August 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC][PATCH 0/3] ALSA: control: delegate TLV eventing to each driver
by Takashi Sakamoto 24 Aug '17

24 Aug '17

Hi, This patchset is to avoid a trouble on ALSA control core, kernel API of ALSA control component and ALSA control interface/protocol to handle TLV operations from user space. In a design of ALSA control core, a set of elements is represented by 'struct snd_kcontrol' to share common attributes. The set of elements shares TLV (Type-Length-Value) data, too. On the other hand, in ALSA control interface/protocol for applications, a TLV operation is committed to an element. Totally, the operation can have sub-effect to the other elements in the set. For example, TLV_WRITE operation is expected to change TLV data, which returns to applications. Applications attempt to change the TLV data per element, but in the above design, they can effect to elements in the same set. As a default, ALSA control core has no implementation except for TLV_READ operation. Thus, the above design looks to have no issue. However, in kernel APIs of ALSA control component, developers can program a handler for any request of the TLV operation. Therefore, for elements in a set which has the handler, applications can commit TLV_WRITE and TLV_COMMAND requests. For the above scenario, ALSA control core assist notification. When the handler returns positive value, the core queueing an event for a requested element. However, this includes design defects that the event is not queued for the other element in a set. Actually, developers can program the handlers to keep per-element TLV data, but it depends on each driver. In this patchset, I attempts to solve the above issue, by delegating the event notification to each driver of ALSA control component. Additionally, this patchset includes a solution against an issue in below message thread: [alsa-devel] snd_ctl_add_enumerated_elem_set and TLV http://mailman.alsa-project.org/pipermail/alsa-devel/2017-July/123364.html Takashi Sakamoto (3): ALSA: control: delegate TLV eventing to each driver ALSA: control: queue TLV event for all of elements in an user-defined set ALSA: control: disable TLV data at initial state of user-defined element set sound/core/control.c | 34 ++++++++++++++++++++++------------ 1 file changed, 22 insertions(+), 12 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH] ASoC: soc-ac97: replace codec to component
by Kuninori Morimoto 24 Aug '17

