February 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH V2 0/3] ASOC: bcm2835: move bcm2835-i2s to use clock framework
by kernel＠martin.sperl.org 13 Feb '16

13 Feb '16

From: Martin Sperl <kernel(a)martin.sperl.org> This patchset enables the bcm2835-i2s driver to use the clock framework which was introduced with commit 94cb7f76caa0 ("ARM: bcm2835: Switch to using the new clock driver support."). This commit resulted in the fact that the bcm2835-i2s driver was no longer working due to some register addresses used by 2 drivers (clk-bcm2835 and bcm2835-i2s). This patchset requires that the patchset for PCM-clock support as well as fractional/mash support is applied to the clk-bcm2835 driver and the corresponding device tree, but as the current version of the driver is not working, that should not be a problem. Note that there is one change: right now the current driver tries to calculate an optimal bclk_ratio based on its knowledge that it is using the 19.2Mhz oscillator. This computation would recommend the use of 40 or 80 bits instead of 32/64 bits that are required. Some of the DACs can not handle this, so most downstream drivers would set snd_soc_dai_set_bclk_ratio explicitly to disable this "non-power-of-2" automatic selection. So it seems wise to leave it out of the current patchset. If it is deemed necessary, then I can provide a separate patch that implements this again. Changelog: V1->V2: * moving clock patches into a separate patchset, which fixes also other issues in the clock framework * remove unnecessary bclk_size assignments Martin Sperl (3): ASoC: bcm2835: move to use the clock framework ARM: bcm2835: I2S: use new register-range and clock framework dt-bindings: bsm2835: fix bindings documentation to use new clock framework .../devicetree/bindings/sound/brcm,bcm2835-i2s.txt | 7 +- arch/arm/boot/dts/bcm2835.dtsi | 5 +- sound/soc/bcm/bcm2835-i2s.c | 284 +++++--------------- 3 files changed, 69 insertions(+), 227 deletions(-) -- 1.7.10.4

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[alsa-devel] [PATCH] midi: avoid freeing umidi object twice
by Andrey Konovalov 13 Feb '16

13 Feb '16

The 'umidi' object will be free'd on the error path by snd_usbmidi_free() when tearing down the rawmidi interface. So we shouldn't try to free it in snd_usbmidi_create() after having registered the rawmidi interface. Found by KASAN. Acked-by: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Andrey Konovalov <andreyknvl(a)gmail.com> --- sound/usb/midi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index cc39f63..007cf58 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2455,7 +2455,6 @@ int snd_usbmidi_create(struct snd_card *card, else err = snd_usbmidi_create_endpoints(umidi, endpoints); if (err < 0) { - snd_usbmidi_free(umidi); return err; } -- 1.9.1

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[alsa-devel] Applied "ASoC: qcom: Don't specify LE device endianness" to the asoc tree
by Mark Brown 12 Feb '16

12 Feb '16

The patch ASoC: qcom: Don't specify LE device endianness has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8faa268ab780a7379cfe54b882c6f6e0083233f2 Mon Sep 17 00:00:00 2001 From: Stephen Boyd <sboyd(a)codeaurora.org> Date: Fri, 12 Feb 2016 09:30:17 -0800 Subject: [PATCH] ASoC: qcom: Don't specify LE device endianness This reverts commit 18560a4e3 (ASoC: qcom: Specify LE device endianness). The commit that caused us to specify LE device endianness here, 29bb45f25ff3 (regmap-mmio: Use native endianness for read/write, 2015-10-29), has been reverted in mainline so now when we specify LE it actively breaks big endian kernels because the byte swapping in regmap-mmio is incorrect. Let's revert this change because it will 1) fix the big endian kernels and 2) be redundant to specify LE because that will become the default soon. Signed-off-by: Stephen Boyd <sboyd(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/lpass-cpu.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 00b6c9d039cf..e5101e0d2d37 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -355,7 +355,6 @@ static struct regmap_config lpass_cpu_regmap_config = { .readable_reg = lpass_cpu_regmap_readable, .volatile_reg = lpass_cpu_regmap_volatile, .cache_type = REGCACHE_FLAT, - .val_format_endian = REGMAP_ENDIAN_LITTLE, }; int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) -- 2.7.0

