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* ALSA: USB-audio: Add support for Novation Nocturn MIDIcontrol surfaceRicard Wanderlof2015-10-161-0/+9
| | | | | | | | | | 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>
* ALSA: hda - Update mixer name for the lower codec addressTakashi Iwai2015-10-153-1/+5
| | | | | | | | | | | | | | | | | 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>
* ALSA: hda - consolidate chip rename functionsTakashi Iwai2015-10-156-39/+53
| | | | | | | | 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>
* ALSA: hda - Enable widget power saving for Cirrus codecsTakashi Iwai2015-10-151-0/+1
| | | | | | | | | 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>
* ALSA: oss: underflow in snd_mixer_oss_proc_write()Dan Carpenter2015-10-151-1/+2
| | | | | | | | 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>
* ALSA: usb-audio: Fix max packet size calculation for USB audioRicard Wanderlof2015-10-131-2/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* Merge branch 'for-linus' into for-nextTakashi Iwai2015-10-1315-40/+69
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| * ALSA: hda - Fix inverted internal mic on Lenovo G50-80David Henningsson2015-10-131-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | Add the appropriate quirk to indicate the Lenovo G50-80 has a stereo mic input where one channel has reverse polarity. Alsa-info available at: https://launchpadlibrarian.net/220846272/AlsaInfo.txt Cc: stable@vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1504778 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hdac: Explicitly add io.hVinod Koul2015-10-131-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Compiling the hdac extended core on arm fails with below error: sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_writel': >> sound/hda/ext/hdac_ext_bus.c:29:2: error: implicit declaration of >> function +'writel' [-Werror=implicit-function-declaration] writel(value, addr); ^ sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_readl': >> sound/hda/ext/hdac_ext_bus.c:34:2: error: implicit declaration of >> function +'readl' [-Werror=implicit-function-declaration] return readl(addr); This is fixed by explicitly including io.h Fixes: 99463b3a3994 - ('ALSA: hda: provide default bus io ops extended hdac') Reported-by: kbuild test robot <lkp@intel.com> Suggested-by: Mark Brown <broonie@kernel.org> Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * Merge tag 'asoc-fix-v4.3-rc4' of ↵Takashi Iwai2015-10-079-38/+58
| |\ | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v4.3 Quite a few fixes here but they're all very small and driver specific, none of them really stand out if you aren't using the relevant hardware but they're all useful if you do happen to have an affected device.
| | *-. Merge remote-tracking branches 'asoc/fix/tlv320aic3x' and 'asoc/fix/wm8962' ↵Mark Brown2015-10-072-9/+15
| | |\ \ | | | | | | | | | | | | | | | into asoc-linus
| | | | * ASoC: wm8962: balance pm_runtime_enableJiada Wang2015-09-301-1/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | pm_runtime_enable is called in probe to enable runtime PM for wm8962 codec, but pm_runtime_disable isn't called in remove callback, nor is called in error path if probe fails after runtime PM is enabled, this causes unbalanced pm_runtime_enable. This patch Adds pm_runtime_disable in remove callback and error path, to balance pm_runtime_enable. Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * | ASoC: tlv320aic3x: Prevent writing reserved registers on tlv320aic3104 CODECsRick Mann2015-10-011-8/+11
| | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | The current code writes a set of registers that are reserved on the tlc320aic3104. The change skips those registers for that IC. Signed-off-by: Rick Mann <rmann@latencyzero.com> Signed-off-by: Mark Brown <broonie@kernel.org>
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| | *-----------. \ Merge remote-tracking branches 'asoc/fix/db1200', 'asoc/fix/dwc', ↵Mark Brown2015-10-077-29/+43
| | |\ \ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 'asoc/fix/imx-ssi', 'asoc/fix/maintainers', 'asoc/fix/rt5645', 'asoc/fix/sgtl5000' and 'asoc/fix/tas2552' into asoc-linus
| | | | | | | | | * | ASoC: tas2552: fix dBscale-min declarationAndreas Dannenberg2015-10-061-1/+1
| | | | | | | | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The minimum volume level for the TAS2552 (control register value 0x00) is -7dB however the driver declares it as -0.07dB. Running amixer before the patch reports: dBscale-min=-0.07dB,step=1.00dB,mute=0 Running amixer with the patch applied reports: dBscale-min=-7.00dB,step=1.00dB,mute=0 Signed-off-by: Andreas Dannenberg <dannenberg@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org
| | | | | | | | * | ASoC: sgtl5000: fix wrong register MIC_BIAS_VOLTAGE setup on probeGianluca Renzi2015-09-301-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Gianluca Renzi <gianlucarenzi@eurekelettronica.it> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org
| | | | | | | | * | ASoC: sgtl5000: fix error message output for MicBias voltageGianluca Renzi2015-09-291-1/+1
| | | | | | | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Takashi Iwai <tiwai@suse.com> Cc: Fabio Estevam <fabio.estevam@freescale.com> Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Gianluca Renzi <gianlucarenzi@eurekelettronica.it> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | | | | | * / ASoC: rt5645: Correct the naming and setting of ADC Boost Volume ControlOder Chiou2015-10-012-10/+12
| | | | | | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Oder Chiou <oder_chiou@realtek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | | | * | / ASoC: imx-ssi: Fix DAI hardware signal inversionsBenoît Thébaudeau2015-09-291-10/+9
| | | | | | |/ | | | | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | SND_SOC_DAIFMT_{IB|NB}_{IF|NF} are defined as inverting or not BCLK or FRM relatively to what is standard for the specified DAI hardware audio format. Consequently, the absolute polarities of these signals cannot be derived only from these settings as this driver did. The format has to be taken into account too. This fixes inverted left/right channels in I²S mode. Signed-off-by: Benoît Thébaudeau <benoit@wsystem.com> Acked-by: Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | | * | | ASoC: dwc: fix dma stop transferring issueyitian2015-10-021-3/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Designware I2S uses tx empty and rx available signals as the DMA handshaking signals. during music playing, if XRUN occurs, i2s_stop() function will be executed and both tx and rx irq are masked, when music continues to be played, i2s_start() is executed but both tx and rx irq are not unmasked which cause I2S stop sending DMA handshaking signal to DMA controller, and it finally causes music playing will be stopped once XRUN occurs for the first time. [On list discussion suggests this may be partly a race condition on slow systems -- broonie] Signed-off-by: Yitian Bu <yitian.bu@tangramtek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | | * | | ASoC: dwc: correct irq clear methodYitian Bu2015-10-021-2/+2
| | | | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | from Designware I2S datasheet, tx/rx XRUN irq is cleared by reading register TOR/ROR, rather than by writing into them. Signed-off-by: Yitian Bu <yitian.bu@tangramtek.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org
| | | * / / ASoC: db1200: Fix DAI link format for db1300 and db1550Lars-Peter Clausen2015-09-251-0/+4
| | | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit b4508d0f95fa ("ASoC: db1200: Use static DAI format setup") switched the db1200 driver over to using static DAI format setup instead of a callback function. But the commit only added the dai_fmt field to one of the three DAI links in the driver. This breaks audio on db1300 and db1550. Add the two missing dai_fmt settings to fix the issue. Fixes: b4508d0f95fa ("ASoC: db1200: Use static DAI format setup") Reported-by: Manuel Lauss <manuel.lauss@gmail.com> Tested-by: Manuel Lauss <manuel.lauss@gmail.com> Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org
| * | | | ALSA: synth: Fix conflicting OSS device registration on AWE32Takashi Iwai2015-10-051-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When OSS emulation is loaded on ISA SB AWE32 chip, we get now kernel warnings like: WARNING: CPU: 0 PID: 2791 at fs/sysfs/dir.c:31 sysfs_warn_dup+0x51/0x80() sysfs: cannot create duplicate filename '/devices/isa/sbawe.0/sound/card0/seq-oss-0-0' It's because both emux synth and opl3 drivers try to register their OSS device object with the same static index number 0. This hasn't been a big problem until the recent rewrite of device management code (that exposes sysfs at the same time), but it's been an obvious bug. This patch works around it just by using a different index number of emux synth object. There can be a more elegant way to fix, but it's enough for now, as this code won't be touched so often, in anyway. Reported-and-tested-by: Michael Shell <list1@michaelshell.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: hda - Disable power_save_node for IDT 92HD73xx chipsTakashi Iwai2015-10-041-1/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The recent widget power saving introduced some unavoidable click noises on old IDT 92HD73xx chips while it still seems working on the compatible new chips. In the bugzilla, we tried lots of tests and workarounds, but they didn't help much. So, let's disable the feature for these specific chips as the least (but safest) fix. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=104981 Cc: <stable@vger.kernel.org> # v4.1+ Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: hda - Apply SPDIF pin ctl to MacBookPro 12,1John Flatness2015-10-031-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The MacBookPro 12,1 has the same setup as the 11 for controlling the status of the optical audio light. Simply apply the existing workaround to the subsystem ID for the 12,1. [sorted the fixup entry by tiwai] Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=105401 Signed-off-by: John Flatness <john@zerocrates.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: hda: Add dock support for ThinkPad T550Laura Abbott2015-10-031-0/+1
| |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Much like all the other Lenovo laptops, add a quirk to make sound work with docking. Reported-and-tested-by: lacknerflo@gmail.com Signed-off-by: Laura Abbott <labbott@fedoraproject.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-tascam: change device probing processingTakashi Sakamoto2015-10-121-46/+32
