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'asoc/fix/rockchip' and 'asoc/fix/rt286' into asoc-linus
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duplex.
Happened when the Playback (or Capture) is running continuously
and Capture (or Playback) is restarted (xrun, manual stop/start...)
Since the RX (or TX) FIFO are only reset when the whole SSI is disabled,
pending samples from previous capture (or playback) session may still
be present. They must be erased to not introduce channel slipping.
FIFO Clear register fields are documented in IMX51, IMX35 reference manual.
They are not documented in IMX50 or IMX6 RM, despite they are
working as expected on IMX6SL and IMX6solo.
Signed-off-by: Arnaud Mouiche <arnaud.mouiche@invoxia.com>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Tested-by: Caleb Crome <caleb@crome.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Previously, SCR.SSIEN and SCR.TE were enabled at once if no capture
stream was also running.
This may not give a chance for the DMA to write the first sample in
TX FIFO before the streaming starts on the PCM bus, inserting void
samples first.
Those void samples are then responsible for slipping the channels.
Signed-off-by: Arnaud Mouiche <arnaud.mouiche@invoxia.com>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Tested-by: Caleb Crome <caleb@crome.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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If the capture is already running while playback is started, it is highly
probable (>80% in a 8 channels scenario) that samples are lost between
the DMA and TX fifo.
The reason is that SIER.TDMAE is set before STCR.TFEN0, leaving a time
window where the FIFO doesn't receive the samples written by the DMA.
This particular case happened only if capture is already enabled as
SCR.SSIEN is already set at the playback startup instant.
Signed-off-by: Arnaud Mouiche <arnaud.mouiche@invoxia.com>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Tested-by: Caleb Crome <caleb@crome.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Most of functions only receive the ssi_private reference and don't have
a knowledge of 'dev' pointer, even for debug purpose.
Signed-off-by: Arnaud Mouiche <arnaud.mouiche@invoxia.com>
Tested-by: Caleb Crome <caleb@crome.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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im6sl reference manual 47.7.4:
"
Bit clock - Used to serially clock the data bits in and out of the SSI port.
This clock is either generated internally (from SSI's sys clock) or taken
from external clock source (through the Tx/Rx clock ports).
[...]
Care should be taken to ensure that the bit clock frequency (either
internally generated by dividing the SSI's sys clock or sourced from
external device through Tx/Rx clock ports) is never greater than 1/5
of the ipg_clk (from CCM) frequency.
"
Since, in master mode, the sysclk is a multiple of bitclk, we can
easily reach a high sysclk value, whereas keeping a reasonable bitclk.
ex: 8ch x 16bit x 48kHz = 6144000, requires a 24576000 sysclk (PM=1)
yet ipg_clk/5 = 66Mhz/5 = 13.2
Signed-off-by: Arnaud Mouiche <arnaud.mouiche@invoxia.com>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Tested-by: Caleb Crome <caleb@crome.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The max number of slots in TDM mode is 32:
- Frame Rate Divider Control is a 5bit value
- Time slot mask registers control 32 slots.
Signed-off-by: Arnaud Mouiche <arnaud.mouiche@invoxia.com>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Tested-by: Caleb Crome <caleb@crome.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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the mixers
The TLV320AIC32x4 has a very flexible mixer on the inputs to the ADCs. Each
mixer has an available set of available pins that can be connected to the
ADC positive and negative pins via three different resistor values. This
allows for configuration of differential inputs as well as doing level
manipulation between sources going into the mixers.
The current code only provides positive pins and I implemented the resistors
in an earlier patch. It turns out that it appears to more accurately model
what's happening to implement each of the pins as a MUX rather than on/off
switches and a mixer. This way each pin can be set to its desired resistor
value. Since there are no switches, the mixer is no longer necessary in the
DAPM path. I set the DAPM paths such that the "off" position of any of the
MUXes turns the path off.
This should allow for any input confiuration available on the codec.
Signed-off-by: Jeremy McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Some definitions to support the PCM5102A codec
by Texas Instruments.
