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

[alsa-devel] [PATCH 1/2] ALSA: usb: Fix a typo in Tascam US-16x08 mixer element
by Takashi Iwai 30 May '17

30 May '17

A mixer element created in a quirk for Tascam US-16x08 contains a typo: it should be "EQ MidLow Q" instead of "EQ MidQLow Q". Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195875 Fixes: d2bb390a2081 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk") Cc: <stable(a)vger.kernel.org> # v4.11+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_us16x08.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 29d2c9282987..442d8f7998e3 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -1135,7 +1135,7 @@ static const struct snd_us16x08_control_params eq_controls[] = { .control_id = SND_US16X08_ID_EQLOWMIDWIDTH, .type = USB_MIXER_U8, .num_channels = 16, - .name = "EQ MidQLow Q", + .name = "EQ MidLow Q", }, { /* EQ mid high gain */ .kcontrol_new = &snd_us16x08_eq_gain_ctl, -- 2.13.0

1 1

[alsa-devel] [PATCH] Revert "ALSA: usb-audio: purge needless variable length array"
by Takashi Iwai 30 May '17

30 May '17

This reverts commit 89b593c30e83 ("ALSA: usb-audio: purge needless variable length array"). The patch turned out to cause a severe regression, triggering an Oops at snd_usb_ctl_msg(). It was overseen that snd_usb_ctl_msg() writes back the response to the given buffer, while the patch changed it to a read-only const buffer. (One should always double-check when an extra pointer cast is present...) As a simple fix, just revert the affected commit. It was merely a cleanup. Although it brings VLA again, it's clearer as a fix. We'll address the VLA later in another patch. Fixes: 89b593c30e83 ("ALSA: usb-audio: purge needless variable length array") Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195875 Cc: <stable(a)vger.kernel.org> # v4.11+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_us16x08.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index dc48eedea92e..29d2c9282987 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -698,12 +698,12 @@ static int snd_us16x08_meter_get(struct snd_kcontrol *kcontrol, struct snd_usb_audio *chip = elem->head.mixer->chip; struct snd_us16x08_meter_store *store = elem->private_data; u8 meter_urb[64]; - char tmp[sizeof(mix_init_msg2)] = {0}; + char tmp[max(sizeof(mix_init_msg1), sizeof(mix_init_msg2))]; switch (kcontrol->private_value) { case 0: - snd_us16x08_send_urb(chip, (char *)mix_init_msg1, - sizeof(mix_init_msg1)); + memcpy(tmp, mix_init_msg1, sizeof(mix_init_msg1)); + snd_us16x08_send_urb(chip, tmp, 4); snd_us16x08_recv_urb(chip, meter_urb, sizeof(meter_urb)); kcontrol->private_value++; @@ -721,7 +721,7 @@ static int snd_us16x08_meter_get(struct snd_kcontrol *kcontrol, case 3: memcpy(tmp, mix_init_msg2, sizeof(mix_init_msg2)); tmp[2] = snd_get_meter_comp_index(store); - snd_us16x08_send_urb(chip, tmp, sizeof(mix_init_msg2)); + snd_us16x08_send_urb(chip, tmp, 10); snd_us16x08_recv_urb(chip, meter_urb, sizeof(meter_urb)); kcontrol->private_value = 0; -- 2.13.0

1 0

[alsa-devel] [RFC PATCH] ASoC: Intel: sst: Delete sst_save_shim64(); saved regs are never used
by Douglas Anderson 30 May '17

30 May '17

In commit 9a075265c6dc ("ASoC: Intel: sst: Remove unused function sst_restore_shim64()"), we deleted the sst_restore_shim64() since it was never used. ...but a quick look at the code shows that we should also be able to remove the sst_save_shim64() function and the structure members we were storing data in. Note that the saving code (and the comments talking about how important it is to do the save) has been around since commit 336cfbb05edf ("ASoC: Intel: mrfld- add ACPI module"). Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- This problem was found only by code inspection and only compile tested. Hence, RFC. sound/soc/intel/atom/sst/sst.c | 19 ------------------- sound/soc/intel/atom/sst/sst.h | 2 -- 2 files changed, 21 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index 2d43b8693c0c..5ee92257ca85 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -382,21 +382,6 @@ void sst_context_cleanup(struct intel_sst_drv *ctx) } EXPORT_SYMBOL_GPL(sst_context_cleanup); -static inline void sst_save_shim64(struct intel_sst_drv *ctx, - void __iomem *shim, - struct sst_shim_regs64 *shim_regs) -{ - unsigned long irq_flags; - - spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags); - - shim_regs->imrx = sst_shim_read64(shim, SST_IMRX); - shim_regs->csr = sst_shim_read64(shim, SST_CSR); - - - spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags); -} - void sst_configure_runtime_pm(struct intel_sst_drv *ctx) { pm_runtime_set_autosuspend_delay(ctx->dev, SST_SUSPEND_DELAY); @@ -416,8 +401,6 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) pm_runtime_set_active(ctx->dev); else pm_runtime_put_noidle(ctx->dev); - - sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); } EXPORT_SYMBOL_GPL(sst_configure_runtime_pm); @@ -441,8 +424,6 @@ static int intel_sst_runtime_suspend(struct device *dev) flush_workqueue(ctx->post_msg_wq); ctx->ops->reset(ctx); - /* save the shim registers because PMC doesn't save state */ - sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); return ret; } diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index 5c9a51cc77aa..d8cac09e8997 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -318,12 +318,10 @@ struct sst_ipc_reg { }; struct sst_shim_regs64 { - u64 csr; u64 pisr; u64 pimr; u64 isrx; u64 isrd; - u64 imrx; u64 imrd; u64 ipcx; u64 ipcd; -- 2.13.0.219.gdb65acc882-goog

3 2

[alsa-devel] [PATCH][V2] sound/core/pcm_timer.c: include pcm_local.h and remove some extraneous tabs
by Colin King 30 May '17