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[alsa-devel] [PATCH] ALSA: at73c213: add error handling of clk_enable()
by Alexey Khoroshilov 12 Feb '16

12 Feb '16

Recent addition of SSC clock management does not check if clk_enable() succeed. The patch adds such checks where it is appropriate. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> --- sound/spi/at73c213.c | 25 ++++++++++++++++++++----- 1 file changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index fac7e6eb9529..36cfaa316eab 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -221,7 +221,11 @@ static int snd_at73c213_pcm_open(struct snd_pcm_substream *substream) runtime->hw = snd_at73c213_playback_hw; chip->substream = substream; - clk_enable(chip->ssc->clk); + err = clk_enable(chip->ssc->clk); + if (err < 0) { + chip->substream = NULL; + return err; + } return 0; } @@ -787,7 +791,9 @@ static int snd_at73c213_chip_init(struct snd_at73c213 *chip) goto out; /* Enable DAC master clock. */ - clk_enable(chip->board->dac_clk); + retval = clk_enable(chip->board->dac_clk); + if (retval) + goto out; /* Initialize at73c213 on SPI bus. */ retval = snd_at73c213_write_reg(chip, DAC_RST, 0x04); @@ -900,7 +906,9 @@ static int snd_at73c213_dev_init(struct snd_card *card, chip->card = card; chip->irq = -1; - clk_enable(chip->ssc->clk); + retval = clk_enable(chip->ssc->clk); + if (retval) + return retval; retval = request_irq(irq, snd_at73c213_interrupt, 0, "at73c213", chip); if (retval) { @@ -1099,9 +1107,16 @@ static int snd_at73c213_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; + int retval; - clk_enable(chip->board->dac_clk); - clk_enable(chip->ssc->clk); + retval = clk_enable(chip->board->dac_clk); + if (retval) + return retval; + retval = clk_enable(chip->ssc->clk); + if (retval) { + clk_disable(chip->board->dac_clk); + return retval; + } ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); return 0; -- 1.9.1

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[alsa-devel] [PATCH 1/2] ALSA: timer: Protect the whole snd_timer_close() with open race
by Takashi Iwai 12 Feb '16

12 Feb '16

In order to make the open/close more robust, widen the register_mutex protection over the whole snd_timer_close() function. Also, the close procedure is slightly shuffled to be in the safer order, as well as a few code refactoring. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is no urgent fix, so queued for 4.6. sound/core/timer.c | 48 +++++++++++++++++++++--------------------------- 1 file changed, 21 insertions(+), 27 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index dca817fc7894..b572a9bc31ad 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -318,25 +318,14 @@ int snd_timer_close(struct snd_timer_instance *timeri) if (snd_BUG_ON(!timeri)) return -ENXIO; + mutex_lock(&register_mutex); + list_del(&timeri->open_list); + /* force to stop the timer */ snd_timer_stop(timeri); - if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) { - /* wait, until the active callback is finished */ - spin_lock_irq(&slave_active_lock); - while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) { - spin_unlock_irq(&slave_active_lock); - udelay(10); - spin_lock_irq(&slave_active_lock); - } - spin_unlock_irq(&slave_active_lock); - mutex_lock(&register_mutex); - list_del(&timeri->open_list); - mutex_unlock(&register_mutex); - } else { - timer = timeri->timer; - if (snd_BUG_ON(!timer)) - goto out; + timer = timeri->timer; + if (timer) { /* wait, until the active callback is finished */ spin_lock_irq(&timer->lock); while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) { @@ -345,11 +334,7 @@ int snd_timer_close(struct snd_timer_instance *timeri) spin_lock_irq(&timer->lock); } spin_unlock_irq(&timer->lock); - mutex_lock(&register_mutex); - list_del(&timeri->open_list); - if (list_empty(&timer->open_list_head) && - timer->hw.close) - timer->hw.close(timer); + /* remove slave links */ spin_lock_irq(&slave_active_lock); spin_lock(&timer->lock); @@ -363,18 +348,27 @@ int snd_timer_close(struct snd_timer_instance *timeri) } spin_unlock(&timer->lock); spin_unlock_irq(&slave_active_lock); - /* release a card refcount for safe disconnection */ - if (timer->card) - put_device(&timer->card->card_dev); - mutex_unlock(&register_mutex); + + /* slave doesn't need to release timer resources below */ + if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) + timer = NULL; } - out: + if (timeri->private_free) timeri->private_free(timeri); kfree(timeri->owner); kfree(timeri); - if (timer) + + if (timer) { + if (list_empty(&timer->open_list_head) && timer->hw.close) + timer->hw.close(timer); + /* release a card refcount for safe disconnection */ + if (timer->card) + put_device(&timer->card->card_dev); module_put(timer->module); + } + + mutex_unlock(&register_mutex); return 0; } -- 2.7.1

