interrupt the read-back process, and the module will stop sending data,
until the initialization is finished.
Bringing back the write address tot he DMA clock domain is totally
redundant, so delete it.
Expose the TX configurable driver ports, more specifically the
TX_DIFFCTRL, TX_POSTCURSORE and TX_PRECURSORE for software. This
provides a soft tunning capability of the transmit side of the
transceivers, in cases where the insertion loss of the channel is too
high or low, comparing to the default value supported by the default
configuration of the GTs.
You can find information about these configuration ports under the
section called 'TX Configurable Driver' in the GT transceivers user
guide. (UG476, UG576)
This commit does not contain any functional modification.
Because the wizard generates the attributes in binary, we should use
binary mode too, so we can compare different configurations more easily.
If the req_valid asserts faster than the ID gets synchronized over we
assert the xfer request without being ready to accept data.
This can lead to overflow assertion when using a FIFO like interface.
Data mover/ src axis changes
Request rewind ID if TLAST received during non-last burst
Consume (ignore) descriptors until last segment received
Block descriptors towards destination until last segment received
Request generator changes
Rewind the burst ID if rewind request received
Consume (ignore) descriptors until last segment received
If TLAST happened on last segment replay next transfer (in progress or
completed) with the adjusted ID
Create completion requests for ignored segments
Response generator changes
Track requests
Complete segments which got ignored
Length of partial transfers are stored in a queue for SW reads.
The presence of partial transfer is indicated by a status bit.
The reporting can be enabled by a control bit.
The progress of any transfer can be followed by a debug register.
Drive the descriptor from the source side to destination
so we can abort consecutive transfers in case TLAST asserts.
For AXIS count the length of the burst and pass that value to the
destination instead the programmed one. This is useful when the
streams aborts early by asserting the TLAST. We want to notify the
destination with the right number of beats received.
For FIFO source interface reuse the same logic due the small footprint
even if the stream does not got interrupted in that case.
For MM source interface wire the burst length from the request side to
destination.
The constraint for the synchronizer that synchronizes the sync_status
signal of the link only works correctly for the first link. For other links
no timing exception is applied, which leads to timing failures.
Fix this by using a wildcard constraint for the synchronizer reg number.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
If DDS_DW is equal to DDS_D_DW there is no signal truncation and
consequentially no rounding should be performed. But the check whether
rounding should be performed currently is for if DDS_DW is less or equal to
DDS_D_DW.
When both are equal C_T_WIDTH is 0. This results in the expression
'{(C_T_WIDTH){dds_data_int[DDS_D_DW-1]}};' being a 0 width signal. This is
not legal Verilog, but both the Intel and Xilinx tools seem to accept it
nevertheless.
But the iverilog simulation tools generates the following error:
ad_dds_2.v:102: error: Concatenation repeat may not be zero in this context.
Xilinx Vivado also generates the following warning:
WARNING: [Synth 8-693] zero replication count - replication ignored [ad_dds_2.v:102]
Change the condition so that truncation is only performed when DDS_DW is
less than DDS_D_DW. This fixes both the error and the warning.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
The DISPLAY_NAME of a module is supposed to be a short human readable
description of the IP core.
Currently this is set to the name of the IP, which already has its own
property called NAME.
This causes Platform Designer to display the descriptive labels if the IP
core basically as "$ip_core_name ($ip_core_name)".
The value that all current user of ad_ip_create pass for the description
parameter matches this criteria (And not so much the requirements for the
actual DESCRIPTION property).
Change things, so that the DISPLAY_NAME property is set to what is
currently passed as the description parameter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
The Xilinx's reset interface expect that every reset have an associated
interface and clock signal. The tool will try to find its clock and interface,
and automatically associated clock signal to it.
The PLL resets are individual asynchronous resets. To simplify the design
and avoid invalid critical warnings all the reset interface inference
for the PLL resets were removed.
Most converters refer to their different operating modes as a "Mode X"
(where X is a number) in their datasheet. Each mode has a specific framer
configuration associated with it.
Provide a set of Platform Designer (previously known as Qsys) preset files
for each mode. This allows to quickly select a specific operating mode
without having to lookup the corresponding framer configuration from the
datasheet.
