| Commit message (Collapse) | Author | Age |
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Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer and is slightly shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer and is slightly shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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| | | | | | | | | | | | | | | | | | | |
| | | | | | | | | | | | | | | | | | | | |
Use the new snd_pcm_hw_constraint_single() helper function instead of
calling snd_pcm_hw_constraint_minmax() with the same value for min and max
to install a constraint that limits the possible configuration values to a
single value. Using snd_pcm_hw_constraint_single() makes the indented
result clearer.
While we are at it also fix some code style issues in the affected lines.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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While there is nothing wrong with the transfer_ack_begin and
transfer_ack_end callbacks per-se, the last documented user was part of the
alsa-driver 0.5.12a package, which was released 14 years ago and even
predates the upstream integration of the ALSA core and has subsequently
been superseded by newer alsa-driver releases.
This seems to indicate that there is no need for having these callbacks and
they are just cruft that can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When MIDI buffer stores two or more MIDI messages, TASCAM driver
transfers asynchronous transactions including one MIDI message and
extra bytes from second MIDI message.
This commit fixes this bug by clearing needless bytes in the buffer. The
consumed bytes are already calculated correctly, thus the sequence of
transactions is already correct.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In transactions for MIDI messages, the first byte is used for label and
the rest is for MIDI bytes. In current code, these are handled correctly,
while there's a small mistake for loop condition to include meaningless
statement.
This commit adds two local variables for them and improve the loop
condition.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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intension
In the callback function of asynchronous MIDI port, the intension of some
local variables are not clear.
This commit improves them. The 'len' variable is used to calculate the
number of MIDI bytes including in the transaction. The 'consume' variable
is used to return the actual number of consumed bytes in ALSA MIDI buffer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In the callback function of asynchronous MIDI port, some local variables
are declared 'unsigned int', while they're assigned to int value of return
from snd_rawmidi_transmit_peek().
This commit fixes the type.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The given buffer to callback function is cleared in caller side.
This commit removes buffer initialization in callee side.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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By setting the MODALIAS variable in uevents, userspace helpers will be
enabled to load modules via the module alias associated with a device.
This information is required to automatically load HDA codec drivers
instead of having to explicitly request the various modules in the HDA
core code.
[Note that currently the legacy HDA controller driver tries to bind
codec modules manually. It's for supporting the fallback generic
drivers. This new udev modalias support was added rather for ASoC
HDA ext drivers, since this addition itself won't hurt the legacy HDA
-- tiwai]
[Use the common helper function to generate the modalias -- tiwai]
Signed-off-by: Thierry Reding <treding@nvidia.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Finally we have a proper infrastructure to generate the modaliases
automatically, let's move to hda_device_id from the legacy
hda_codec_preset that contains basically the same information.
The patch function hook is stored in driver_data field, which is long,
and we need an explicit cast. Other than that, the conversion is
mostly straightforward. Each entry is even simplified using a macro,
and the lengthy (and error-prone) manual modaliases got removed.
As a result, we achieved a quite good diet:
14 files changed, 407 insertions(+), 595 deletions(-)
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch just adds modalias sysfs entry to each hdaudio bus entry.
[rewritten to call the common helper function by tiwai]
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch provide a new common helper function,
snd_hdac_codec_modalias(), to give the codec modalias name string.
This function will be used by multiple places in the later patches.
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For generating modalias entries automatically, move the definition of
struct hda_device_id to linux/mod_devicetable.h and add the handling
of this record in file2alias helper. The new modalias is represented
with combination of vendor id, device id, and api version as
"hdaudio:vNrNaN".
This patch itself doesn't convert the existing modaliases. Since they
were added manually, this patch won't give any regression by itself at
this point.
[Modified the modalias format to adapt the api_version field, and drop
invalid ANY_ID definition by tiwai]
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Let's leave space for the NUL char otherwise the static checkers
complain that we go beyond the end of the array.
Fixes: 53b3ffee7885 ('ALSA: firewire-tascam: change device probing processing')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The device has no mixer (and identifies itself as such), so just skip
the mixer definition.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For the Zoom R16/24 (tx_length_quirk set), when calculating the maximum
sample frequency, consideration must be made for the fact that four bytes
of the packet contain a length descriptor and consequently must not be
counted as part of the audio data.