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[alsa-devel] Usb audio - unsuported sampling rates
by Gordon Dilan 12 Feb '16

12 Feb '16

Hello, I'm using an usb mic which(according to it's producer should support up to 48kHz sampling rate). The problem is ALSA allows me to use only 16 kHz for this device. I do not want to resample the data, I want to sample it at 48kHz in order to be able to record some high pitch sounds 18-20kHz. I would want to try to do any modifications necessary in order to try support 48kHz, but I'm not sure where to start. Some information: cat /proc/asound/modules ... 1 snd_usb_audio arecord --list-devices **** List of CAPTURE Hardware Devices **** .... card 1: CameraB409241 [USB Camera-B4.09.24.1], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 when I try to use: arecord -v -D hw:1,0 -f S16_LE -c 4 -r 48000 gigi.wav I get this: Recording WAVE 'gigi.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Channels 4 Warning: rate is not accurate (requested = 48000Hz, got = 16000Hz) please, try the plug plugin Hardware PCM card 1 'USB Camera-B4.09.24.1' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 4 rate : 16000 exact rate : 16000 (16000/1) msbits : 16 buffer_size : 8000 period_size : 2000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 2000 period_event : 0 start_threshold : 1 stop_threshold : 8000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 appl_ptr : 0 hw_ptr : 0 Any sampling rate I would try higher or lower than 16kHz, I will always get 16000 kHz. Some more information: lsusb Bus 001 Device 007: ID 1415:2000 Nam Tai E&E Products Ltd. or OmniVision Technologies, Inc. Sony Playstation Eye lsusb -t /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=ehci_hcd/2p, 480M |__ Port 1: Dev 2, If 0, Class=hub, Driver=hub/6p, 480M |__ Port 2: Dev 7, If 0, Class=vend., Driver=ov534, 480M |__ Port 2: Dev 7, If 1, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 7, If 2, Class=audio, Driver=snd-usb-audio, 480M At the moment I'm looking inside ALSA's source code to try to see from where this limitation of 16000kHz for my device appears, but I have to admit that I'm kind of lost. Any pointers on what direction should I go to try make my desired rate supported would be greatly appreciated! Thank you and best regards, Gordon

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[alsa-devel] [PATCH 0/2] ALSA: dice: improve Dice notification handling
by Takashi Sakamoto 12 Feb '16