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently, this driver picks up model name with be32_to_cpu() macro to align characters. This is wrong operation because the result is different depending on CPU endiannness. Additionally, vendor released several versions of firmware for this series. It's not better to assign model-dependent information to device entry according to the version field. This commit fixes these bugs. The name of model is picked up correctly and used to identify model-dependent information. Cc: Stefan Richter <stefanr@s5r6.in-berlin.de> Fixes: c0949b278515 ('ALSA: firewire-tascam: add skeleton for TASCAM FireWire series') Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-tascam: Turn on/off FireWire LEDTakashi Sakamoto2015-10-122-0/+16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | TASCAM FireWire series has some LEDs on its surface. These LEDs can be turned on/off by receiving asynchronous transactions to a certain address. One of the LEDs is labels as 'FireWire'. It's better to light it up when this driver starts to work. Besides, the LED for 'FireWire' is turned off at bus reset. This commit implements this idea. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-tascam: add support for MIDI functionalityTakashi Sakamoto2015-10-124-1/+142
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In former commits, this driver got functionalities to transfer/receive MIDI messages to/from TASCAM FireWire series. This commit adds some ALSA MIDI ports to enable userspace applications to use the functionalities. I note that this commit doesn't support virtual MIDI ports which console models support. A physical controls can be assigned to a certain MIDI ports including physical and virtual. But the way is not clear. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-tascam: add support for outgoing MIDI messages by ↵Takashi Sakamoto2015-10-122-1/+102
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | asynchronous transaction TASCAM FireWire series use asynchronous transaction to receive MIDI messages. The transaction should be sent to a certain address. This commit supports the outgoing MIDI messages. The messages in the transaction includes some quirks: * One MIDI message is transferred in one quadlet transaction, except for system exclusives. * MIDI running status is not allowed, thus transactions always include status byte. * The basic data format is the same as transferring MIDI messages supported in previous commit. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-tascam: add support for incoming MIDI messages by ↵Takashi Sakamoto2015-10-124-1/+213
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | asynchronous transaction TASCAM FireWire series use asynchronous transaction to transfer MIDI messages. The transaction is sent to a registered address. This commit supports the incoming MIDI messages. The messages in the transaction include some quirks: * Two quadlets are used for one MIDI message and one timestamp. * Usually, the first byte of the first quadlet includes MIDI port and MSB 4 bit of MIDI status. For system exclusive message, the first byte includes MIDI port and 0x04, or 0x07 in the end of the message. * The rest of the first quadlet includes MIDI bytes up to 3. * Several set of MIDI messages and timestamp can be transferred in one block transaction, up to 8 sets. I note that TASCAM FireWire series ignores ID bytes of system exclusive message. When receiving system exclusive messages with ID bytes on physical MIDI bus, the series transfers the messages without ID bytes on IEEE 1394 bus, and vice versa. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-digi00x: add support for MIDI ports for physical controlsTakashi Sakamoto2015-10-111-16/+45
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In former commits, asynchronous transactions are supported for physical controls. This commit adds a pair of MIDI ports for them. This driver already adds diferrent number of ALSA MIDI ports for physical MIDI ports, and the number of in/out ports are different. As seeing as 'amidi' program in alsa-utils package, a pair of in/out MIDI ports is expected with the same name. Therefore, this commit adds a pair of new ports to the first. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-digi00x: add support of asynchronous transaction for outgoing ↵Takashi Sakamoto2015-10-112-4/+26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | MIDI messages to physical controls In previous commit, asynchronous transaction for incoming MIDI messages from physical controls is supported. The physical controls may be controlled by receiving MIDI messages at a certain address. This commit supports asynchronous transaction for this purpose. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-digi00x: add support of asynchronous transaction for incoming ↵Takashi Sakamoto2015-10-112-3/+41
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | MIDI messages from physical controls Digi 00x series has two types of model; rack and console. The console models have physical controls. The model can transmit control messages. These control messages are transferred by asynchronous transactions to registered address. This commit supports the asynchronous transaction. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-digi00x: add support for MIDI ports corresponding to ↵Takashi Sakamoto2015-10-115-1/+140