This is corroborated by the wMaxPacketSize for this device, which is 108
bytes according for the USB playback endpoint descriptor. The frame size
is 8 bytes (2 channels of 4 bytes each), and the 108 bytes thus work out
as 13 * 8 + 4, i.e. corresponding to 13 frames plus the additional 4 byte
length descriptor.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Zoom R16/24 have a nonstandard playback format where each isochronous
packet contains a length descriptor in the first four bytes. (Curiously,
capture data does not contain this and requires no quirk.)
The quirk involves adding the extra length descriptor whenever outgoing
isochronous packets are generated, both in pcm.c (outgoing audio) and
endpoint.c (silent data).
In order to make the quirk as unintrusive as possible, for
pcm.c:prepare_playback_urb(), the isochronous packet descriptors are
initially set up in the same way no matter if the quirk is enabled or not.
Once it is time to actually copy the data into the outgoing packet buffer
(together with the added length descriptors) the isochronous descriptors
are adjusted in order take the increased payload length into account.
For endpoint.c:prepare_silent_urb() it makes more sense to modify the
actual function, partly because the function is less complex to start with
and partly because it is not as time-critical as prepare_playback_urb()
(whose bulk is run with interrupts disabled), so the (minute) additional
time spent in the non-quirk case is motivated by the simplicity of having
a single function for all cases.
The quirk is controlled by the new tx_length_quirk member in struct
snd_usb_substream and struct snd_usb_audio, which is conveyed to pcm.c
and endpoint.c from quirks.c in a similar manner to the txfr_quirk member
in the same structs.
In contrast to txfr_quirk however, the quirk is enabled directly in
quirks.c:create_standard_audio_quirk() by checking the USB ID in that
function. Another option would be to introduce a new
QUIRK_AUDIO_ZOOM_INTERFACE or somesuch, which would have made the quirk
very plain to see in the quirk table, but it was felt that the additional
code needed to implement it this way would just make the implementation
more complex with no real gain.
Tested with a Zoom R16, both by doing capture and playback separately
using arecord and aplay (8 channel capture and 2 channel playback,
respectively), as well as capture and playback together using Ardour, as
well as Audacity and Qtractor together with jackd.
The R24 is reportedly compatible with the R16 when used as an audio
interface. Both devices share the same USB ID and have the same number of
inputs (8) and outputs (2). Therefore "R16/24" is mentioned throughout the
patch.
Regression tested using an Edirol UA-5 in both class compliant (16-bit)
and "advanced" (24 bit, forces the use of quirks) modes.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Tested-by: Panu Matilainen <pmatilai@laiskiainen.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Preparation for adding Zoom R16/24 quirk.
No functional change.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Refactoring in preparation for adding Zoom R16/24 quirk.
No functional change.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Refactoring in preparation for adding Zoom R16/24 quirk.
No functional change.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Refactoring in preparation for adding Zoom R16/24 quirk.
No functional change.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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TASCAM FireOne is based on OXFW971 and ALSA OXFW driver can support it.
These are values of identical registers.
$ ./firewire-request /dev/fw1 read 0xfffff0050000
result: 97100105
$ ./firewire-request /dev/fw1 read 0xfffff0090020
result: 39373100
This commit adds an entry for this model. This model has physical controls
and its MIDI control messages are transferred to second MIDI data stream
multiplexed in one MIDI conformant data channel.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In IEC 61883-6, sequence multiplexing is applied to MIDI conformant data
channel. As a result, eight MIDI data streams are included in the channel.
Although ALSA AM824 data block processing layer implements this
multiplexing, current OXFW driver doesn't utilize it due to wrong
calculation of MIDI ports.
This commit fixes this bug to add proper calculation. Although this commit
allows to use 8 MIDI data streams, the number of available MIDI ports is
limited by the number of ALSA MIDI ports added by the driver.
Fixes: df075feefbd3('ALSA: firewire-lib: complete AM824 data block processing layer')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Current OXFW driver calculates the number of MIDI ports just before adding
ALSA MIDI ports. It's convenient for some devices with quirks to move
these codes before handling quirks.
This commit implements this idea.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When asynchronous MIDI port is closed before callbacked, the callback
function causes NULL pointer dereference to missing MIDI substream.
This commit fixes this bug.
Fixes: e8a40d9bcb23('ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages')
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The contents of Config ROM in firewire device structure are already
aligned to CPU-endianness. Thus, no need to convert it again.