Signed-off-by: Florian Meier <florian.meier@koalo.de>
Changes to original patch by Florian Meier:
* rebased (Makefile and Kconfig
* fixed checkpath errors (spaces, newlines)
* added dt-binding documentation
Signed-off-by: Martin Sperl <kernel@martin.sperl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Manage the hda idisp link using shiny new link APIs. We need to
keep link On while we probe and also hold the reference in runtime
resume and drop in suspend
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Use shiny new link APIs to manage the links. Also remove old link
configuration logic from driver.
We need to keep link and cmd dma to off during active suspend
to allow system to enter low power state and turn it on if
the link and cmd dma was on before active suspend in active
resume.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The HDA links can be switched off when not is use, similarly
command DMA can be stopped as well. This calls for a reference
counting mechanism on the link by it's users to manage the link
power. The DMA can be turned off when all links are off
For this we add two APIs
snd_hdac_ext_bus_link_get
snd_hdac_ext_bus_link_put
They help users to turn up/down link and manage the DMA as well
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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There are a few calls of memset() to stream->resources, but they all
are called in a wrong size, sizeof(unsigned char) * VORTEX_RESOURCE_LAST,
while this field is a u32 array. This may leave the memories not
zero-cleared.
Fix it by replacing them with a simpler sizeof(stream->resources)
instead.
Reported-by: David Binderman <dcb314@hotmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The current device tree representation of the R-Car Sample Rate Converters
(SRC) assumes that they are numbered consecutively, starting from 0. Alas,
this is not the case with the R8A7794 SoC where SRC0 isn't present. In
order to keep the existing device trees working, I'm suggesting to use a
disabled node for SRC0. Teach the SRC probe to just skip disabled nodes.
Signed-off-by: Sergei Shtylyov <sergei.shtylyov@cogentembedded.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Broxton-P reference platform also uses combo jack for audio
connector so we need to set codec pdata to use this based on DMI
match for this board.
Signed-off-by: Ramesh Babu <ramesh.babu@intel.com>
Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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ACPI driver data can be NULL so we need to check that before
dereference the driver data.
Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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On most of audio and music units on IEEE 1394 bus which ALSA firewire stack
supports (or plans to support), CIP with two quadlets header is used.
Thus, there's no cases to queue packets with blank payload. If such packets
are going to be queued, it means that they're for skips of the cycle.
This commit simplifies helper functions to queue a packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In current implementation, packet processing is done in both of software
IRQ contexts of IR/IT contexts and process contexts.
This is usual interrupt handling of IR/IT context for 1394 OHCI.
(in hardware IRQ context)
irq_handler() (drivers/firewire/ohci.c)
->tasklet_schedule()
(in software IRQ context)
handle_it_packet() or handle_ir_packet_per_buffer() (drivers/firewire/ohci.c)
->flush_iso_completions()
->struct fw_iso_context.callback.sc()
= out_stream_callback() or in_stream_callback()
However, we have another chance for packet processing. It's done in PCM
frame handling via ALSA PCM interfaces.
(in process context)
ioctl(i.e. SNDRV_PCM_IOCTL_HWSYNC)
->snd_pcm_hwsync() (sound/core/pcm_native.c)
->snd_pcm_update_hw_ptr() (sound/core/pcm_lib.c)
->snd_pcm_update_hw_ptr0()
->struct snd_pcm_ops.pointer()
= amdtp_stream_pcm_pointer()
->fw_iso_context_flush_completions() (drivers/firewire/core-iso.c)
->struct fw_card_driver.flush_iso_completions()
= ohci_flush_iso_completions() (drivers/firewire/ohci.c)
->flush_iso_completions()
->struct fw_iso_context.callback.sc()
= out_stream_callback() or in_stream_callback()
This design is for a better granularity of PCM pointer. When ioctl(2) is
executed with some commands for ALSA PCM interface, queued packets are
handled at first. Then, the latest number of handled PCM frames is
reported. The number can represent PCM frames transferred in most near
isochronous cycle.
Current tracepoints include no information to distinguish running contexts.
When tracing the interval of software IRQ context, this is not good.