24 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> codec will be replaced to component soon. AC97 is codec related feature now. Thus this patch replaces codec to component on AC97. To keep compatibility, previous snd_soc_xxx_ac97_codec() is now defined as macro to convert snd_soc_xxx_ac97_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- - I have no AC97 board. Compile test only had done. include/sound/soc.h | 10 +++++-- sound/soc/soc-ac97.c | 84 +++++++++++++++++++++++++++------------------------- 2 files changed, 51 insertions(+), 43 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 180c41b..f8fa9d0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -568,10 +568,14 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); #ifdef CONFIG_SND_SOC_AC97_BUS -struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec); -struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec, +#define snd_soc_alloc_ac97_codec(codec) snd_soc_alloc_ac97_component(&codec->component) +#define snd_soc_new_ac97_codec(codec, id, id_mask) snd_soc_new_ac97_component(&codec->component, id, id_mask) +#define snd_soc_free_ac97_codec(ac97) snd_soc_free_ac97_component(ac97) + +struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component); +struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component, unsigned int id, unsigned int id_mask); -void snd_soc_free_ac97_codec(struct snd_ac97 *ac97); +void snd_soc_free_ac97_component(struct snd_ac97 *ac97); int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 36dae41..3f424f2 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -44,7 +44,7 @@ struct snd_ac97_gpio_priv { struct gpio_chip gpio_chip; #endif unsigned int gpios_set; - struct snd_soc_codec *codec; + struct snd_soc_component *component; }; static struct snd_ac97_bus soc_ac97_bus = { @@ -57,11 +57,11 @@ static void soc_ac97_device_release(struct device *dev) } #ifdef CONFIG_GPIOLIB -static inline struct snd_soc_codec *gpio_to_codec(struct gpio_chip *chip) +static inline struct snd_soc_component *gpio_to_component(struct gpio_chip *chip) { struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); - return gpio_priv->codec; + return gpio_priv->component; } static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned offset) @@ -75,20 +75,22 @@ static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned offset) static int snd_soc_ac97_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct snd_soc_codec *codec = gpio_to_codec(chip); + struct snd_soc_component *component = gpio_to_component(chip); - dev_dbg(codec->dev, "set gpio %d to output\n", offset); - return snd_soc_update_bits(codec, AC97_GPIO_CFG, + dev_dbg(component->dev, "set gpio %d to output\n", offset); + return snd_soc_component_update_bits(component, AC97_GPIO_CFG, 1 << offset, 1 << offset); } static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct snd_soc_codec *codec = gpio_to_codec(chip); + struct snd_soc_component *component = gpio_to_component(chip); int ret; - ret = snd_soc_read(codec, AC97_GPIO_STATUS); - dev_dbg(codec->dev, "get gpio %d : %d\n", offset, + if (snd_soc_component_read(component, AC97_GPIO_STATUS, &ret) < 0) + ret = -1; + + dev_dbg(component->dev, "get gpio %d : %d\n", offset, ret < 0 ? ret : ret & (1 << offset)); return ret < 0 ? ret : !!(ret & (1 << offset)); @@ -98,22 +100,24 @@ static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); - struct snd_soc_codec *codec = gpio_to_codec(chip); + struct snd_soc_component *component = gpio_to_component(chip); gpio_priv->gpios_set &= ~(1 << offset); gpio_priv->gpios_set |= (!!value) << offset; - snd_soc_write(codec, AC97_GPIO_STATUS, gpio_priv->gpios_set); - dev_dbg(codec->dev, "set gpio %d to %d\n", offset, !!value); + snd_soc_component_write(component, AC97_GPIO_STATUS, + gpio_priv->gpios_set); + dev_dbg(component->dev, "set gpio %d to %d\n", offset, !!value); } static int snd_soc_ac97_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct snd_soc_codec *codec = gpio_to_codec(chip); + struct snd_soc_component *component = gpio_to_component(chip); - dev_dbg(codec->dev, "set gpio %d to output\n", offset); + dev_dbg(component->dev, "set gpio %d to output\n", offset); snd_soc_ac97_gpio_set(chip, offset, value); - return snd_soc_update_bits(codec, AC97_GPIO_CFG, 1 << offset, 0); + return snd_soc_component_update_bits(component, AC97_GPIO_CFG, + 1 << offset, 0); } static const struct gpio_chip snd_soc_ac97_gpio_chip = { @@ -128,24 +132,24 @@ static int snd_soc_ac97_gpio_direction_out(struct gpio_chip *chip, }; static int snd_soc_ac97_init_gpio(struct snd_ac97 *ac97, - struct snd_soc_codec *codec) + struct snd_soc_component *component) { struct snd_ac97_gpio_priv *gpio_priv; int ret; - gpio_priv = devm_kzalloc(codec->dev, sizeof(*gpio_priv), GFP_KERNEL); + gpio_priv = devm_kzalloc(component->dev, sizeof(*gpio_priv), GFP_KERNEL); if (!gpio_priv) return -ENOMEM; ac97->gpio_priv = gpio_priv; - gpio_priv->codec = codec; + gpio_priv->component = component; gpio_priv->gpio_chip = snd_soc_ac97_gpio_chip; gpio_priv->gpio_chip.ngpio = AC97_NUM_GPIOS; - gpio_priv->gpio_chip.parent = codec->dev; + gpio_priv->gpio_chip.parent = component->dev; gpio_priv->gpio_chip.base = -1; ret = gpiochip_add_data(&gpio_priv->gpio_chip, gpio_priv); if (ret != 0) - dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); + dev_err(component->dev, "Failed to add GPIOs: %d\n", ret); return ret; } @@ -155,7 +159,7 @@ static void snd_soc_ac97_free_gpio(struct snd_ac97 *ac97) } #else static int snd_soc_ac97_init_gpio(struct snd_ac97 *ac97, - struct snd_soc_codec *codec) + struct snd_soc_component *component) { return 0; } @@ -166,8 +170,8 @@ static void snd_soc_ac97_free_gpio(struct snd_ac97 *ac97) #endif /** - * snd_soc_alloc_ac97_codec() - Allocate new a AC'97 device - * @codec: The CODEC for which to create the AC'97 device + * snd_soc_alloc_ac97_component() - Allocate new a AC'97 device + * @component: The COMPONENT for which to create the AC'97 device * * Allocated a new snd_ac97 device and intializes it, but does not yet register * it. The caller is responsible to either call device_add(&ac97->dev) to @@ -175,7 +179,7 @@ static void snd_soc_ac97_free_gpio(struct snd_ac97 *ac97) * * Returns: A snd_ac97 device or a PTR_ERR in case of an error. */ -struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec) +struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component) { struct snd_ac97 *ac97; @@ -187,26 +191,26 @@ struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec) ac97->num = 0; ac97->dev.bus = &ac97_bus_type; - ac97->dev.parent = codec->component.card->dev; + ac97->dev.parent = component->card->dev; ac97->dev.release = soc_ac97_device_release; dev_set_name(&ac97->dev, "%d-%d:%s", - codec->component.card->snd_card->number, 0, - codec->component.name); + component->card->snd_card->number, 0, + component->name); device_initialize(&ac97->dev); return ac97; } -EXPORT_SYMBOL(snd_soc_alloc_ac97_codec); +EXPORT_SYMBOL(snd_soc_alloc_ac97_component); /** - * snd_soc_new_ac97_codec - initailise AC97 device - * @codec: audio codec + * snd_soc_new_ac97_component - initailise AC97 device + * @component: audio component * @id: The expected device ID * @id_mask: Mask that is applied to the device ID before comparing with @id * - * Initialises AC97 codec resources for use by ad-hoc devices only. + * Initialises AC97 component resources for use by ad-hoc devices only. * * If @id is not 0 this function will reset the device, then read the ID from * the device and check if it matches the expected ID. If it doesn't match an @@ -214,20 +218,20 @@ struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec) * * Returns: A PTR_ERR() on failure or a valid snd_ac97 struct on success. */ -struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec, +struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component, unsigned int id, unsigned int id_mask) { struct snd_ac97 *ac97; int ret; - ac97 = snd_soc_alloc_ac97_codec(codec); + ac97 = snd_soc_alloc_ac97_component(component); if (IS_ERR(ac97)) return ac97; if (id) { ret = snd_ac97_reset(ac97, false, id, id_mask); if (ret < 0) { - dev_err(codec->dev, "Failed to reset AC97 device: %d\n", + dev_err(component->dev, "Failed to reset AC97 device: %d\n", ret); goto err_put_device; } @@ -237,7 +241,7 @@ struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec, if (ret) goto err_put_device; - ret = snd_soc_ac97_init_gpio(ac97, codec); + ret = snd_soc_ac97_init_gpio(ac97, component); if (ret) goto err_put_device; @@ -247,22 +251,22 @@ struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec, put_device(&ac97->dev); return ERR_PTR(ret); } -EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); +EXPORT_SYMBOL_GPL(snd_soc_new_ac97_component); /** - * snd_soc_free_ac97_codec - free AC97 codec device + * snd_soc_free_ac97_component - free AC97 component device * @ac97: snd_ac97 device to be freed * - * Frees AC97 codec device resources. + * Frees AC97 component device resources. */ -void snd_soc_free_ac97_codec(struct snd_ac97 *ac97) +void snd_soc_free_ac97_component(struct snd_ac97 *ac97) { snd_soc_ac97_free_gpio(ac97); device_del(&ac97->dev); ac97->bus = NULL; put_device(&ac97->dev); } -EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); +EXPORT_SYMBOL_GPL(snd_soc_free_ac97_component); static struct snd_ac97_reset_cfg snd_ac97_rst_cfg; -- 1.9.1