12 Feb '16

Hi, This patchset improves Dice notification handling mainly for old firmwares. Dice-based models with earlier firmwares brings 'fail to ensure phase lock' message to ALSA dice driver. This is due to no Dice transaction at confirming clock status. Although this patchet enables ALSA dice driver to handle Dice-based models with old firmwares, I reccomend users to work with updated firmwares because they may include some bug fixes and improve device's behaviours. Regards Takashi Sakamoto (2): ALSA: dice: change notification mask to detect lock status change ALSA: dice: old firmware optimization for Dice notification sound/firewire/dice/dice-stream.c | 17 ++++++++++++++--- sound/firewire/dice/dice-transaction.c | 2 +- 2 files changed, 15 insertions(+), 4 deletions(-) -- 2.5.0

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[alsa-devel] ASoC fixes for v4.5
by Mark Brown 12 Feb '16

12 Feb '16

The following changes since commit 388f7b1d6e8ca06762e2454d28d6c3c55ad0fe95: Linux 4.5-rc3 (2016-02-07 15:38:30 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.5-rc4 for you to fetch changes up to f11aec0d7c835c3b83e74e3fd60d2e48a850c857: Merge remote-tracking branches 'asoc/fix/rt5659', 'asoc/fix/sigmadsp', 'asoc/fix/simple', 'asoc/fix/wm5110' and 'asoc/fix/wm8960' into asoc-linus (2016-02-10 19:23:09 +0000) ---------------------------------------------------------------- ASoC: Fixes for v4.5 A rather large batch of fixes here, almost all in the Intel driver. The changes that got merged in this merge window for Skylake were rather large and as well as issues that you'd expect in a large block of new code there were some problems created for older processors which needed fixing up. Things are largely settling down now hopefully. ---------------------------------------------------------------- Aaro Koskinen (1): ASoC: simple-card: don't fail if sysclk setting is not supported Arnd Bergmann (2): ASoC: mediatek: add i2c dependency ASoC: rt5659: avoid unused variable warning for rt5659_acpi_match Axel Lin (1): ASoC: rt5659: Fix irq leak Bard Liao (2): ASoC: rt286: fix capture doesn't work at some cases ASoC: rt5645: fix the shift bit of IN1 boost Dan Carpenter (1): ASoC: AMD: free memory on error Dharageswari.R (1): ASoC: Intel: Skylake: Fix pipe memory allocation leak Guneshwor Singh (2): ASoC: Intel: Skylake: Fix delay wrap condition ASoC: Intel: Skylake: Fix mcps freeup after module unbind failure Jeeja KP (6): ASoC: Intel: Skylake: Fix return of skl_get_queue_index ASoC: Intel: Skylake: Fix the module state check condition ASoC: Intel: Skylake: Fix not to stop sink pipe in pga pmd event ASoC: Intel: Skylake: Fix bind of source with multiple sinks ASoC: Intel: Skylake: Add missing PRE/POST_PMU handlers for vmixer ASoC: Intel: Skylake: Fix stereo DMIC record Jon Medhurst (Tixy) (1): ASoC: dwc: Ensure i2s_reg_comp{1,2} is always initialised Lars-Peter Clausen (2): ASoC: imx-spdif: Fix crash on suspend ASoC: dapm: Don't prefix autodisable widgets twice Maciej S. Szmigiero (1): ASoC: fsl_ssi: remove explicit register defaults Mans Rullgard (1): ASoC: mxs-saif: fix clk_prepare() without matching clk_unprepare() Mark Brown (7): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel-kconfig' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/amd', 'asoc/fix/arizona', 'asoc/fix/dpcm', 'asoc/fix/dwc', 'asoc/fix/fsl' and 'asoc/fix/fsl-ssi' into asoc-linus Merge remote-tracking branches 'asoc/fix/imx-spdif', 'asoc/fix/mtk', 'asoc/fix/mxs-saif', 'asoc/fix/qcom' and 'asoc/fix/rt286' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5659', 'asoc/fix/sigmadsp', 'asoc/fix/simple', 'asoc/fix/wm5110' and 'asoc/fix/wm8960' into asoc-linus Pascal Huerst (1): ASoC: sigmadsp: Fix missleading return value Pierre-Louis Bossart (2): ASoC: Intel: Atom: fix regression on compress DAI ASoC: Intel: Load the atom DPCM driver only Richard Fitzgerald (2): ASoC: wm5110: Unregister compressed platform when driver is removed ASoC: arizona: fref must be limited in pseudo-fractional mode Srinivas Kandagatla (1): ASoC: qcom: use correct device pointer in dma allocation Stuart Henderson (2): ASoC: wm8960: Fix input boost mixer left/right naming ASoC: wm8960: Fix WM8960_SYSCLK_PLL mode Sudip Mukherjee (1): ASoC: Intel: Skylake: Fix memory leak Vinod Koul (6): ASoC: dpcm: fix the BE state on hw_free ASoC: Intel: Skylake: Fix the memory overwrite of tlv buffer ASoC: Intel: Skylake: Remove autosuspend delay ASoC: Intel: Revert "ASoC: Intel: fix ACPI probe regression with Atom DPCM driver" ASoC: Intel: Create independent acpi match module ASoC: Intel: Add module tags for common match module Xiubo Li (1): ASoC: fsl: document DT compatible string "fsl,imx-audio-wm8960" .../devicetree/bindings/sound/fsl-asoc-card.txt | 2 + sound/soc/amd/acp-pcm-dma.c | 1 + sound/soc/codecs/arizona.c | 43 ++++++++++++- sound/soc/codecs/rt286.c | 26 +------- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5659.c | 31 ++++----- sound/soc/codecs/rt5659.h | 1 - sound/soc/codecs/sigmadsp-i2c.c | 5 +- sound/soc/codecs/wm5110.c | 1 + sound/soc/codecs/wm8960.c | 40 ++++++------ sound/soc/dwc/designware_i2s.c | 5 +- sound/soc/fsl/fsl_ssi.c | 42 ++++++------ sound/soc/fsl/imx-spdif.c | 2 - sound/soc/generic/simple-card.c | 2 +- sound/soc/intel/Kconfig | 13 +++- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 + sound/soc/intel/boards/skl_rt286.c | 5 +- sound/soc/intel/common/Makefile | 9 +-- sound/soc/intel/common/sst-acpi.c | 4 ++ sound/soc/intel/common/sst-match-acpi.c | 3 + sound/soc/intel/skylake/skl-messages.c | 6 +- sound/soc/intel/skylake/skl-pcm.c | 1 + sound/soc/intel/skylake/skl-topology.c | 75 +++++++++++++++------- sound/soc/intel/skylake/skl.c | 2 - sound/soc/mediatek/Kconfig | 4 +- sound/soc/mxs/mxs-saif.c | 13 +++- sound/soc/qcom/lpass-platform.c | 15 ++--- sound/soc/soc-dapm.c | 8 +-- sound/soc/soc-pcm.c | 3 +- 29 files changed, 221 insertions(+), 144 deletions(-)