This commit adds more information for current context. Additionally, the
index of packet processed in one context is added in a case that packet
processing is executed in continuous context of the same kind,
As a result, the output includes 11 fields with additional two fields
to commit 0c95c1d6197f ("ALSA: firewire-lib: add tracepoints to dump a part
of isochronous packet data"):
17131.9186: out_packet: 07 7494 ffc0 ffc1 00 000700c0 9001a496 058 45 1 13
17131.9186: out_packet: 07 7495 ffc0 ffc1 00 000700c8 9001ba00 058 46 1 14
17131.9186: out_packet: 07 7496 ffc0 ffc1 00 000700d0 9001ffff 002 47 1 15
17131.9189: out_packet: 07 7497 ffc0 ffc1 00 000700d0 9001d36a 058 00 0 00
17131.9189: out_packet: 07 7498 ffc0 ffc1 00 000700d8 9001e8d4 058 01 0 01
17131.9189: out_packet: 07 7499 ffc0 ffc1 00 000700e0 9001023e 058 02 0 00
17131.9206: in_packet: 07 7447 ffc1 ffc0 01 3f070072 9001783d 058 32 1 00
17131.9206: in_packet: 07 7448 ffc1 ffc0 01 3f070072 90ffffff 002 33 1 01
17131.9206: in_packet: 07 7449 ffc1 ffc0 01 3f07007a 900191a8 058 34 1 02
(Here, some common fields are omitted so that a line is within 80
characters.)
The legend is:
- The second of cycle scheduled for the packet
- The count of cycle scheduled for the packet
- The ID of node as source (hex)
- The ID of node as destination (hex)
- The value of isochronous channel
- The first quadlet of CIP header (hex)
- The second quadlet of CIP header (hex)
- The number of included quadlets
- The index of packet in a buffer maintained by this module
- 0 in process context, 1 in IRQ context
- The index of packet processed in the context
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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for better PCM period granularity
These three commits were merged to improve PCM pointer granularity.
commit 76fb87894828 ("ALSA: firewire-lib: taskletize the snd_pcm_period_elapsed() call")
commit e9148dddc3c7 ("ALSA: firewire-lib: flush completed packets when reading PCM position")
commit 92b862c7d685 ("ALSA: firewire-lib: optimize packet flushing")
The point of them is to handle queued packets not only in software IRQ
context of IR/IT contexts, but also in process context. As a result of
handling packets, period tasklet is scheduled when acrossing PCM period
boundary. This is to prevent recursive call of
'struct snd_pcm_ops.pointer()' in the same context.
When the pointer callback is executed in the process context, it's
better to avoid the second callback in the software IRQ context. The
software IRQ context runs immediately after scheduled in the process
context because few packets are queued yet.
For the aim, 'pointer_flush' is used, however it causes a race condition
between the process context and software IRQ context of IR/IT contexts.
Practically, this race is not so critical because it influences process
context to skip flushing queued packet and to get worse granularity of
PCM pointer. The race condition is quite rare but it should be improved
for stable service.
The similar effect can be achieved by using 'in_interrupt()' macro. This
commit obsoletes 'pointer_flush' with it.
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Phoenix Audio has yet another device with another id (even a different
vendor id, 0556:0014) that requires the same quirk for the sample
rate.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-intel
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Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Format number after 0x in hex.
Cc: Jie Yang <yang.jie@linux.intel.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Signed-off-by: Joonas Lahtinen <joonas.lahtinen@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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When the codec is in standby we do not need to keep the HPPLL active.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The HDMI/DP audio output on ATI/AMD chips got broken due to the recent
restructuring of chmap. Fortunately, Daniel Exner could bisect, and
pointed the culprit commit [739ffee97ed5: ALSA: hda - Add hdmi chmap
verb programming ops to chmap object].
This commit moved some ops from hdmi_ops to chmap_ops, and reassigned
the ops in the embedded chmap object in hdmi_spec instead.
Unfortunately, the reassignment of these ops in patch_atihdmi() were
moved into an if block that is performed only for old chips. Thus, on
newer chips, the generic ops is still used, which doesn't work for
such ATI/AMD chips.