A preset can be selected either in the Platform Designer GUI or from a tcl
script using the apply_preset command. E.g.
add_instance ad9172_transport ad_ip_jesd204_tpl_dac
apply_preset ad9172_transport "AD9172 Mode 10"
The preset files are generated using the scripts/generate_presets.py
script and the scripts/modes.txt file.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
A converter typically only supports a specific subset of framer
configurations.
Add a configuration parameter to select a specific converter part number.
Based on the selected part a mode validation will be performed and if the
selected framer configuration is not supported by the part an error will be
generated.
This helps to catch invalid configurations early on rather than having to
first build the bitstream and then notice that it does not work.
When using "Generic" for the part configuration parameter no validation
will be done and any framer configuration can be selected.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
The exact layout of the input data into the DAC transport layer core
depends on the framer configuration. The number of input channels is
always equal to the NUM_CHANNELS parameter, but the number of samples per
channel per beat depends on the ratio of number of lanes, number of
channels and bits per sample.
It is possible to compute this manually, but this might require in-depth
knowledge about how the JESD204 framer works. Add read-only parameters that
display the number of samples per channel per beat as well as the total
width of the channel data signal.
This information can also be queried in QSys scripts and used to
automatically configure the input pipeline. E.g. like the upack core:
set NUM_OF_CHANNELS [get_instance_parameter_value jesd204_transport NUM_CHANNELS]
set CHANNEL_DATA_WIDTH [get_instance_parameter_value jesd204_transport CHANNEL_DATA_WIDTH]
add_instance util_dac_upack util_upack
set_instance_parameter_values util_dac_upack [list \
CHANNEL_DATA_WIDTH $CHANNEL_DATA_WIDTH \
NUM_OF_CHANNELS $NUM_OF_CHANNELS \
]
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
For a specific set of L, M and NP framer configuration parameters there is
an infinite set of possible values for the S and F configuration parameters
as long as S and F are integer and the following relationship is met
S / F = (L * 8) / (M * NP)
Typically the preferred framer configuration is the one with the lowest
latency. The lowest latency is achieved when S is minimal.
Automatically compute and select this value for S instead of having the
user to manually provide a value.
Since some converters allow modes where S is not minimal provide a manual
overwrite to specify S manually in case somebody wants to use such a mode.
For completeness also add a read-only OCTETS_PER_FRAME (F) parameter that
can be used to verify and check which value for F was chosen.
There is no manual overwrite for F since if L, M, NP and S are set to a
fixed value there is only a single possible value for F, which is computed
automatically.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
The ad_ip_jesd204_tpl_dac currently only supports JESD204 modes that have
both N and N' set to 16.
Newer DACs like the AD9172 support modes where N and N' are not equal to
16. Add support for these modes.
The width of the internal channel data path is set to N, only processing as
many bits as necessary. At the framer the data is up-sized to N' bits with
tail bits inserted as necessary. This data is then passed to the link
layer.
The width at the DMA interface is kept at 16 bits per sample regardless of
the configuration of either N or N'. This is done to keep the interface
consistent with the existing infrastructure it will connect to like upack
and DMA. The data is expected to the LSB aligned, the unused MSBs will be
ignored.
Same is true for the test-pattern data registers. These register keep their
existing 16-bit layout, but unused MSBs will be ignored by the core.
The PN generators are modified to create only N bits of data per sample.
Note that while the core can now support modes with N' = 12 there is still
the restriction that requires the number of frames per beat to be an even
number. Which means that not all modes with N' = 12 can be supported yet.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
The current framer implementation is limited in that it only supports N'=16
and either S=1 or F=1.
Rework the framer implementation to be more flexible and support more
framer setting combinations.
The new framer implementation performs the mapping in two steps. First it
groups samples into frames, as there might be more than one frame per beat.
In the second step the frames are distributed onto the lanes.
Note that this still results in a single input bit being mapped onto a
single output bit and no combinatorial logic is involved. The two step
implementation just makes it (hopefully) easier to follow.
The only restriction that remains is that number of frames per beat must be
integer. This means that F must be either 1, 2 or 4. Supporting partial
frames would result in partial sample sets being consumed at the input,
which is not supported by input pipeline.
The new framer has provisions for handling values for the number of octets
per beat other than 4, but this is not exposed as a configuration option
yet since the link layer can only handle 4 octets per beat. Making the
octets per beat configurable is something for future iterations of the
core.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Adrian Costina
Istvan Csomortani
Lars-Peter Clausen
AndreiGrozav
Laszlo Nagy