30 May '17

From: Colin Ian King <colin.king(a)canonical.com> We need to include pcm_local.h to clean up some smatch warnings: symbol 'snd_pcm_timer_done' was not declared. Should it be static? symbol 'snd_pcm_timer_init' was not declared. Should it be static? symbol 'snd_pcm_timer_resolution_change' was not declared. Should it be static? Also remove some extraneous tabs on empty lines and replace space intentation with a tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/core/pcm_timer.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c index 20ecd8f18080..11389f13de73 100644 --- a/sound/core/pcm_timer.c +++ b/sound/core/pcm_timer.c @@ -25,6 +25,8 @@ #include <sound/pcm.h> #include <sound/timer.h> +#include "pcm_local.h" + /* * Timer functions */ @@ -33,8 +35,8 @@ void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream) { unsigned long rate, mult, fsize, l, post; struct snd_pcm_runtime *runtime = substream->runtime; - - mult = 1000000000; + + mult = 1000000000; rate = runtime->rate; if (snd_BUG_ON(!rate)) return; @@ -65,7 +67,7 @@ void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream) static unsigned long snd_pcm_timer_resolution(struct snd_timer * timer) { struct snd_pcm_substream *substream; - + substream = timer->private_data; return substream->runtime ? substream->runtime->timer_resolution : 0; } @@ -73,7 +75,7 @@ static unsigned long snd_pcm_timer_resolution(struct snd_timer * timer) static int snd_pcm_timer_start(struct snd_timer * timer) { struct snd_pcm_substream *substream; - + substream = snd_timer_chip(timer); substream->timer_running = 1; return 0; @@ -82,7 +84,7 @@ static int snd_pcm_timer_start(struct snd_timer * timer) static int snd_pcm_timer_stop(struct snd_timer * timer) { struct snd_pcm_substream *substream; - + substream = snd_timer_chip(timer); substream->timer_running = 0; return 0; @@ -112,7 +114,7 @@ void snd_pcm_timer_init(struct snd_pcm_substream *substream) { struct snd_timer_id tid; struct snd_timer *timer; - + tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE; tid.dev_class = SNDRV_TIMER_CLASS_PCM; tid.card = substream->pcm->card->number; -- 2.11.0

2 1

[alsa-devel] [alsa-lib] bug in USB-Audio.conf
by Alexander Tsoy 30 May '17

30 May '17

Hello! I've noticed that there is a bug in USB-Audio.conf. "USB- Audio.pcm.default" doesn't allow custom "default" definitions, but there is one: USB-Audio."Audiophile USB (tm)".pcm.default Moreover there is a typo in this definition in dsnoop args: DEVICE=1 -> DEV=1. So what is the best course of action here? We can fix USB- Audio.conf (I have a working patch that I've tested with another card) or given that Audiophile USB is likely already in a semi-broken state for a long time, may be it is better to remove the exclusion in the driver [1] and remove all definitions for this card from USB- Audio.conf? [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/ commit/?id=5ee20bc792467d7d612157e0a9962765aa943b08 -- Alexander Tsoy <alexander(a)tsoy.me>

2 1

[alsa-devel] [PATCH] pcm: dshare: Call snd_pcm_dshare_state() directly
by Takashi Iwai 30 May '17

30 May '17

... otherwise it may be a deadlock if recursive lock isn't available. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_dshare.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 4c57fb4a3028..7cdfea2a1572 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -227,6 +227,8 @@ static int snd_pcm_dshare_sync_ptr(snd_pcm_t *pcm) * plugin implementation */ +static snd_pcm_state_t snd_pcm_dshare_state(snd_pcm_t *pcm); + static int snd_pcm_dshare_status(snd_pcm_t *pcm, snd_pcm_status_t * status) { snd_pcm_direct_t *dshare = pcm->private_data; @@ -244,7 +246,7 @@ static int snd_pcm_dshare_status(snd_pcm_t *pcm, snd_pcm_status_t * status) default: break; } - status->state = snd_pcm_state(pcm); + status->state = snd_pcm_dshare_state(pcm); status->trigger_tstamp = dshare->trigger_tstamp; status->avail = snd_pcm_mmap_playback_avail(pcm); status->avail_max = status->avail > dshare->avail_max ? status->avail : dshare->avail_max; -- 2.13.0

1 0

[alsa-devel] [PATCH v2 0/3] ASoC: Intel: Skylake: Add Support for MIC select module
by Subhransu S. Prusty 30 May '17

30 May '17

This series adds support in DSP to add support for MIC select module which allows user to select 1 to N channels from DMIC channel input. Further it also adds support in bxt-rt298 machine for such a module. Changes in v2: - check for index in array range Dharageswari R (3): ASoC: Intel: Skylake: Add mic-select module type ASoC: Intel: Skylake: Add enum control for mic selection ASoC: Intel: Boards: Add 4-channel DMIC fixup. sound/soc/intel/boards/bxt_rt298.c | 7 +- sound/soc/intel/skylake/skl-messages.c | 2 + sound/soc/intel/skylake/skl-topology.c | 155 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 20 ++++ sound/soc/intel/skylake/skl-tplg-interface.h | 2 + 5 files changed, 181 insertions(+), 5 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH] ASoC: sgtl5000: add avc enable control
by Richard Leitner 30 May '17

30 May '17

This control enables the automatic volume control (AVC) of SGTL5000 using its DAP_AVC_CTRL register. Signed-off-by: Richard Leitner <richard.leitner(a)skidata.com> --- sound/soc/codecs/sgtl5000.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 5a2702e..ad8c9a8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -434,6 +434,8 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 0x1f, 1, lineout_volume), SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1), + + SOC_SINGLE("AVC Enable", SGTL5000_DAP_AVC_CTRL, 0, 1, 0), }; /* mute the codec used by alsa core */ -- 2.1.4

2 2

[alsa-devel] [PATCH] gpiolib: remove unused variable 'i'
by Colin King 30 May '17

30 May '17

From: Colin Ian King <colin.king(a)canonical.com> Fixes smatch warning: unused variable 'i' [-Wunused-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- drivers/gpio/gpiolib.c | 2 -- sound/soc/generic/audio-graph-scu-card.c | 2 +- 2 files changed, 1 insertion(+), 3 deletions(-) diff --git a/drivers/gpio/gpiolib.c b/drivers/gpio/gpiolib.c index be8097097326..62ffb4e293d2 100644 --- a/drivers/gpio/gpiolib.c +++ b/drivers/gpio/gpiolib.c @@ -1471,8 +1471,6 @@ static struct gpio_chip *find_chip_by_name(const char *name) static int gpiochip_irqchip_init_valid_mask(struct gpio_chip *gpiochip) { - int i; - if (!gpiochip->irq_need_valid_mask) return 0; diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index c5c52d1f500c..e15f63c74f59 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -246,7 +246,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) continue; if (rcpu_ep != cpu_ep) { - dev_err(dev, "remote-endpoint missmatch (%s/%s/%s)\n", + dev_err(dev, "remote-endpoint mismatch (%s/%s/%s)\n", cpu_ep->name, codec_ep->name, rcpu_ep->name); ret = -EINVAL; goto parse_of_err; -- 2.11.0