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[alsa-devel] [PATCH 00/11] ASoC: make snd_soc_platform_driver const
by Bhumika Goyal 23 Aug '17

23 Aug '17

Make snd_soc_platform_driver const. Bhumika Goyal (11): ASoC: codecs: make snd_soc_platform_driver const ASoC: Intel: make snd_soc_platform_driver const ASoC: fsl: make snd_soc_platform_driver const ASoC: nuc900: make snd_soc_platform_driver const ASoC: omap: make snd_soc_platform_driver const ASoC: pxa: make snd_soc_platform_driver const ASoC: qcom: make snd_soc_platform_driver const ASoC: samsung: make snd_soc_platform_driver const ASoC: sh: make snd_soc_platform_driver const ASoC: txx9: make snd_soc_platform_driver const ASoC: soc-utils: make snd_soc_platform_driver const sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/fsl/imx-pcm-fiq.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 2 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 2 +- sound/soc/nuc900/nuc900-pcm.c | 2 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/samsung/idma.c | 2 +- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-utils.c | 2 +- sound/soc/txx9/txx9aclc.c | 2 +- 21 files changed, 21 insertions(+), 21 deletions(-) -- 1.9.1

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[alsa-devel] [ANNOUNCE RESENT] Audio Mini Summit 2017 at Prague, Oct 27
by Takashi Iwai 23 Aug '17

23 Aug '17

Hi, this is a gentle reminder for people who missed the previous announcement for Audio Mini Summit. Below is the content I posted in a month ago. If anyone is interested, please join up! thanks, Takashi === Hi, this is an announcement of Audio Mini Summit 2017. It's an annual meeting we regularly have along with major Linux conferences, and this year, we'll hold the meeting at Prague (Czech Republic) on Friday, October 27, right after GStreamer Summit, ELCE and Open Source Summit Europe. We planned a whole-day meeting, and will likely have an evening dinner event, too. Please note that the meeting will be held in a conference room of SUSE Prague office, not in Hilton hotel where ELCE/OSSE is held. It's located near Hilton, just a couple of blocks away. The address of SUSE office is: Corso II, Křižíkova 148/34, 18600 Praha 8 https://goo.gl/maps/anET1yh3XXu Also, since the conference room is hosted by SUSE, it's an open event at this time; that is, you can attend only this meeting without registration for other Linux Foundation events in that week, too. If you are interested in the event, please fill up the registration form in Google doc spreadsheet below: https://docs.google.com/spreadsheets/d/1a-RaYtTVVtgh_cyRN5GwATiixppGCMEwqw6… Just add yourself in the attendee list from the web page. And, now we are gathering the topics to discuss, too. If you have any interesting topics to be discussed in the meeting, feel free to post to alsa-devel ML. I'm going to write up later. thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: aaci: constify amba_id
by Arvind Yadav 23 Aug '17

23 Aug '17

amba_id are not supposed to change at runtime. All functions working with const amba_id. So mark the non-const structs as const. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/arm/aaci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 4140b1b..05562ba 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -1085,7 +1085,7 @@ static int aaci_remove(struct amba_device *dev) return 0; } -static struct amba_id aaci_ids[] = { +static const struct amba_id aaci_ids[] = { { .id = 0x00041041, .mask = 0x000fffff, -- 2.7.4