This commit removes needless conversions
Fixes: 9edf723fd858('ALSA: firewire-digi00x: add skeleton for Digi 002/003 family')
Fixes: c0949b278515('ALSA: firewire-tascam: add skeleton for TASCAM FireWire series')
Reported-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In former commit, metering is supported for BeBoB based models
customized by M-Audio. The data in transaction is aligned to
big-endianness, while in the driver code u16 typed variable is assigned
to the data. This causes sparse warnings.
bebob_maudio.c:651:31: warning: cast to restricted __be16
bebob_maudio.c:651:31: warning: cast to restricted __be16
bebob_maudio.c:651:31: warning: cast to restricted __be16
bebob_maudio.c:651:31: warning: cast to restricted __be16
This commit fixes this bug by using __be16 variable for the data.
Fixes: 3149ac489ff8('ALSA: bebob: Add support for M-Audio special Firewire series')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In former commit, snd_efw_command_get_phys_meters() was added to handle
metering data. The given buffer is used to save transaction result and to
convert between endianness. But this causes sparse warnings.
fireworks_command.c:269:25: warning: incorrect type in argument 1 (different base types)
fireworks_command.c:269:25: expected unsigned int [usertype] *p
fireworks_command.c:269:25: got restricted __be32 [usertype] *
This commit fixes this bug.
Fixes: bde8a8f23bbe('ALSA: fireworks: Add transaction and some commands')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In former commit, u32 data was assigned to __be32 variable instead of an
int variable. This is not enough solution because it still causes sparse
warnings.
dice.c:80:23: warning: incorrect type in assignment (different base types)
dice.c:80:23: expected restricted __be32 [usertype] value
dice.c:80:23: got unsigned int
dice.c:81:21: warning: restricted __be32 degrades to integer
dice.c:81:46: warning: restricted __be32 degrades to integer
This commit fixes this bug.
Fixes: 7c2d4c0cf5ba('ALSA: dice: Split transaction functionality into a file')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some local variables in some functions are typed as unsigned int, while
__be32 value is assigned to them. This causes sparse warnings.
dice-stream.c:50:17: warning: incorrect type in assignment (different base types)
dice-stream.c:50:17: expected unsigned int [unsigned] channel
dice-stream.c:50:17: got restricted __be32 [usertype] <noident>
dice-stream.c:74:17: warning: incorrect type in assignment (different base types)
dice-stream.c:74:17: expected unsigned int [unsigned] channel
dice-stream.c:74:17: got restricted __be32 [usertype] <noident>
This commit fixes this bug.
Fixes: 288a8d0cb04f('ALSA: dice: Change the way to start stream')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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A former commit moves oxfw-related codes to a sub-directory, while it
forgot to remove an entry from Makefile in parent directory.
Fixes: 1a4e39c2e5ca('ALSA: oxfw: Move to its own directory')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When committed to upstream, these four modules had wrong entries for
Makefile. This forces them to be loadable modules even if they're set
as built-in.
This commit fixes this bug.
Fixes: b5b04336015e('ALSA: fireworks: Add skelton for Fireworks based devices')
Fixes: fd6f4b0dc167('ALSA: bebob: Add skelton for BeBoB based devices')
Fixes: 1a4e39c2e5ca('ALSA: oxfw: Move to its own directory')
Fixes: 14ff6a094815('ALSA: dice: Move file to its own directory')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It was forgotten to be removed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The recent commit [7fbe824a0f0e: ALSA: hda - Update mixer name for the
lower codec address] tried to improve the mixer chip name assignment
in the order of codec address. However, this fix was utterly bogus;
it checks the field set in each codec, thus this value is reset at
each codec creation, of course. For really handling this priority,
the assignment has to be remembered in the common place, namely in
hda_bus, instead of hda_codec.
Fixes: 7fbe824a0f0e ('ALSA: hda - Update mixer name for the lower codec address')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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My static checker complains because tscm->spec->midi_capture_ports is
either 2 or 4 but the tscm->tx_midi_substreams[] array has 4 elements so
this is possibly off by one. I have looked at the code and I think it
should be >= instead of > as well.
Fixes: 107cc0129a68 ('ALSA: firewire-tascam: add support for incoming MIDI messages by asynchronous transaction')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We recently tried to add some new code to support turning the LED on and
off but the code in snd_tscm_transaction_reregister() is unreachable.
Fixes: e65e2cb99e44 ('ALSA: firewire-tascam: Turn on/off FireWire LED')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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PCM timer is not always used. For embedded device, we need an interface
to disable it when it is not needed, to shrink the kernel size and
memory footprint, here add CONFIG_SND_PCM_TIMER for it.
When both CONFIG_SND_PCM_TIMER and CONFIG_SND_TIMER is unselected,
about 25KB saving bonus we can get.