3 2

[alsa-devel] [PATCH] ALSA: sound/core/pcm_timer.c: include pcm_local.h and remove some extraneous tabs
by Colin King 30 May '17

30 May '17

From: Colin Ian King <colin.king(a)canonical.com> We need to include pcm_local.h to clean up some smatch warnings: symbol 'snd_pcm_timer_done' was not declared. Should it be static? symbol 'snd_pcm_timer_init' was not declared. Should it be static? symbol 'snd_pcm_timer_resolution_change' was not declared. Should it be static? Also remove some extraneous tabs on empty lines. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/core/pcm_timer.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c index 20ecd8f18080..b182cac00dd8 100644 --- a/sound/core/pcm_timer.c +++ b/sound/core/pcm_timer.c @@ -25,6 +25,8 @@ #include <sound/pcm.h> #include <sound/timer.h> +#include "pcm_local.h" + /* * Timer functions */ @@ -33,7 +35,7 @@ void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream) { unsigned long rate, mult, fsize, l, post; struct snd_pcm_runtime *runtime = substream->runtime; - + mult = 1000000000; rate = runtime->rate; if (snd_BUG_ON(!rate)) @@ -65,7 +67,7 @@ void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream) static unsigned long snd_pcm_timer_resolution(struct snd_timer * timer) { struct snd_pcm_substream *substream; - + substream = timer->private_data; return substream->runtime ? substream->runtime->timer_resolution : 0; } @@ -73,7 +75,7 @@ static unsigned long snd_pcm_timer_resolution(struct snd_timer * timer) static int snd_pcm_timer_start(struct snd_timer * timer) { struct snd_pcm_substream *substream; - + substream = snd_timer_chip(timer); substream->timer_running = 1; return 0; @@ -82,7 +84,7 @@ static int snd_pcm_timer_start(struct snd_timer * timer) static int snd_pcm_timer_stop(struct snd_timer * timer) { struct snd_pcm_substream *substream; - + substream = snd_timer_chip(timer); substream->timer_running = 0; return 0; @@ -112,7 +114,7 @@ void snd_pcm_timer_init(struct snd_pcm_substream *substream) { struct snd_timer_id tid; struct snd_timer *timer; - + tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE; tid.dev_class = SNDRV_TIMER_CLASS_PCM; tid.card = substream->pcm->card->number; -- 2.11.0

2 1

[alsa-devel] [PATCH 00/15] make more driver use devm_of_platform_populate()
by Benjamin Gaignard 30 May '17

30 May '17

Number of calls to of_platform_populate() aren't unbalanced by a call to of_platform_depopulate() that could generate issue will loading/unloading the drivers. Make those drivers use devm_of_platform_populate() fix the problem without need to add remove function. In some case replacing of_platform_populate() by devm_of_platform_populate() allow to delete driver remove function and save some lines of code. This series of patches based on v4.12-rc3 tag. CC: Alexandre Torgue <alexandre.torgue(a)st.com> CC: David Airlie <airlied(a)linux.ie> CC: Fabrice Gasnier <fabrice.gasnier(a)st.com> CC: Hartmut Knaack <knaack.h(a)gmx.de> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Javier Martinez Canillas <javier(a)osg.samsung.com> CC: Jonathan Cameron <jic23(a)kernel.org> CC: Krzysztof Kozlowski <krzk(a)kernel.org> CC: Kukjin Kim <kgene(a)kernel.org> CC: Kyungmin Park <kyungmin.park(a)samsung.com> CC: Lars-Peter Clausen <lars(a)metafoo.de> CC: Lee Jones <lee.jones(a)linaro.org> CC: Liam Girdwood <lgirdwood(a)gmail.com> CC: Mark Brown <broonie(a)kernel.org> CC: Mauro Carvalho Chehab <mchehab(a)kernel.org> CC: Olivier Moysan <olivier.moysan(a)st.com> CC: Rob Clark <robdclark(a)gmail.com> CC: Shawn Guo <shawnguo(a)kernel.org> CC: Sylwester Nawrocki <s.nawrocki(a)samsung.com> CC: Takashi Iwai <tiwai(a)suse.com> CC: Tony Lindgren <tony(a)atomide.com> CC: linux-iio(a)vger.kernel.org CC: linux-arm-kernel(a)lists.infradead.org CC: linux-kernel(a)vger.kernel.org CC: dri-devel(a)lists.freedesktop.org CC: linux-arm-msm(a)vger.kernel.org CC: freedreno(a)lists.freedesktop.org CC: linux-samsung-soc(a)vger.kernel.org CC: alsa-devel(a)alsa-project.org CC: linux-media(a)vger.kernel.org Benjamin Gaignard (15): iio: adc: stm32: use devm_of_platform_populate() iio: dac: stm32: use devm_of_platform_populate() drm: zte: use devm_of_platform_populate() drm: msm: use devm_of_platform_populate() mfd: stm32-timers: use devm_of_platform_populate mfd: atmel: use devm_of_platform_populate() mfd: cros_ec: use devm_of_platform_populate() mfd: exynos: use devm_of_platform_populate() mfd: fsl-imx25: use devm_of_platform_populate() mfd: motorola-cpcap: use devm_of_platform_populate() mfd: palmas: use devm_of_platform_populate() mfd: qcom-spmi-pmic: use devm_of_platform_populate() mfd: smsc-ece: use devm_of_platform_populate() sound: stm32: use devm_of_platform_populate() media: exynos4-is: use devm_of_platform_populate() drivers/gpu/drm/msm/msm_drv.c | 10 ++-------- drivers/gpu/drm/zte/zx_drm_drv.c | 2 +- drivers/iio/adc/stm32-adc-core.c | 4 +--- drivers/iio/dac/stm32-dac-core.c | 3 +-- drivers/media/platform/exynos4-is/fimc-is.c | 7 ++----- drivers/mfd/atmel-flexcom.c | 2 +- drivers/mfd/cros_ec.c | 2 +- drivers/mfd/exynos-lpass.c | 2 +- drivers/mfd/fsl-imx25-tsadc.c | 5 +---- drivers/mfd/motorola-cpcap.c | 13 +------------ drivers/mfd/palmas.c | 2 +- drivers/mfd/qcom-spmi-pmic.c | 9 +-------- drivers/mfd/smsc-ece1099.c | 3 +-- drivers/mfd/stm32-timers.c | 10 +--------- sound/soc/stm/stm32_sai.c | 11 +---------- 15 files changed, 17 insertions(+), 68 deletions(-) -- 1.9.1