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[alsa-devel] Linking ALSA sequencer kernel clients to their device
by Martin Koegler 12 Feb '16

12 Feb '16

In my domain, people use multiple identical USB-MIDI keyboards - each is assigned a different functions. They expect to kept this mapping across reboots. The sound card number are dynamic, therefore they can change after reboots. The only stable device ID is the sysfs path of the used USB port. This means, that I need to associate a kernel sequencer client number with a sysfs device. rawmidi devices expose the card number via IOCTLs, which allows to find the corresponding device in sysfs. The sequencer provides no identifing data. Chromium works around this issue by scanning rawmidi as well as sequencer devices and matching them by using assumtions, how the kernel register sequencer devices. So I thought about exposing the missing information via sysfs: >From 776dd02d8d7eae89c2699be27eedf81eaf690a0e Mon Sep 17 00:00:00 2001 From: Martin Koegler <martin.koegler(a)chello.at> Date: Mon, 8 Feb 2016 22:53:42 +0100 Subject: [PATCH] Provide sequencer client number via sysfs Provides the missing link between real hardware devices and kernel sequencer clients. Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- sound/core/seq/seq_midi.c | 38 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 5dd0ee2..0d807e2 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -271,6 +271,31 @@ static void snd_seq_midisynth_delete(struct seq_midisynth *msynth) snd_midi_event_free(msynth->parser); } +static ssize_t +client_number_show_attr(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_seq_device *device = container_of(dev, struct snd_seq_device, dev); + struct snd_card *card = device->card; + struct seq_midisynth_client *client; + + client = card ? synths[card->number] : NULL; + return snprintf(buf, PAGE_SIZE, "%i\n", client ? client->seq_client : -1); +} + +static DEVICE_ATTR(client, S_IRUGO, client_number_show_attr, NULL); + +static ssize_t +device_number_show_attr(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_seq_device *device = container_of(dev, struct snd_seq_device, dev); + + return snprintf(buf, PAGE_SIZE, "%i\n", device->device); +} + +static DEVICE_ATTR(device, S_IRUGO, device_number_show_attr, NULL); + /* register new midi synth port */ static int snd_seq_midisynth_probe(struct device *_dev) @@ -287,6 +312,7 @@ snd_seq_midisynth_probe(struct device *_dev) struct snd_card *card = dev->card; int device = dev->device; unsigned int input_count = 0, output_count = 0; + int error; if (snd_BUG_ON(!card || device < 0 || device >= SNDRV_RAWMIDI_DEVICES)) return -EINVAL; @@ -332,6 +358,16 @@ snd_seq_midisynth_probe(struct device *_dev) kfree(info); return -ENOMEM; } + error = device_create_file(&dev->dev, &dev_attr_client); + if (error >= 0) + error = device_create_file(&dev->dev, &dev_attr_device); + if (error < 0) { + snd_seq_delete_kernel_client(client->seq_client); + kfree(client); + mutex_unlock(&register_mutex); + kfree(info); + return error; + } } msynth = kcalloc(ports, sizeof(struct seq_midisynth), GFP_KERNEL); @@ -451,6 +487,8 @@ snd_seq_midisynth_remove(struct device *_dev) kfree(msynth); client->num_ports--; if (client->num_ports <= 0) { + device_remove_file(&dev->dev, &dev_attr_client); + device_remove_file(&dev->dev, &dev_attr_device); snd_seq_delete_kernel_client(client->seq_client); synths[card->number] = NULL; kfree(client); -- 2.1.4

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[alsa-devel] Resend [GIT PULL]: Update sst audio firmware for Skylake
by Vinod Koul 11 Feb '16

11 Feb '16

Resending again as previous email bounced for linux-firmware(a)kernel.org... Hello Linux firmware maintainers, Please PULL to receive the updated version of Skylake audio firmware. This PULL request has two patches, first one creates a sym link for binary name which helps us to keep different versions of firmware in git Second one updates to latest FW v869 The following changes since commit 52442afee9907bc32a058f22bb3295d040677c26: brcm: add firmware for bcm4366 device (2016-01-25 15:15:34 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to f7996657e8f25386541593242cb8e906d9808ec7: linux-firmware: intel: Update Skylake audio firmware (2016-02-04 10:36:52 +0530) ---------------------------------------------------------------- Vinod Koul (2): linux-firmware: intel: add symbolic link and file versions linux-firmware: intel: Update Skylake audio firmware WHENCE | 4 ++-- intel/dsp_fw_release.bin | Bin 233472 -> 23 bytes intel/dsp_fw_release_v827.bin | Bin 0 -> 233472 bytes intel/dsp_fw_release_v869.bin | Bin 0 -> 233472 bytes 4 files changed, 2 insertions(+), 2 deletions(-) mode change 100644 => 120000 intel/dsp_fw_release.bin create mode 100644 intel/dsp_fw_release_v827.bin create mode 100644 intel/dsp_fw_release_v869.bin Thanks -- ~Vinod

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