This patch addresses the regression, simply by moving the assignment
of chmap ops to the right place.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=114981
Fixes: 739ffee97ed5 ('ALSA: hda - Add hdmi chmap verb programming ops to chmap object')
Reported-and-tested-by: Daniel Exner <dex@dragonslave.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Current checking for PLL 32KHz mode fails in driver code when
bypassing the PLL. This is due to an incorrect check of PLL
source type when 32KHz clock is provided. Removal of this check
resolves the issue.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This update changes the dividers used for ranges of input MCLK
frequencies, to improve PLL locking for a corner case when at edge
of MCLK frequency input divider range.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Currently PC counter is always synchronised to DAI which means that
when the DAI is disabled, features such as ALC calibration cannot
be executed successfully. This patch makes sure that when the DAI
is disabled, PC counter is set to free-running.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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When the codec is DAI clk slave, and the SRM feature of the PLL
is being used, the enabling of the DAI should occur only after
the PLL has locked to the incoming WCLK. This update adds checking
to the the DAI widget event, so it waits for SRM to lock. There is
also a timeout if that lock doesn't occur within a given time.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Currently, when Codec is I2S master DAI clocks are continuously
generated even if all audio streams have stopped. To improve
efficiency, control of the DAI clocks for master mode have been
moved to a DAPM widget event so they're only enabled as required.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds the Broxton-P machine driver for Intel Broxton-P
reference boards. This machine uses the RT298 codec
Signed-off-by: Ramesh Babu <ramesh.babu@intel.com>
Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The Broxton-P platform has 6 SSPs so we need to add ssp2 thru
ssp5 to DAI list for the driver.
Signed-off-by: Pardha Saradhi K <pardha.saradhi.kesapragada@intel.com>
Signed-off-by: Ramesh Babu <ramesh.babu@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Current rsnd_dmapp_get_id() returns 0xFF as error code if system used
strange connection. It will be used as PDMACHCRn.SRS, but 0xFF is
prohibited number.
In order not to use prohibited number, this patch indicates error message
and returns 0x00 (same as SSI00) in error case.
Special thanks to Dung-san.
Reported-by: Nguyen Viet Dung <nv-dung@jinso.co.jp>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Although the ES8328 does support different rates for capture and
playback, only very limited combinations are supported (8kHz and 48kHz
or 8.0182kHz and 44.1kHz) with most rates required to be symmetric.
Instead of adding a lot of complexity for little gain, let's enforce
symmetric rates.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The values are the same for the DAC and ADC so remove the specific
values and use values with shifts.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This is a refactor in preparation for supporting more sample sizes which
has no functional change.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The chip only supports single reads and writes.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This is always used along with ES8328_CONTROL1_ENREF so there is no
change in the generated code as a result of this fix.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The ADCCONTROL4 and DACCONTROL1 registers are similar but not identical,
with the DACCONTROL1 having each field starting one bit higher than
ADCCONTROL4.
Instead of introducing a magic shift, add new constants for the values
in ADCCONTROL4 and use a second variable to setup the ADC.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This ensures that the clock is setup after its frequency has been set;
the existing code in set_dai_fmt may be called before the clock rate has
been set resulting in an incorrect configuration.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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When snd_pcm_add_chmap_ctls() is called to the PCM stream to which a
chmap has been already assigned, it returns as an error due to the
conflicting snd_ctl_add() result. However, this also clears the
already assigned chmap_kctl field via pcm_chmap_ctl_private_free(),
and becomes inconsistent in the later operation.
This patch adds the check of the conflicting chmap kctl before
actually trying to allocate / assign. The check failure is treated as
a kernel warning, as the double call of snd_pcm_add_chmap_ctls() is
basically a driver bug and having the stack trace would help
developers to figure out the bad code path.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In former commit, drivers in ALSA firewire stack always starts IT context
before IR context. If IR context starts after packets are transmitted by
peer unit, packet discontinuity may be detected because the context starts
in the middle of packet streaming. This situation is rare because IT
context usually starts immediately. However, it's better to solve this
issue. This is suppressed with CIP_SKIP_INIT_DBC_CHECK flag.