2 2

[alsa-devel] [PATCH] ALSA: declare snd_kcontrol_new structures as const
by Bhumika Goyal 30 May '17

30 May '17

Declare snd_kcontrol_new structures as const as they are only passed an argument to the function snd_ctl_new1. This argument is of type const, so snd_kcontrol_new structures having this property can be made const. Done using Coccinelle: @r disable optional_qualifier@ identifier x; position p; @@ static struct snd_kcontrol_new x@p={...}; @ok@ identifier r.x; position p; @@ snd_ctl_new1(&x@p,...) @bad@ position p != {r.p,ok.p}; identifier r.x; @@ x@p @depends on !bad disable optional_qualifier@ identifier r.x; @@ +const struct snd_kcontrol_new x; Cross compiled these files: sound/aoa/codecs/tas.c - powerpc sound/mips/{hal2.c/sgio2audio.c} - mips sound/ppc/{awacs.c/beep.c/tumbler.c} - powerpc sound/soc/sh/siu_dai.c - sh Could not find an architecture to compile sound/sh/aica.c. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/aoa/codecs/tas.c | 14 +++++++------- sound/aoa/fabrics/layout.c | 8 ++++---- sound/core/ctljack.c | 2 +- sound/drivers/vx/vx_mixer.c | 24 ++++++++++++------------ sound/firewire/bebob/bebob_maudio.c | 8 ++++---- sound/mips/hal2.c | 4 ++-- sound/mips/sgio2audio.c | 14 +++++++------- sound/pcmcia/vx/vxp_mixer.c | 4 ++-- sound/ppc/awacs.c | 4 ++-- sound/ppc/beep.c | 2 +- sound/ppc/tumbler.c | 8 ++++---- sound/sh/aica.c | 4 ++-- sound/soc/sh/siu_dai.c | 4 ++-- sound/usb/mixer_quirks.c | 2 +- 14 files changed, 51 insertions(+), 51 deletions(-) diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 78ed1ff..733b636 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -271,7 +271,7 @@ static int tas_snd_vol_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new volume_control = { +static const struct snd_kcontrol_new volume_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -314,7 +314,7 @@ static int tas_snd_mute_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new mute_control = { +static const struct snd_kcontrol_new mute_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -426,7 +426,7 @@ static int tas_snd_drc_range_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new drc_range_control = { +static const struct snd_kcontrol_new drc_range_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "DRC Range", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -466,7 +466,7 @@ static int tas_snd_drc_switch_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new drc_switch_control = { +static const struct snd_kcontrol_new drc_switch_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "DRC Range Switch", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -524,7 +524,7 @@ static int tas_snd_capture_source_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new capture_source_control = { +static const struct snd_kcontrol_new capture_source_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* If we name this 'Input Source', it properly shows up in * alsamixer as a selection, * but it's shown under the @@ -586,7 +586,7 @@ static int tas_snd_treble_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new treble_control = { +static const struct snd_kcontrol_new treble_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Treble", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -637,7 +637,7 @@ static int tas_snd_bass_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new bass_control = { +static const struct snd_kcontrol_new bass_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Bass", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index a0c4a5d..1eddf8f 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -707,7 +707,7 @@ static int detect_choice_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new headphone_detect_choice = { +static const struct snd_kcontrol_new headphone_detect_choice = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Headphone Detect Autoswitch", .info = control_info, @@ -717,7 +717,7 @@ static struct snd_kcontrol_new headphone_detect_choice = { .private_value = 0, }; -static struct snd_kcontrol_new lineout_detect_choice = { +static const struct snd_kcontrol_new lineout_detect_choice = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line-Out Detect Autoswitch", .info = control_info, @@ -749,7 +749,7 @@ static int detected_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new headphone_detected = { +static const struct snd_kcontrol_new headphone_detected = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Headphone Detected", .info = control_info, @@ -758,7 +758,7 @@ static struct snd_kcontrol_new headphone_detected = { .private_value = 0, }; -static struct snd_kcontrol_new lineout_detected = { +static const struct snd_kcontrol_new lineout_detected = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line-Out Detected", .info = control_info, diff --git a/sound/core/ctljack.c b/sound/core/ctljack.c index 84a3cd6..0249d5e 100644 --- a/sound/core/ctljack.c +++ b/sound/core/ctljack.c @@ -23,7 +23,7 @@ static int jack_detect_kctl_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new jack_detect_kctl = { +static const struct snd_kcontrol_new jack_detect_kctl = { /* name is filled later */ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .access = SNDRV_CTL_ELEM_ACCESS_READ, diff --git a/sound/drivers/vx/vx_mixer.c b/sound/drivers/vx/vx_mixer.c index be9477e..98a41ac 100644 --- a/sound/drivers/vx/vx_mixer.c +++ b/sound/drivers/vx/vx_mixer.c @@ -455,7 +455,7 @@ static int vx_output_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele return 0; } -static struct snd_kcontrol_new vx_control_output_level = { +static const struct snd_kcontrol_new vx_control_output_level = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), @@ -514,7 +514,7 @@ static int vx_audio_src_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v return 0; } -static struct snd_kcontrol_new vx_control_audio_src = { +static const struct snd_kcontrol_new vx_control_audio_src = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .info = vx_audio_src_info, @@ -558,7 +558,7 @@ static int vx_clock_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ return 0; } -static struct snd_kcontrol_new vx_control_clock_mode = { +static const struct snd_kcontrol_new vx_control_clock_mode = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Clock Mode", .info = vx_clock_mode_info, @@ -717,7 +717,7 @@ static int vx_monitor_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ static const DECLARE_TLV_DB_SCALE(db_scale_audio_gain, -10975, 25, 0); -static struct snd_kcontrol_new vx_control_audio_gain = { +static const struct snd_kcontrol_new vx_control_audio_gain = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), @@ -727,14 +727,14 @@ static struct snd_kcontrol_new vx_control_audio_gain = { .put = vx_audio_gain_put, .tlv = { .p = db_scale_audio_gain }, }; -static struct snd_kcontrol_new vx_control_output_switch = { +static const struct snd_kcontrol_new vx_control_output_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Switch", .info = vx_audio_sw_info, .get = vx_audio_sw_get, .put = vx_audio_sw_put }; -static struct snd_kcontrol_new vx_control_monitor_gain = { +static const struct snd_kcontrol_new vx_control_monitor_gain = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Monitoring Volume", .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | @@ -744,7 +744,7 @@ static struct snd_kcontrol_new vx_control_monitor_gain = { .put = vx_audio_monitor_put, .tlv = { .p = db_scale_audio_gain }, }; -static struct snd_kcontrol_new vx_control_monitor_switch = { +static const struct snd_kcontrol_new vx_control_monitor_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Monitoring Switch", .info = vx_audio_sw_info, /* shared */ @@ -805,7 +805,7 @@ static int vx_iec958_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu return 0; } -static struct snd_kcontrol_new vx_control_iec958_mask = { +static const struct snd_kcontrol_new vx_control_iec958_mask = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK), @@ -813,7 +813,7 @@ static struct snd_kcontrol_new vx_control_iec958_mask = { .get = vx_iec958_mask_get, }; -static struct snd_kcontrol_new vx_control_iec958 = { +static const struct snd_kcontrol_new vx_control_iec958 = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .info = vx_iec958_info, @@ -878,7 +878,7 @@ static int vx_saturation_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ return 0; } -static struct snd_kcontrol_new vx_control_vu_meter = { +static const struct snd_kcontrol_new vx_control_vu_meter = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, /* name will be filled later */ @@ -886,7 +886,7 @@ static struct snd_kcontrol_new vx_control_vu_meter = { .get = vx_vu_meter_get, }; -static struct snd_kcontrol_new vx_control_peak_meter = { +static const struct snd_kcontrol_new vx_control_peak_meter = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, /* name will be filled later */ @@ -894,7 +894,7 @@ static struct snd_kcontrol_new vx_control_peak_meter = { .get = vx_peak_meter_get, }; -static struct snd_kcontrol_new vx_control_saturation = { +static const struct snd_kcontrol_new vx_control_saturation = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Saturation", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index 07e5abd..d10208f 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -396,7 +396,7 @@ static int special_clk_ctl_put(struct snd_kcontrol *kctl, return err; } -static struct snd_kcontrol_new special_clk_ctl = { +static const struct snd_kcontrol_new special_clk_ctl = { .name = "Clock Source", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -429,7 +429,7 @@ static int special_sync_ctl_get(struct snd_kcontrol *kctl, return 0; } -static struct snd_kcontrol_new special_sync_ctl = { +static const struct snd_kcontrol_new special_sync_ctl = { .name = "Sync Status", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READ, @@ -521,7 +521,7 @@ static int special_dig_in_iface_ctl_set(struct snd_kcontrol *kctl, mutex_unlock(&bebob->mutex); return err; } -static struct snd_kcontrol_new special_dig_in_iface_ctl = { +static const struct snd_kcontrol_new special_dig_in_iface_ctl = { .name = "Digital Input Interface", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -577,7 +577,7 @@ static int special_dig_out_iface_ctl_set(struct snd_kcontrol *kctl, mutex_unlock(&bebob->mutex); return err; } -static struct snd_kcontrol_new special_dig_out_iface_ctl = { +static const struct snd_kcontrol_new special_dig_out_iface_ctl = { .name = "Digital Output Interface", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index 00fc924..ba1ba3c 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -264,7 +264,7 @@ static int hal2_gain_put(struct snd_kcontrol *kcontrol, return old != new; } -static struct snd_kcontrol_new hal2_ctrl_headphone = { +static const struct snd_kcontrol_new hal2_ctrl_headphone = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Headphone Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -274,7 +274,7 @@ static struct snd_kcontrol_new hal2_ctrl_headphone = { .put = hal2_gain_put, }; -static struct snd_kcontrol_new hal2_ctrl_mic = { +static const struct snd_kcontrol_new hal2_ctrl_mic = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Mic Capture Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index f07aa39..0ebc1c3 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -230,7 +230,7 @@ static int sgio2audio_source_put(struct snd_kcontrol *kcontrol, } /* dac1/pcm0 mixer control */ -static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 = { +static const struct snd_kcontrol_new sgio2audio_ctrl_pcm0 = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Volume", .index = 0, @@ -242,7 +242,7 @@ static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 = { }; /* dac2/pcm1 mixer control */ -static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 = { +static const struct snd_kcontrol_new sgio2audio_ctrl_pcm1 = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Volume", .index = 1, @@ -254,7 +254,7 @@ static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 = { }; /* record level mixer control */ -static struct snd_kcontrol_new sgio2audio_ctrl_reclevel = { +static const struct snd_kcontrol_new sgio2audio_ctrl_reclevel = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -265,7 +265,7 @@ static struct snd_kcontrol_new sgio2audio_ctrl_reclevel = { }; /* record level source control */ -static struct snd_kcontrol_new sgio2audio_ctrl_recsource = { +static const struct snd_kcontrol_new sgio2audio_ctrl_recsource = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -275,7 +275,7 @@ static struct snd_kcontrol_new sgio2audio_ctrl_recsource = { }; /* line mixer control */ -static struct snd_kcontrol_new sgio2audio_ctrl_line = { +static const struct snd_kcontrol_new sgio2audio_ctrl_line = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line Playback Volume", .index = 0, @@ -287,7 +287,7 @@ static struct snd_kcontrol_new sgio2audio_ctrl_line = { }; /* cd mixer control */ -static struct snd_kcontrol_new sgio2audio_ctrl_cd = { +static const struct snd_kcontrol_new sgio2audio_ctrl_cd = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line Playback Volume", .index = 1, @@ -299,7 +299,7 @@ static struct snd_kcontrol_new sgio2audio_ctrl_cd = { }; /* mic mixer control */ -static struct snd_kcontrol_new sgio2audio_ctrl_mic = { +static const struct snd_kcontrol_new sgio2audio_ctrl_mic = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Mic Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, diff --git a/sound/pcmcia/vx/vxp_mixer.c b/sound/pcmcia/vx/vxp_mixer.c index a4a6642..425315c 100644 --- a/sound/pcmcia/vx/vxp_mixer.c +++ b/sound/pcmcia/vx/vxp_mixer.c @@ -69,7 +69,7 @@ static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v static const DECLARE_TLV_DB_SCALE(db_scale_mic, -21, 3, 0); -static struct snd_kcontrol_new vx_control_mic_level = { +static const struct snd_kcontrol_new vx_control_mic_level = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), @@ -109,7 +109,7 @@ static int vx_mic_boost_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v return 0; } -static struct snd_kcontrol_new vx_control_mic_boost = { +static const struct snd_kcontrol_new vx_control_mic_boost = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Mic Boost", .info = vx_mic_boost_info, diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 1468e4b..d1e4ef1 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -514,7 +514,7 @@ static struct snd_kcontrol_new snd_pmac_awacs_amp_vol[] = { }, }; -static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw = { +static const struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Headphone Playback Switch", .info = snd_pmac_boolean_stereo_info, @@ -523,7 +523,7 @@ static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw = { .private_value = AMP_CH_HD, }; -static struct snd_kcontrol_new snd_pmac_awacs_amp_spk_sw = { +static const struct snd_kcontrol_new snd_pmac_awacs_amp_spk_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Speaker Playback Switch", .info = snd_pmac_boolean_stereo_info, diff --git a/sound/ppc/beep.c b/sound/ppc/beep.c index d3524f9..f19eb3e 100644 --- a/sound/ppc/beep.c +++ b/sound/ppc/beep.c @@ -206,7 +206,7 @@ static int snd_pmac_put_beep(struct snd_kcontrol *kcontrol, return oval != chip->beep->volume; } -static struct snd_kcontrol_new snd_pmac_beep_mixer = { +static const struct snd_kcontrol_new snd_pmac_beep_mixer = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Beep Playback Volume", .info = snd_pmac_info_beep, diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 58ee808..0779a29 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -897,7 +897,7 @@ static struct snd_kcontrol_new snapper_mixers[] = { }, }; -static struct snd_kcontrol_new tumbler_hp_sw = { +static const struct snd_kcontrol_new tumbler_hp_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Headphone Playback Switch", .info = snd_pmac_boolean_mono_info, @@ -905,7 +905,7 @@ static struct snd_kcontrol_new tumbler_hp_sw = { .put = tumbler_put_mute_switch, .private_value = TUMBLER_MUTE_HP, }; -static struct snd_kcontrol_new tumbler_speaker_sw = { +static const struct snd_kcontrol_new tumbler_speaker_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Speaker Playback Switch", .info = snd_pmac_boolean_mono_info, @@ -913,7 +913,7 @@ static struct snd_kcontrol_new tumbler_speaker_sw = { .put = tumbler_put_mute_switch, .private_value = TUMBLER_MUTE_AMP, }; -static struct snd_kcontrol_new tumbler_lineout_sw = { +static const struct snd_kcontrol_new tumbler_lineout_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line Out Playback Switch", .info = snd_pmac_boolean_mono_info, @@ -921,7 +921,7 @@ static struct snd_kcontrol_new tumbler_lineout_sw = { .put = tumbler_put_mute_switch, .private_value = TUMBLER_MUTE_LINE, }; -static struct snd_kcontrol_new tumbler_drc_sw = { +static const struct snd_kcontrol_new tumbler_drc_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "DRC Switch", .info = snd_pmac_boolean_mono_info, diff --git a/sound/sh/aica.c b/sound/sh/aica.c index fbbc252..ab4802d 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -535,7 +535,7 @@ static int aica_pcmvolume_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new snd_aica_pcmswitch_control = { +static const struct snd_kcontrol_new snd_aica_pcmswitch_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Switch", .index = 0, @@ -544,7 +544,7 @@ static struct snd_kcontrol_new snd_aica_pcmswitch_control = { .put = aica_pcmswitch_put }; -static struct snd_kcontrol_new snd_aica_pcmvolume_control = { +static const struct snd_kcontrol_new snd_aica_pcmvolume_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Volume", .index = 0, diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index 76b2ab8..4a22aad 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -441,7 +441,7 @@ static int siu_dai_put_volume(struct snd_kcontrol *kctrl, return 0; } -static struct snd_kcontrol_new playback_controls = { +static const struct snd_kcontrol_new playback_controls = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Volume", .index = 0, @@ -451,7 +451,7 @@ static struct snd_kcontrol_new playback_controls = { .private_value = VOLUME_PLAYBACK, }; -static struct snd_kcontrol_new capture_controls = { +static const struct snd_kcontrol_new capture_controls = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Capture Volume", .index = 0, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 4fa0053..e3d1dec 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -362,7 +362,7 @@ static int snd_audigy2nx_led_resume(struct usb_mixer_elem_list *list) } /* name and private_value are set dynamically */ -static struct snd_kcontrol_new snd_audigy2nx_control = { +static const struct snd_kcontrol_new snd_audigy2nx_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .info = snd_audigy2nx_led_info, .get = snd_audigy2nx_led_get, -- 2.7.4