1 0

[alsa-devel] [GIT PULL]: intel audio firmware updates
by Vinod Koul 23 Aug '17

23 Aug '17

Hi, Please pull to receive updated KBL audio firmware. Along with this time for some cleaning of the exiting files and remove no longer used SKL and BXT fw files. The following changes since commit 7d2c913dcd1be083350d97a8cb1eba24cfacbc8a: ath10k: update year in license (2017-06-22 12:06:02 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git intel for you to fetch changes up to 7b7513e03eb6949f39fba2eb4b661c31f8ea6f5b: linux-firmware: intel: remove old broxton firmware (2017-08-21 10:35:32 +0530) ---------------------------------------------------------------- Subhransu S. Prusty (1): linux-firmware: intel: Update Kabylake audio firmware Vinod Koul (2): linux-firmware: intel: remove old skylake firmware linux-firmware: intel: remove old broxton firmware WHENCE | 26 +++----------------------- intel/dsp_fw_bxtn_v1118.bin | Bin 505608 -> 0 bytes intel/dsp_fw_bxtn_v2010.bin | Bin 505608 -> 0 bytes intel/dsp_fw_bxtn_v430.bin | Bin 505608 -> 0 bytes intel/dsp_fw_bxtn_v702.bin | Bin 505608 -> 0 bytes intel/dsp_fw_kbl.bin | 2 +- intel/dsp_fw_kbl_v2630.bin | Bin 0 -> 243016 bytes intel/dsp_fw_release_v827.bin | Bin 233472 -> 0 bytes intel/dsp_fw_release_v869.bin | Bin 233472 -> 0 bytes intel/dsp_fw_release_v896.bin | Bin 233472 -> 0 bytes intel/dsp_fw_release_v927.bin | Bin 233472 -> 0 bytes intel/dsp_fw_release_v948.bin | Bin 221184 -> 0 bytes intel/dsp_fw_release_v951.bin | Bin 221184 -> 0 bytes intel/dsp_fw_release_v958.bin | Bin 221184 -> 0 bytes 14 files changed, 4 insertions(+), 24 deletions(-) delete mode 100644 intel/dsp_fw_bxtn_v1118.bin delete mode 100644 intel/dsp_fw_bxtn_v2010.bin delete mode 100644 intel/dsp_fw_bxtn_v430.bin delete mode 100644 intel/dsp_fw_bxtn_v702.bin create mode 100644 intel/dsp_fw_kbl_v2630.bin delete mode 100644 intel/dsp_fw_release_v827.bin delete mode 100644 intel/dsp_fw_release_v869.bin delete mode 100644 intel/dsp_fw_release_v896.bin delete mode 100644 intel/dsp_fw_release_v927.bin delete mode 100644 intel/dsp_fw_release_v948.bin delete mode 100644 intel/dsp_fw_release_v951.bin delete mode 100644 intel/dsp_fw_release_v958.bin Thanks -- ~Vinod

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[alsa-devel] [PATCH] ASoC: rt5645: make rt5645_platform_data const
by Bhumika Goyal 23 Aug '17

23 Aug '17

Make these const as they are only used during a copy operation. Done using Coccinelle. @match disable optional_qualifier@ identifier s; @@ static struct rt5645_platform_data s = {...}; @ref@ position p; identifier match.s; @@ s@p @good1@ position ref.p; identifier match.s,f,c; expression e; @@ ( e = s@p | e = s(a)p.f | c(...,s@p.f,...) | c(...,s@p,...) ) @bad depends on !good1@ position ref.p; identifier match.s; @@ s@p @depends on forall !bad disable optional_qualifier@ identifier match.s; @@ static + const struct rt5645_platform_data s; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/codecs/rt5645.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index f14f648..efec452 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3559,7 +3559,7 @@ static int rt5645_resume(struct snd_soc_codec *codec) MODULE_DEVICE_TABLE(acpi, rt5645_acpi_match); #endif -static struct rt5645_platform_data general_platform_data = { +static const struct rt5645_platform_data general_platform_data = { .dmic1_data_pin = RT5645_DMIC1_DISABLE, .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, .jd_mode = 3, @@ -3593,7 +3593,7 @@ static int rt5645_resume(struct snd_soc_codec *codec) { } }; -static struct rt5645_platform_data buddy_platform_data = { +static const struct rt5645_platform_data buddy_platform_data = { .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, .jd_mode = 3, @@ -3610,7 +3610,7 @@ static int rt5645_resume(struct snd_soc_codec *codec) { } }; -static struct rt5645_platform_data gpd_win_platform_data = { +static const struct rt5645_platform_data gpd_win_platform_data = { .jd_mode = 3, .inv_jd1_1 = true, }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: tegra: Remove superfluous 'machine' variable
by Arnd Bergmann 23 Aug '17

23 Aug '17

The last cleanup left behind an unused variable: sound/soc/tegra/tegra_wm8903.c: In function 'tegra_wm8903_remove': sound/soc/tegra/tegra_wm8903.c:206:23: error: unused variable 'machine' [-Werror=unused-variable] This removes it as welremoves it as welll Fixes: 2a8bd83af30e ("ASoC: tegra: Remove superfluous snd_soc_jack_free_gpios() call") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/tegra/tegra_wm8903.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 2384cb9cb95c..18bdae59a4df 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -203,7 +203,6 @@ static int tegra_wm8903_remove(struct snd_soc_card *card) snd_soc_get_pcm_runtime(card, card->dai_link[0].name); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); wm8903_mic_detect(codec, NULL, 0, 0); -- 2.9.0

2 2

[alsa-devel] [PATCH] ALSA: ctxfi: make hw structures const
by Bhumika Goyal 23 Aug '17

23 Aug '17

Make these const as they are only used in a copy operation. Done using Coccinelle: @match disable optional_qualifier@ identifier s; @@ static struct hw s = {...}; @ref@ position p; identifier match.s; @@ s@p @good1@ position ref.p; identifier match.s,f,c; expression e; @@ ( e = s@p | e = s(a)p.f | c(...,s@p.f,...) | c(...,s@p,...) ) @bad depends on !good1@ position ref.p; identifier match.s; @@ s@p @depends on forall !bad disable optional_qualifier@ identifier match.s; @@ static + const struct hw s; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/ctxfi/cthw20k1.c | 2 +- sound/pci/ctxfi/cthw20k2.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 79edd88..8e6eb9d 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -2154,7 +2154,7 @@ static void hw_write_pci(struct hw *hw, u32 reg, u32 data) &container_of(hw, struct hw20k1, hw)->reg_pci_lock, flags); } -static struct hw ct20k1_preset = { +static const struct hw ct20k1_preset = { .irq = -1, .card_init = hw_card_init, diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 18ee776..b866d6b 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -2220,7 +2220,7 @@ static void hw_write_20kx(struct hw *hw, u32 reg, u32 data) writel(data, hw->mem_base + reg); } -static struct hw ct20k2_preset = { +static const struct hw ct20k2_preset = { .irq = -1, .card_init = hw_card_init, -- 1.9.1