This commit enables the same feature as CIP_SKIP_INIT_DBC_CHECK.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In previous commit, this module has no need to reuse parameters of
incoming packets for outgoing packets anymore. This commit arranges some
needless codes for outgoing packet processing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In previous commit, this module has no need to reuse parameters of
incoming packets for outgoing packets anymore. This commit arranges some
needless codes for incoming packet processing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In clause 6.3 of IEC 61883-6:2000, there's an explanation about processing
of presentation timestamp. In the clause, we can see "If a function block
receives a CIP, processes it and subsequently re-transmits it, then the
SYT of the outgoing CIP shall be the sum of the incoming SYT and the
processing delay." ALSA firewire stack has an implementation to partly
satisfy this specification. Developers assumed the stack to perform as an
Audio function block[1].
Following to the assumption, current implementation of ALSA firewire stack
use one software interrupt context to handle both of in/out packets. In
most case, this is processed in 1394 OHCI IR context independently of the
opposite context. Thus, this implementation uses longer CPU time in the
software interrupt context. This is not better for whole system.
Against the assumption, I confirmed that each ASIC for IEC 61883-1/6
doesn't necessarily expect it to the stack. Thus, current implementation
of ALSA firewire stack includes over-engineering.
This commit purges the implementation. As a result, packets of one
direction are handled in one software interrupt context and spends
minimum CPU time.
[1] [alsa-devel] [PATCH 0/8] [RFC] new driver for Echo Audio's Fireworks based devices
http://mailman.alsa-project.org/pipermail/alsa-devel/2013-June/062660.html
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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packet parameter
In packet streaming protocol applied to TASCAM FireWire series, the value
of SYT field in CIP header is always zero, therefore it has no meaning.
There's no need to synchronize packets in both direction for the series.
In current implementation of ALSA firewire stack, driver for the series
uses incoming packet parameter for outgoing packet parameter to calculate
the number of data blocks. This can be simplified because the task of
corresponding driver is to transfer data blocks enough to sampling transfer
frequency.
This commit purges support of full duplex synchronization to prevent
over-engineering implementation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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parameter
On Fireworks board module of Echo Audio, TSB43Cx43A (IceLynx Micro, iCEM)
is used to process payload of isochronous packets. There's an public
document of this chip[1]. This document is for firmware programmers to
transfer/receive AMDTP with IEC60958 data format, however in clause 2.5,
2.6 and 2.7, we can see system design to utilize the sequence of value in
SYT field of CIP header. In clause 2.3, we can see the specification of
Audio Master Clock (MCLK) from iCEM.
Well, this clock is actually not used for sampling clock. This can be
confirmed when corresponding driver transfer random value as the sequence
of SYT field. Even if in this case, the unit generates proper sound.
Additionally, in unique command set for this board module, the format
of CIP is changed; for IEC 61883-6 mode which we use, and for Windows
Operating System. In the latter mode, the whole 32 bit field in second CIP
header from Windows driver is used to represent counter of packets (NO-DATA
code is still used for packets without data blocks). If the master clock
was physically used by DSP on the board module, the Windows driver must
have transferred correct sequence of SYT field.
Furthermore, as long as seeing capacities of AudioFire2, AudioFire4,
AudioFire8, AudioFirePre8 and AudioFire12, these models don't support
SYT-Match clock source.
Summary, we have no need to relate incoming/outgoing packets. This commit
drops reusing SYT sequence of incoming packets for outgoing packets.
[1] Using TSB43Cx43A: S/PDIF over 1394 (2003, Texus Instruments
Incorporated)
http://www.ti.com/analog/docs/litabsmultiplefilelist.tsp?literatureNumber=slla148&docCategoryId=1&familyId=361
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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parameter
Windows driver for BeBoB-based models mostly wait for transmitted packets,
then transfer packets to the models. This looks for the relationship
between incoming packets and outgoing packets to synchronize the sequence
of presentation timestamp.
However, the sequence between packets of both direction has no
relationship. Even if receiving NO-DATA packets, the drivers transfer
packets with meaningful value in SYT field. Additionally, the order of
starting packets is always the same, independently of the source of clock.
The corresponding driver is expected as a generator of presentation
timestamp and these models can select it as a source of sampling clock.
This commit drops reusing SYT sequence from ALSA bebob driver. The driver
always transfer packets with presentation timestamp generated by ALSA
firewire stack, without re-using the sequence of value in SYT field in
incoming packets to outgoing packets.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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