4 6

[alsa-devel] [PATCH v2 0/3] ALSA: USB-descriptor endianness fixes
by Johan Hovold 30 May '17

30 May '17

These patches add missing endianness conversions when accessing the USB device-descriptor fields. In the process, clean up the us122l driver which had product-id conditionals sprinkled throughout. The final patch enables compile testing for us122l which currently depends on x86 but compiles just fine for arm. Note that these patches have only been compile tested. Johan v2 - enable compile testing rather than drop the us122l x86-dependency Johan Hovold (3): ALSA: usb-audio: fix Amanero Combo384 quirk on big-endian hosts ALSA: us122l: clean up US144 handling ALSA: us122l: enable compile testing sound/usb/Kconfig | 2 +- sound/usb/quirks.c | 2 +- sound/usb/usx2y/us122l.c | 36 ++++++++++++++++++------------------ sound/usb/usx2y/us122l.h | 2 ++ 4 files changed, 22 insertions(+), 20 deletions(-) -- 2.13.0

2 5

[alsa-devel] [PATCH] ALSA: Adjust structure(snd_timer_tread) members to avoid 8 padding bytes
by Manjeet Pawar 30 May '17

30 May '17

This patch is made to fix below CVE issue where 8 padding bytes are sent to user without being initialized. Structure members are adjusted to avoid 8 padding bytes. CVE-2016-4569/CVE-2016-4578: The stack object tread has a total size of 32 bytes. Its field event and val both contain 4 bytes padding. These 8 bytes padding bytes are sent to user without being initialized. Fix patch: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?h=f… These patches does memset on structure object to initialize 8 padding bytes, which can be skipped by adjusting structure members. Signed-off-by: Manjeet Pawar <manjeet.p(a)samsung.com> Signed-off-by: Vaneet Narang <v.narang(a)samsung.com> --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index a45be6b..820c5a0 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -789,8 +789,8 @@ enum { struct snd_timer_tread { int event; - struct timespec tstamp; unsigned int val; + struct timespec tstamp; }; /**************************************************************************** -- 1.9.1

2 1

[alsa-devel] [PATCH - Fix building alsa-plugins against libav-10 1/1] Fix building alsa-plugins against libav-10
by plevine457＠gmail.com 30 May '17