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[alsa-devel] [PATCH 0/4] constify ac97_pcm structures
by Arvind Yadav 23 Aug '17

23 Aug '17

ac97_pcm are not supposed to change at runtime. All functions working with ac97_pcm provided by <sound/ac97_codec.h> work with const ac97_pcm. So mark the non-const structs as const. Arvind Yadav (4): [PATCH 1/4] ALSA: aaci: constify ac97_pcm structures [PATCH 2/4] ALSA: ac97c: constify ac97_pcm structures [PATCH 3/4] ALSA: atiixp: constify ac97_pcm structures [PATCH 4/4] ALSA: intel8x0: constify ac97_pcm structures sound/arm/aaci.c | 2 +- sound/atmel/ac97c.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/intel8x0.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH] ALSA: Fireface: Use common error handling code in pcm_open()
by SF Markus Elfring 23 Aug '17

23 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 22 Aug 2017 21:45:42 +0200 Add a jump target so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/firewire/fireface/ff-pcm.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index d12a0e3a4219..40975395a64a 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -138,16 +138,12 @@ static int pcm_open(struct snd_pcm_substream *substream) return err; err = pcm_init_hw_params(ff, substream); - if (err < 0) { - snd_ff_stream_lock_release(ff); - return err; - } + if (err < 0) + goto release_lock; err = ff->spec->protocol->get_clock(ff, &rate, &src); - if (err < 0) { - snd_ff_stream_lock_release(ff); - return err; - } + if (err < 0) + goto release_lock; if (src != SND_FF_CLOCK_SRC_INTERNAL) { for (i = 0; i < CIP_SFC_COUNT; ++i) { @@ -177,6 +173,10 @@ static int pcm_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); return 0; + +release_lock: + snd_ff_stream_lock_release(ff); + return err; } static int pcm_close(struct snd_pcm_substream *substream) -- 2.14.0

3 4

[alsa-devel] Fixed latency between multiple channels
by cuhl＠eit.uni-kl.de 23 Aug '17

23 Aug '17

Hello Alsa-Developers, currently I try to decide, if ALSA is useable for a certain multi-channel signal processing task. I've read, that even with a realtime kernel the input-output latency varies with time - especially if USB is inbetween. Even so this is not what I actually want, I think the application might be sufficiently robust against this. However I expect to require a fixed latency between multiple ( I2S ) channels. This shall include stereo, but extend to all channels ( between 6 to 12 mono / 3 to 6 srereo channels probably ). So every channel's signal shall be exactly determined to its relative position with respect to the other channels, e. g. it shall be known inside the system which events occured simultanously. Likewise, the processed or generated signals shall be transferred to the outputs in a way, that they occur there with intended and modifiable relative latency to each other. Is this ensured by the ALSA architecture or is it at least possible to ensure it, if it is used? Any comments or answers are appreciated. Kind regards Christian

2 1

[alsa-devel] [asoc:topic/intel 52/54] sound/soc/intel/skylake/skl-messages.c:320:9: warning: 'ret' may be used uninitialized in this function
by kbuild test robot 23 Aug '17