Please be noted that when disabled, those stubs who using pcm timer
(e.g. dmix, dsnoop & co) may work incorrectlly.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Nocturn needs the MIDI_RAW_BYTES quirk, like other Novation devices.
Tested that the Nocturn shows up in aconnect, and that it can be used
as a control surface (using the xtor synthesizer patch editor).
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In most cases, we prefer the onboard codec as the primary device, thus
it's better to set it as the mixer name. Currently, however, the
mixer name is updated per the device instantiation order, and user
gets often HDMI/DP or other seen as a mixer chip name. Also, if a
codec name is renamed by the driver, the old chip name might be left
still as the mixer name.
This patch addresses these issues by remembering the chip address that
was referred as the mixer name. When a codec with the same or lower
address gives its name, renew the mixer name accordingly, as it's
either the update of the codec name or we get likely the more
appropriate chip as the reference.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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A few multiple codec drivers do renaming the chip_name string but all
these are open-coded and some of them have even no error check. Let's
make common helpers to do it properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Cirrus codecs have also fine power controls on each widget, thus it
gets benefit from the recent widget power-saving feature. As we
haven't seen any obvious regressions with tests on some MacBooks,
let's try to enable it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We cap the upper bound of "idx" but not the negative side. Let's make
it unsigned to fix this.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Rounding must take place before multiplication with the frame size, since
each packet contains a whole number of frames.
We must also properly consider the data interval, as a larger data
interval will result in larger packets, which, depending on the sampling
frequency, can result in packet sizes that are less than integral
multiples of the packet size for a lower data interval.
Detailed explanation and rationale:
The code before this commit had the following expression on line 613 to
calculate the maximum isochronous packet size:
maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3))
>> (16 - ep->datainterval);
Here, ep->freqmax is the maximum assumed sample frequency, calculated from the
nominal sample frequency plus 25%. It is ultimately derived from ep->freqn,
which is in the units of frames per packet, from get_usb_full_speed_rate()
or usb_high_speed_rate(), as applicable, in Q16.16 format.
The expression essentially adds the Q16.16 equivalent of 0.999... (i.e.
the largest number less than one) to the sample rate, in order to get a
rate whose integer part is rounded up from the fractional value. The
multiplication with (frame_bits >> 3) yields the number of bytes in a
packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back
to an integer, taking into consideration the bDataInterval field of the
endpoint descriptor (which describes how often isochronous packets are
transmitted relative to the (micro)frame rate (125us or 1ms, for USB high
speed and full speed, respectively)). For this discussion we will initially
assume a bDataInterval of 0, so the second line of the expression just
converts the Q16.16 value to an integer.
In order to illustrate the problem, we will set frame_bits 64, which
corresponds to a frame size of 8 bytes.
The problem here is twofold. First, the rounding operation consists
of the addition of 0x0.ffff and subsequent conversion to integer, but as the
expression stands, the conversion to integer is done after multiplication
with the frame size, rather than before. This results in the resulting
maxsize becoming too large.
Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is
0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000.
The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 .
However, if we do the number of bytes calculation in a less obscure way it's
more apparent what the true corresponding packet size is: we get
ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612,
and the 8000 is the number of isochronous packets per second on a high
speed USB connection (125 us microframe interval).
This is fixed by performing the complete rounding operation prior to
multiplication with the frame rate.
The second problem is that when considering the ep->datainterval, this
must be done before rounding, in order to take the advantage of the fact
that if the number of bytes per packet is not an integer, the resulting
rounded-up integer is not necessarily a factor of two when the data
interval is increased by the same factor.
For instance, assuming a freqency of 41 kHz, the resulting
bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or
0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0),
this means that 6 frames per packet are needed, whereas with a data
interval of 2 we need 10.25, i.e. 11 frames needed.
Rephrasing the maxsize expression to:
maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) *
(frame_bits >> 3);
for the above 96 kHz example we instead get
((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value.
We can also do the calculation with a non-integer sample rate which is when
rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn =
0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)):
Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down)
True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56
New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56
This is also corroborated by the wMaxPacketSize check on line 616. Assume
that wMaxPacketSize = 104, with ep->maxpacksize then having the same value.
As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to
(104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize
calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111
(with decimals 111.99988). Clearly, we should get back the 104 here,
which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 .
(The error has not been a problem because it only results in maxsize being
a bit too big which just wastes a couple of bytes, either as a result of
the first maxsize calculation, or because the resulting calculation will
hit the wMaxPacketSize value before the packet is too big, resulting in
fixing the size to wMaxPacketSize even though the packet is actually not
too long.)
Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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