30 May '17

From: Peter Levine <plevine457(a)gmail.com> Signed-off-by: Peter Levine <plevine457(a)gmail.com> diff --git a/Makefile.am b/Makefile.am index 69cfe0d..9195b56 100644 --- a/Makefile.am +++ b/Makefile.am @@ -9,8 +9,14 @@ if HAVE_SAMPLERATE SUBDIRS += rate endif if HAVE_AVCODEC +SUBDIRS += a52 +if !HAVE_AVRESAMPLE SUBDIRS += a52 rate-lavc endif +endif +if HAVE_AVRESAMPLE +SUBDIRS += rate-lavr +endif if HAVE_MAEMO_PLUGIN SUBDIRS += maemo endif diff --git a/configure.ac b/configure.ac index f42601c..0af5ec9 100644 --- a/configure.ac +++ b/configure.ac @@ -127,6 +127,10 @@ if test $HAVE_AVCODEC = yes; then if test -z "$AVCODEC_HEADER"; then HAVE_AVCODEC=no fi + SAVE_LIBS=$LIBS + LIBS="$LIBS $AVCODEC_LIBS" + AC_CHECK_FUNCS([av_resample_init]) + LIBS=$SAVE_LIBS CFLAGS="$CFLAGS_saved" fi @@ -135,6 +139,18 @@ AC_SUBST(AVCODEC_CFLAGS) AC_SUBST(AVCODEC_LIBS) AC_SUBST(AVCODEC_HEADER) +AC_ARG_ENABLE([avresample], + AS_HELP_STRING([--disable-avresample], [Do not build plugins depending on avcodec (lavrate)])) + +if test "x$enable_avresample" != "xno"; then + PKG_CHECK_MODULES(AVRESAMPLE, [libavresample libavutil], [HAVE_AVRESAMPLE=yes], [HAVE_AVRESAMPLE=no]) +fi + +AM_CONDITIONAL(HAVE_AVRESAMPLE, test x$HAVE_AVCODEC = xyes) +AC_SUBST(AVRESAMPLE_CFLAGS) +AC_SUBST(AVRESAMPLE_LIBS) +AC_SUBST(AVRESAMPLE_HEADER) + AC_ARG_ENABLE([speexdsp], AS_HELP_STRING([--disable-speexdsp], [Disable building of speexdsp plugin])) @@ -217,7 +233,7 @@ AC_OUTPUT([ mix/Makefile rate/Makefile a52/Makefile - rate-lavc/Makefile + rate-lavr/Makefile maemo/Makefile doc/Makefile usb_stream/Makefile diff --git a/rate-lavr/Makefile.am b/rate-lavr/Makefile.am new file mode 100644 index 0000000..a1dca35 --- /dev/null +++ b/rate-lavr/Makefile.am @@ -0,0 +1,22 @@ +asound_module_rate_lavr_LTLIBRARIES = libasound_module_rate_lavr.la + +asound_module_rate_lavrdir = @ALSA_PLUGIN_DIR@ + +AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ @AVRESAMPLE_CFLAGS@ +AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED) + +libasound_module_rate_lavr_la_SOURCES = rate_lavr.c +libasound_module_rate_lavr_la_LIBADD = @ALSA_LIBS@ @AVRESAMPLE_LIBS@ + + +install-exec-hook: + rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate*.so + $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate.so + $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_higher.so + $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_high.so + $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_fast.so + $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_faster.so + +uninstall-hook: + rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate*.so + rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavr*.so diff --git a/rate-lavc/rate_lavcrate.c b/rate-lavr/rate_lavr.c similarity index 58% copy from rate-lavc/rate_lavcrate.c copy to rate-lavr/rate_lavr.c index 14a2198..fe3bf4b 100644 --- a/rate-lavc/rate_lavcrate.c +++ b/rate-lavr/rate_lavr.c @@ -1,9 +1,6 @@ /* - * Rate converter plugin using libavcodec's resampler - * Copyright (c) 2007 by Nicholas Kain <njkain(a)gmail.com> - * - * based on rate converter that uses libsamplerate - * Copyright (c) 2006 by Takashi Iwai <tiwai(a)suse.de> + * Rate converter plugin using libavresample + * Copyright (c) 2014 by Anton Khirnov * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public @@ -19,21 +16,23 @@ #include <stdio.h> #include <alsa/asoundlib.h> #include <alsa/pcm_rate.h> -#include AVCODEC_HEADER -#include "gcd.h" + +#include <libavresample/avresample.h> +#include <libavutil/channel_layout.h> +#include <libavutil/opt.h> +#include <libavutil/mathematics.h> +#include <libavutil/samplefmt.h> + static int filter_size = 16; static int phase_shift = 10; /* auto-adjusts */ static double cutoff = 0; /* auto-adjusts */ struct rate_src { - struct AVResampleContext *context; + AVAudioResampleContext *avr; + int in_rate; int out_rate; - int stored; - int point; - int16_t **out; - int16_t **in; unsigned int channels; }; @@ -50,26 +49,7 @@ static snd_pcm_uframes_t output_frames(void *obj, snd_pcm_uframes_t frames) static void pcm_src_free(void *obj) { struct rate_src *rate = obj; - int i; - - if (rate->out) { - for (i=0; i<rate->channels; i++) { - free(rate->out[i]); - } - free(rate->out); - } - if (rate->in) { - for (i=0; i<rate->channels; i++) { - free(rate->in[i]); - } - free(rate->in); - } - rate->out = rate->in = NULL; - - if (rate->context) { - av_resample_close(rate->context); - rate->context = NULL; - } + avresample_free(&rate->avr); } static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info) @@ -77,12 +57,14 @@ static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info) struct rate_src *rate = obj; int i, ir, or; - if (! rate->context || rate->channels != info->channels) { + if (!rate->avr || rate->channels != info->channels) { + int ret; + pcm_src_free(rate); rate->channels = info->channels; ir = rate->in_rate = info->in.rate; or = rate->out_rate = info->out.rate; - i = gcd(or, ir); + i = av_gcd(or, ir); if (or > ir) { phase_shift = or/i; } else { @@ -93,25 +75,27 @@ static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info) if (cutoff < 0.80) cutoff = 0.80; } - rate->context = av_resample_init(info->out.rate, info->in.rate, - filter_size, phase_shift, - (info->out.rate >= info->in.rate ? 0 : 1), cutoff); - if (!rate->context) - return -EINVAL; - } - rate->out = malloc(rate->channels * sizeof(int16_t *)); - rate->in = malloc(rate->channels * sizeof(int16_t *)); - for (i=0; i<rate->channels; i++) { - rate->out[i] = calloc(info->out.period_size * 2, - sizeof(int16_t)); - rate->in[i] = calloc(info->in.period_size * 2, - sizeof(int16_t)); - } - rate->point = info->in.period_size / 2; - if (!rate->out || !rate->in) { - pcm_src_free(rate); - return -ENOMEM; + rate->avr = avresample_alloc_context(); + if (!rate->avr) + return -ENOMEM; + + av_opt_set_int(rate->avr, "in_sample_rate", info->in.rate, 0); + av_opt_set_int(rate->avr, "out_sample_rate", info->out.rate, 0); + av_opt_set_int(rate->avr, "in_sample_format", AV_SAMPLE_FMT_S16, 0); + av_opt_set_int(rate->avr, "out_sample_format", AV_SAMPLE_FMT_S16, 0); + av_opt_set_int(rate->avr, "in_channel_layout", av_get_default_channel_layout(rate->channels), 0); + av_opt_set_int(rate->avr, "out_channel_layout", av_get_default_channel_layout(rate->channels), 0); + + av_opt_set_int(rate->avr, "filter_size", filter_size, 0); + av_opt_set_int(rate->avr, "phase_shift", phase_shift, 0); + av_opt_set_double(rate->avr, "cutoff", cutoff, 0); + + ret = avresample_open(rate->avr); + if (ret < 0) { + avresample_free(&rate->avr); + return -EINVAL; + } } return 0; @@ -129,48 +113,10 @@ static int pcm_src_adjust_pitch(void *obj, snd_pcm_rate_info_t *info) static void pcm_src_reset(void *obj) { struct rate_src *rate = obj; - rate->stored = 0; -} -static void deinterleave(const int16_t *src, int16_t **dst, unsigned int frames, - unsigned int chans, int overflow) -{ - int i, j; - - if (chans == 1) { - memcpy(dst + overflow, src, frames*sizeof(int16_t)); - } else if (chans == 2) { - for (j=overflow; j<(frames + overflow); j++) { - dst[0][j] = *(src++); - dst[1][j] = *(src++); - } - } else { - for (j=overflow; j<(frames + overflow); j++) { - for (i=0; i<chans; i++) { - dst[i][j] = *(src++); - } - } - } -} - -static void reinterleave(int16_t **src, int16_t *dst, unsigned int frames, - unsigned int chans) -{ - int i, j; - - if (chans == 1) { - memcpy(dst, src, frames*sizeof(int16_t)); - } else if (chans == 2) { - for (j=0; j<frames; j++) { - *(dst++) = src[0][j]; - *(dst++) = src[1][j]; - } - } else { - for (j=0; j<frames; j++) { - for (i=0; i<chans; i++) { - *(dst++) = src[i][j]; - } - } + if (rate->avr) { + avresample_close(rate->avr); + avresample_open(rate->avr); } } @@ -179,22 +125,13 @@ static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int { struct rate_src *rate = obj; int consumed = 0, chans=rate->channels, ret=0, i; - int total_in = rate->stored + src_frames, new_stored; - - deinterleave(src, rate->in, src_frames, chans, rate->point); - for (i=0; i<chans; ++i) { - ret = av_resample(rate->context, rate->out[i], - rate->in[i]+rate->point-rate->stored, &consumed, - total_in, dst_frames, i == (chans - 1)); - new_stored = total_in-consumed; - memmove(rate->in[i]+rate->point-new_stored, - rate->in[i]+rate->point-rate->stored+consumed, - new_stored*sizeof(int16_t)); - } - av_resample_compensate(rate->context, - total_in-src_frames>filter_size?0:1, src_frames); - reinterleave(rate->out, dst, ret, chans); - rate->stored = total_in-consumed; + int total_in = avresample_get_delay(rate->avr) + src_frames; + + ret = avresample_convert(rate->avr, &dst, dst_frames * chans * 2, dst_frames, + &src, src_frames * chans * 2, src_frames); + + avresample_set_compensation(rate->avr, + total_in - src_frames > filter_size ? 0 : 1, src_frames); } static void pcm_src_close(void *obj) @@ -212,7 +149,7 @@ static int get_supported_rates(void *obj, unsigned int *rate_min, static void dump(void *obj, snd_output_t *out) { - snd_output_printf(out, "Converter: liblavc\n"); + snd_output_printf(out, "Converter: libavr\n"); } #endif @@ -220,7 +157,6 @@ static snd_pcm_rate_ops_t pcm_src_ops = { .close = pcm_src_close, .init = pcm_src_init, .free = pcm_src_free, - .reset = pcm_src_reset, .adjust_pitch = pcm_src_adjust_pitch, .convert_s16 = pcm_src_convert_s16, .input_frames = input_frames, @@ -248,7 +184,7 @@ int pcm_src_open(unsigned int version, void **objp, snd_pcm_rate_ops_t *ops) return -ENOMEM; *objp = rate; - rate->context = NULL; + rate->avr = NULL; #if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002 if (version == 0x010001) memcpy(ops, &pcm_src_ops, sizeof(snd_pcm_rate_old_ops_t)); -- 2.13.0