23 Aug '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 7b992c24de78a206b3abd07192686d2d5db5012c commit: f77d443c4c299aff5ad9c74811dd063f4d8bebcf [52/54] ASoC: Intel: Skylake: Fix to free resources for dsp_init failure config: x86_64-randconfig-ne0-08232016 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout f77d443c4c299aff5ad9c74811dd063f4d8bebcf # save the attached .config to linux build tree make ARCH=x86_64 Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): sound/soc/intel/skylake/skl-messages.c: In function 'skl_init_dsp': >> sound/soc/intel/skylake/skl-messages.c:320:9: warning: 'ret' may be used uninitialized in this function [-Wmaybe-uninitialized] return ret; ^~~ vim +/ret +320 sound/soc/intel/skylake/skl-messages.c bc23ca35c Jeeja KP 2016-03-11 256 d255b0955 Jeeja KP 2015-07-21 257 int skl_init_dsp(struct skl *skl) d255b0955 Jeeja KP 2015-07-21 258 { d255b0955 Jeeja KP 2015-07-21 259 void __iomem *mmio_base; d255b0955 Jeeja KP 2015-07-21 260 struct hdac_ext_bus *ebus = &skl->ebus; d255b0955 Jeeja KP 2015-07-21 261 struct hdac_bus *bus = ebus_to_hbus(ebus); d255b0955 Jeeja KP 2015-07-21 262 struct skl_dsp_loader_ops loader_ops; bc23ca35c Jeeja KP 2016-03-11 263 int irq = bus->irq; 73a675816 Vinod Koul 2016-07-26 264 const struct skl_dsp_ops *ops; f0a550a88 Guneshwor Singh 2017-08-02 265 struct skl_dsp_cores *cores; 73a675816 Vinod Koul 2016-07-26 266 int ret; d255b0955 Jeeja KP 2015-07-21 267 d255b0955 Jeeja KP 2015-07-21 268 /* enable ppcap interrupt */ d255b0955 Jeeja KP 2015-07-21 269 snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true); d255b0955 Jeeja KP 2015-07-21 270 snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true); d255b0955 Jeeja KP 2015-07-21 271 d255b0955 Jeeja KP 2015-07-21 272 /* read the BAR of the ADSP MMIO */ d255b0955 Jeeja KP 2015-07-21 273 mmio_base = pci_ioremap_bar(skl->pci, 4); d255b0955 Jeeja KP 2015-07-21 274 if (mmio_base == NULL) { d255b0955 Jeeja KP 2015-07-21 275 dev_err(bus->dev, "ioremap error\n"); d255b0955 Jeeja KP 2015-07-21 276 return -ENXIO; d255b0955 Jeeja KP 2015-07-21 277 } d255b0955 Jeeja KP 2015-07-21 278 73a675816 Vinod Koul 2016-07-26 279 ops = skl_get_dsp_ops(skl->pci->device); f77d443c4 Subhransu S. Prusty 2017-08-22 280 if (!ops) { f77d443c4 Subhransu S. Prusty 2017-08-22 281 goto unmap_mmio; f77d443c4 Subhransu S. Prusty 2017-08-22 282 ret = -EIO; f77d443c4 Subhransu S. Prusty 2017-08-22 283 } bc23ca35c Jeeja KP 2016-03-11 284 73a675816 Vinod Koul 2016-07-26 285 loader_ops = ops->loader_ops(); 73a675816 Vinod Koul 2016-07-26 286 ret = ops->init(bus->dev, mmio_base, irq, 73a675816 Vinod Koul 2016-07-26 287 skl->fw_name, loader_ops, 73a675816 Vinod Koul 2016-07-26 288 &skl->skl_sst); bc23ca35c Jeeja KP 2016-03-11 289 2ac454ffe Jeeja KP 2015-10-22 290 if (ret < 0) f77d443c4 Subhransu S. Prusty 2017-08-22 291 goto unmap_mmio; 2ac454ffe Jeeja KP 2015-10-22 292 7bd86a305 G Kranthi 2017-03-13 293 skl->skl_sst->dsp_ops = ops; f0a550a88 Guneshwor Singh 2017-08-02 294 cores = &skl->skl_sst->cores; f0a550a88 Guneshwor Singh 2017-08-02 295 cores->count = ops->num_cores; f0a550a88 Guneshwor Singh 2017-08-02 296 f0a550a88 Guneshwor Singh 2017-08-02 297 cores->state = kcalloc(cores->count, sizeof(*cores->state), GFP_KERNEL); f77d443c4 Subhransu S. Prusty 2017-08-22 298 if (!cores->state) { f77d443c4 Subhransu S. Prusty 2017-08-22 299 ret = -ENOMEM; f77d443c4 Subhransu S. Prusty 2017-08-22 300 goto unmap_mmio; f77d443c4 Subhransu S. Prusty 2017-08-22 301 } f0a550a88 Guneshwor Singh 2017-08-02 302 f0a550a88 Guneshwor Singh 2017-08-02 303 cores->usage_count = kcalloc(cores->count, sizeof(*cores->usage_count), f0a550a88 Guneshwor Singh 2017-08-02 304 GFP_KERNEL); f0a550a88 Guneshwor Singh 2017-08-02 305 if (!cores->usage_count) { f77d443c4 Subhransu S. Prusty 2017-08-22 306 ret = -ENOMEM; f77d443c4 Subhransu S. Prusty 2017-08-22 307 goto free_core_state; f0a550a88 Guneshwor Singh 2017-08-02 308 } 363d45385 Dharageswari R 2017-08-02 309 d255b0955 Jeeja KP 2015-07-21 310 dev_dbg(bus->dev, "dsp registration status=%d\n", ret); d255b0955 Jeeja KP 2015-07-21 311 f77d443c4 Subhransu S. Prusty 2017-08-22 312 return 0; f77d443c4 Subhransu S. Prusty 2017-08-22 313 f77d443c4 Subhransu S. Prusty 2017-08-22 314 free_core_state: f77d443c4 Subhransu S. Prusty 2017-08-22 315 kfree(cores->state); f77d443c4 Subhransu S. Prusty 2017-08-22 316 f77d443c4 Subhransu S. Prusty 2017-08-22 317 unmap_mmio: f77d443c4 Subhransu S. Prusty 2017-08-22 318 iounmap(mmio_base); f77d443c4 Subhransu S. Prusty 2017-08-22 319 d255b0955 Jeeja KP 2015-07-21 @320 return ret; d255b0955 Jeeja KP 2015-07-21 321 } d255b0955 Jeeja KP 2015-07-21 322 :::::: The code at line 320 was first introduced by commit :::::: d255b09555e8ea0e517a4d3368ec60097ac9cf96 ASoC: Intel: Skylake: Add dsp and ipc init helpers :::::: TO: Jeeja KP <jeeja.kp(a)intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: tegra: Fix unused variable warning
by Takashi Iwai 23 Aug '17

23 Aug '17

Fixes: 2a8bd83af30e ("ASoC: tegra: Remove superfluous snd_soc_jack_free_gpios() call") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/tegra/tegra_wm8903.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index ce498cc0914b..05f95944669a 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -203,7 +203,6 @@ static int tegra_wm8903_remove(struct snd_soc_card *card) snd_soc_get_pcm_runtime(card, card->dai_link[0].name); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); wm8903_mic_detect(codec, NULL, 0, 0); -- 2.14.0