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | isochronous packet streaming This commit adds MIDI functionality to capture/playback MIDI messages from/to physical MIDI ports. These messages are transferred in isochronous packets. When no substreams request AMDTP streams to run, this driver starts the streams at current sampling rate. When other substreams start at different sampling rate, the streams are stopped temporarily, then start again at requested sampling rate. This operation can generate missing MIDI bytes, thus it's preferable to start PCM substreams at favorite sampling rate in advance. Digi 002/003 console also has a set of MIDI port for physical controls. These ports are added in later commits. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-digi00x: handle MIDI messages in isochronous packetsTakashi Sakamoto2015-10-113-8/+125
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In Digi 002/003 protocol, MIDI messages are transferred in the last data channel of data blocks. Although this data channel has a label of 0x80, it's not fully MIDI conformant data channel especially because the Counter field always zero independently of included MIDI bytes. The 4th byte of the data channel in LSB tells the number of included MIDI bytes. This byte also includes the number of MIDI port. Therefore, the data format in this data channel is: * 1st: 0x80 as label * 2nd: MIDI bytes * 3rd: 0 or MIDI bytes * 4th: the number of MIDI byte and the number of MIDI port This commit adds support of MIDI messages in data block processing layer. Like AM824 data format, this data channel has a capability to transfer more MIDI messages than the capability of phisical MIDI bus. Therefore, a throttle for data rate is required to prevent devices' internal buffer to overflow. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-digi00x: use in-kernel representation for the type of 8 bitsTakashi Sakamoto2015-10-111-15/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Original code for 'DoubleOhThree' encoding was written with '__u8' type, while the type is usually used to export something to userspace. This commit replaces the type with 'u8'. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: usb-audio: Allow any MIDI endpoint to drive use of interrupt transfer ↵Keith A. Milner2015-10-111-4/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | on newer Roland devices This patch enables interrupt transfer mode for MIDI ports on newer Boss/Roland devices such as the GT-100/001 which support interrupt transfer on both IN and OUT MIDI endpoints. Previously this wasn't being enabled for these devices as the code was specifically looking for the scenario where the IN endpoint supported interrupt transfer and the OUT endpoint was bulk transfer. Newer devices support interrupt transfer for both endpoints. This has been tested on Boss devices GT-001, BR-80 and JS-8 and Roland VS-20. It would benefit from some regresison testing with other devices if possible. Signed-off-by: Keith A. Milner <maillist@superlative.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-lib: continue packet processing at detecting wrong CIP headersTakashi Sakamoto2015-10-111-4/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In firewire-lib, isochronous packet streaming is stopped when detecting wrong value for FMT field of CIP headers. Although this is appropriate to IEC 61883-1 and 6, some BeBoB based devices with vendors' customization use invalid value to FMT field of CIP headers in the beginning of streaming. $ journalctl snd-bebob fw1.0: Detect unexpected protocol: 01000000 8000ffff I got this log with M-Audio FireWire 1814. In this line, the value of FMT field is 0x00, while it should be 0x10 in usual AMDTP. Except for the beginning, these devices continue to transfer packets with valid value for FMT field, except for the beginning. Therefore, in this case, firewire-lib should continue to process packets. The former implementation of firewire-lib performs it. This commit loosens the handling of wrong value, to continue packet processing in the case. Fixes: 414ba022a528 ('ALSA: firewire-lib: add support arbitrary value for fmt/fdf fields in CIP header') Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: bebob: constify various snd_bebob structuresJulia Lawall2015-10-119-61/+61
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The structures of type snd_bebob_clock_spec, snd_bebob_rate_spec, snd_bebob_meter_spec, and snd_bebob_spec are never modified after they are initialized. Make them all const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal errorTakashi Sakamoto2015-10-092-4/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently, when asynchronous transactions finish in error state and retries, work scheduling and work running also continues. This should be canceled at fatal error because it can cause endless loop. This commit enables to cancel transferring MIDI messages when transactions encounter fatal errors. This is achieved by setting error state. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-lib: add throttle for MIDI data rateTakashi Sakamoto2015-10-092-1/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Typically, the target devices have internal buffer to adjust output of received MIDI messages for MIDI serial bus, while the capacity of the buffer is limited. IEEE 1394 transactions can transfer more MIDI messages than MIDI serial bus can. This can cause buffer over flow in device side. This commit adds throttle to limit MIDI data rate by counting intervals between two MIDI messages. Usual MIDI messages consists of two or three bytes. This requires 1.302 to 1.953 mili-seconds interval between these messages. This commit uses kernel monotonic time service to calculate the time of next transaction. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI ↵Takashi Sakamoto2015-10-091-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | messages Currently, when two MIDI trigger callbacks can be called immediately, transactions for the second MIDI messages can be postpone till next trigger callback. This is not good for real-time message transmission. This commit schedules work again at response handling callback if the MIDI substream still includes untransferred MIDI messages. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-lib: add a restriction for a transaction at onceTakashi Sakamoto2015-10-092-0/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently, when waiting for a response, callers can start another transaction by scheduling another work. This is not good for error processing of transaction, especially the first response is too late. This commit serialize request/response transactions, by adding one boolean member to represent idling state. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: firewire-lib: add helper functions for asynchronous transactions to ↵Takashi Sakamoto2015-10-092-0/+153
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | transfer MIDI messages Some models receive MIDI messages via IEEE 1394 asynchronous transactions. In this case, MIDI messages are transferred in fixed-length payload. It's nice that firewire-lib module has common helper functions. This commit implements this idea. Each driver adds 'struct snd_fw_async_midi_port' in its instance structure. In probing, it should call snd_fw_async_midi_port_init() to initialize the structure with some parameters such as target address, the length of payload in a transaction and a pointer for callback function to fill the payload buffer. At 'struct snd_rawmidi_ops.trigger()' callback, it should call 'snd_fw_async_midi_port_run()' to start transactions. Each driver should ensure that the lifetime of MIDI substream continues till calling 'snd_fw_async_midi_port_finish()'. The helper functions support retries to transferring MIDI messages when transmission errors occur. When transactions are successful, the helper functions call 'snd_rawmidi_transmit_ack()' internally to consume MIDI bytes in the buffer. Therefore, Each driver is expected to use 'snd_rawmidi_transmit_peek()' to tell the number of bytes to transfer to return value of 'fill' callback. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: seq_oss: fix waitqueue_active without memory barrier in snd-seq-ossKosuke Tatsukawa2015-10-092-7/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_seq_oss_readq_put_event() seems to be missing a memory barrier which might cause the waker to not notice the waiter and miss sending a wake_up as in the following figure. snd_seq_oss_readq_put_event snd_seq_oss_readq_wait ------------------------------------------------------------------------ /* wait_event_interruptible_timeout */ /* __wait_event_interruptible_timeout */ /* ___wait_event */ for (;;) { prepare_to_wait_event(&wq, &__wait, state); spin_lock_irqsave(&q->lock, flags); if (waitqueue_active(&q->midi_sleep)) /* The CPU might reorder the test for the waitqueue up here, before prior writes complete */ if ((q->qlen>0 || q->head==q->tail) ... __ret = schedule_timeout(__ret) if (q->qlen >= q->maxlen - 1) { memcpy(&q->q[q->tail], ev, sizeof(*ev)); q->tail = (q->tail + 1) % q->maxlen; q->qlen++; ------------------------------------------------------------------------ There are two other place in sound/core/seq/oss/ which have similar code. The attached patch removes the call to waitqueue_active() leaving just wake_up() behind. This fixes the problem because the call to spin_lock_irqsave() in wake_up() will be an ACQUIRE operation. I found this issue when I was looking through the linux source code for places calling waitqueue_active() before wake_up*(), but without preceding memory barriers, after sending a patch to fix a similar issue in drivers/tty/n_tty.c (Details about the original issue can be found here: https://lkml.org/lkml/2015/9/28/849). Signed-off-by: Kosuke Tatsukawa <tatsu@ab.jp.nec.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: hda: make use of core codec fnsVinod Koul2015-10-083-54/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Now that we have introduced the core fns we should make hda use these helpers Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: hdac: Copy codec helpers to coreSubhransu S. Prusty2015-10-081-0/+81
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The current codec helpers are local to hda code and needs to be moved to core so that other users can use it. The helpers to read/write the codec and to check the power state of widgets is copied Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com> Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: hdac: Fix to check if stream not in use in releaseJeeja KP2015-10-051-4/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | if the stream is decoupled and both link and host are used, while releasing the stream, need to check if link and host stream are not in use. This patch adds fix to check if the host/link stream is in used before coupling it back when releasing the stream. Signed-off-by: Jeeja KP <jeeja.kp@intel.com> Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: hdac: Fix incorrect update of stream id mappingSubhransu S. Prusty2015-10-051-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Bits in LOSIDV need to be set to map the stream id for specific link. Fixing this by setting the required bits in the register. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com> Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>