2 1

[alsa-devel] [PATCH] ASoC: wm8524: remove unnecessary snd_soc_unregister_platform()
by Kuninori Morimoto 23 Aug '17

23 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> wm8524 doesn't register platform, unregister platform is not needed Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/wm8524.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index b66147e..856a695 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -233,10 +233,8 @@ static int wm8524_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8524, &wm8524_dai, 1); - if (ret < 0) { + if (ret < 0) dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); - snd_soc_unregister_platform(&pdev->dev); - } return ret; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: simple-scu-card: Add pm callbacks to platform driver
by Kuninori Morimoto 23 Aug '17

23 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using simple-scu-card. Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-scu-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 0bb3da7..48606c6 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -300,6 +300,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev) static struct platform_driver asoc_simple_card = { .driver = { .name = "simple-scu-audio-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-card: Add pm callbacks to platform driver
by Kuninori Morimoto 23 Aug '17

23 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using audio-graph-card. Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 2f7c8cf..488c52f 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -325,6 +325,7 @@ static int asoc_graph_card_remove(struct platform_device *pdev) static struct platform_driver asoc_graph_card = { .driver = { .name = "asoc-audio-graph-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_graph_of_match, }, .probe = asoc_graph_card_probe, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-scu-card: Add pm callbacks to platform driver
by Kuninori Morimoto 23 Aug '17

23 Aug '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using audio-graph-scu-card. Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-scu-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 06b6fcf..a967aa1 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -401,6 +401,7 @@ static int asoc_graph_card_remove(struct platform_device *pdev) static struct platform_driver asoc_graph_card = { .driver = { .name = "asoc-audio-graph-scu-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_graph_of_match, }, .probe = asoc_graph_card_probe, -- 1.9.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: Intel: Skylake: Fixes to free resources on init failure
by Subhransu S. Prusty 23 Aug '17

23 Aug '17

Some resources are not freed for ipc and dsp init failures. This results in random failures in a stress test environment. So fixed to free the required resources and also fixed the DSP core reference counting for firmware load failure. Subhransu S. Prusty (4): ASoC: Intel: Skylake: Fix to free dsp resource on ipc_init failure ASoC: Intel: Skylake: Fix to free resources for dsp_init failure ASoC: Intel: Skylake: Fix to free correct dev id in free_irq ASoC: Intel: Skylake: Fix DSP core ref count for init failure sound/soc/intel/skylake/bxt-sst.c | 4 +++- sound/soc/intel/skylake/cnl-sst.c | 4 +++- sound/soc/intel/skylake/skl-messages.c | 26 +++++++++++++++++++------- sound/soc/intel/skylake/skl-sst-dsp.c | 4 ++-- sound/soc/intel/skylake/skl-sst.c | 4 +++- sound/soc/intel/skylake/skl.c | 2 +- 6 files changed, 31 insertions(+), 13 deletions(-) -- 1.9.1

3 9

[alsa-devel] [PATCH][next] ASoC: tegra: remove unused variable machine, fixes build warning
by Colin King 23 Aug '17

23 Aug '17

From: Colin Ian King <colin.king(a)canonical.com> Remove unused variable 'machine', fixes build warning: sound/soc/tegra/tegra_wm8903.c: In function 'tegra_wm8903_remove': sound/soc/tegra/tegra_wm8903.c:206:23: warning: unused variable 'machine' [-Wunused-variable] struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); ^~~~~~~ Fixes: 2a8bd83af30e ("ASoC: tegra: Remove superfluous snd_soc_jack_free_gpios() call") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/tegra/tegra_wm8903.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 2384cb9cb95c..18bdae59a4df 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -203,7 +203,6 @@ static int tegra_wm8903_remove(struct snd_soc_card *card) snd_soc_get_pcm_runtime(card, card->dai_link[0].name); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); wm8903_mic_detect(codec, NULL, 0, 0); -- 2.14.1

2 1

[alsa-devel] [PATCH] ALSA: USB-MIDI: Use common error handling code in __snd_usbmidi_create()
by SF Markus Elfring 23 Aug '17

23 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 23 Aug 2017 12:20:07 +0200 Add jump targets so that a bit of exception handling can be better reused at the end of this function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/usb/midi.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index a35f41467237..bd9d02268724 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2435,10 +2435,8 @@ int __snd_usbmidi_create(struct snd_card *card, err = -ENXIO; break; } - if (err < 0) { - kfree(umidi); - return err; - } + if (err < 0) + goto free_midi; /* create rawmidi device */ out_ports = 0; @@ -2448,23 +2446,25 @@ int __snd_usbmidi_create(struct snd_card *card, in_ports += hweight16(endpoints[i].in_cables); } err = snd_usbmidi_create_rawmidi(umidi, out_ports, in_ports); - if (err < 0) { - kfree(umidi); - return err; - } + if (err < 0) + goto free_midi; /* create endpoint/port structures */ if (quirk && quirk->type == QUIRK_MIDI_MIDIMAN) err = snd_usbmidi_create_endpoints_midiman(umidi, &endpoints[0]); else err = snd_usbmidi_create_endpoints(umidi, endpoints); - if (err < 0) { - return err; - } + if (err < 0) + goto exit; usb_autopm_get_interface_no_resume(umidi->iface); list_add_tail(&umidi->list, midi_list); return 0; + +free_midi: + kfree(umidi); +exit: + return err; } EXPORT_SYMBOL(__snd_usbmidi_create); -- 2.14.0

3 2

[alsa-devel] [PATCH] ASoC: Add delay quirk for H650e/Jabra 550a USB headsets
by Joakim Tjernlund 23 Aug '17

23 Aug '17

These headsets reports a lot of: cannot set freq 44100 to ep 0x81 and need a small delay between sample rate settings, just like Zoom R16/24. Add both headsets to the Zoom R16/24 quirk for a 1 ms delay between control msgs. Signed-off-by: Joakim Tjernlund <joakim.tjernlund(a)infinera.com> --- sound/usb/quirks.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index eb4b9f7a571e..6f395150243f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1308,10 +1308,13 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); - /* Zoom R16/24 needs a tiny delay here, otherwise requests like - * get/set frequency return as failed despite actually succeeding. + /* Zoom R16/24, Logitech H650e, Jabra 550a needs a tiny delay here, + * otherwise requests like get/set frequency return as failed despite + * actually succeeding. */ - if (chip->usb_id == USB_ID(0x1686, 0x00dd) && + if ((chip->usb_id == USB_ID(0x1686, 0x00dd) || + chip->usb_id == USB_ID(0x046d, 0x0a46) || + chip->usb_id == USB_ID(0x0b0e, 0x0349)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(1); } -- 2.13.5

2 2

[alsa-devel] [PATCH] ALSA: core: Use common error handling code in two functions
by SF Markus Elfring 23 Aug '17

23 Aug '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 22 Aug 2017 21:01:01 +0200 Add jump targets so that a bit of exception handling can be better reused at the end of these functions. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/core/pcm.c | 30 ++++++++++++++++++------------ sound/core/timer.c | 10 ++++++---- 2 files changed, 24 insertions(+), 16 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 89c7485519cb..048df9658f50 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -783,21 +783,27 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device, INIT_LIST_HEAD(&pcm->list); if (id) strlcpy(pcm->id, id, sizeof(pcm->id)); - if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) { - snd_pcm_free(pcm); - return err; - } - if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_CAPTURE, capture_count)) < 0) { - snd_pcm_free(pcm); - return err; - } - if ((err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops)) < 0) { - snd_pcm_free(pcm); - return err; - } + + err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, + playback_count); + if (err < 0) + goto free_pcm; + + err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_CAPTURE, capture_count); + if (err < 0) + goto free_pcm; + + err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops); + if (err < 0) + goto free_pcm; + if (rpcm) *rpcm = pcm; return 0; + +free_pcm: + snd_pcm_free(pcm); + return err; } /** diff --git a/sound/core/timer.c b/sound/core/timer.c index a9b9a277e00c..6d73a63f6e2b 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -2096,8 +2096,7 @@ static int __init alsa_timer_init(void) err = snd_timer_register_system(); if (err < 0) { pr_err("ALSA: unable to register system timer (%i)\n", err); - put_device(&timer_dev); - return err; + goto put_timer; } err = snd_register_device(SNDRV_DEVICE_TYPE_TIMER, NULL, 0, @@ -2105,12 +2104,15 @@ static int __init alsa_timer_init(void) if (err < 0) { pr_err("ALSA: unable to register timer device (%i)\n", err); snd_timer_free_all(); - put_device(&timer_dev); - return err; + goto put_timer; } snd_timer_proc_init(); return 0; + +put_timer: + put_device(&timer_dev); + return err; } static void __exit alsa_timer_exit(void) -- 2.14.0

2 14

[alsa-devel] [sound:topic/asus-e100h-4.13 13/17] arch/x86/platform/atom/punit_atom_debug.c:230:41: error: 'punit_freeze_nb' undeclared
by kbuild test robot 23 Aug '17

23 Aug '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/asus-e100h-4.13 head: b50dc59e3e31709349588be7a1ac8991e9c5f8a5 commit: 7cfa6efe32e12013183d68f991dd586814b740da [13/17] Intel Atom suspend: add debug check for S0ix blockers config: x86_64-randconfig-x008-201734 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 7cfa6efe32e12013183d68f991dd586814b740da # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): arch/x86/platform/atom/punit_atom_debug.c: In function 'punit_atom_debug_exit': >> arch/x86/platform/atom/punit_atom_debug.c:230:41: error: 'punit_freeze_nb' undeclared (first use in this function) intel_idle_freeze_notifier_unregister(&punit_freeze_nb.nb); ^~~~~~~~~~~~~~~ arch/x86/platform/atom/punit_atom_debug.c:230:41: note: each undeclared identifier is reported only once for each function it appears in vim +/punit_freeze_nb +230 arch/x86/platform/atom/punit_atom_debug.c 226 227 static void __exit punit_atom_debug_exit(void) 228 { 229 #ifdef CONFIG_PM_DEBUG > 230 intel_idle_freeze_notifier_unregister(&punit_freeze_nb.nb); 231 #endif 232 punit_dbgfs_unregister(); 233 } 234 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ALSA: hda - Add stereo mic quirk for Lenovo G50-70 (17aa:3978)
by Takashi Iwai 23 Aug '17

23 Aug '17

Lenovo G50-70 (17aa:3978) with Conexant codec chip requires the similar workaround for the inverted stereo dmic like other Lenovo models. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1020657 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8c1289963c80..a81aacf684b2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -947,6 +947,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo G50-70", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), -- 2.14.0

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