2 1

[alsa-devel] [PATCH - Fix building alsa-plugins against libav-10 1/1] Fix building alsa-plugins against libav-10
by plevine457＠gmail.com 30 May '17

30 May '17

From: Peter Levine <plevine457(a)gmail.com> Signed-off-by: Peter Levine <plevine457(a)gmail.com> diff --git a/Makefile.am b/Makefile.am index 69cfe0d..9195b56 100644 --- a/Makefile.am +++ b/Makefile.am @@ -9,8 +9,14 @@ if HAVE_SAMPLERATE SUBDIRS += rate endif if HAVE_AVCODEC +SUBDIRS += a52 +if !HAVE_AVRESAMPLE SUBDIRS += a52 rate-lavc endif +endif +if HAVE_AVRESAMPLE +SUBDIRS += rate-lavr +endif if HAVE_MAEMO_PLUGIN SUBDIRS += maemo endif diff --git a/configure.ac b/configure.ac index f42601c..0af5ec9 100644 --- a/configure.ac +++ b/configure.ac @@ -127,6 +127,10 @@ if test $HAVE_AVCODEC = yes; then if test -z "$AVCODEC_HEADER"; then HAVE_AVCODEC=no fi + SAVE_LIBS=$LIBS + LIBS="$LIBS $AVCODEC_LIBS" + AC_CHECK_FUNCS([av_resample_init]) + LIBS=$SAVE_LIBS CFLAGS="$CFLAGS_saved" fi @@ -135,6 +139,18 @@ AC_SUBST(AVCODEC_CFLAGS) AC_SUBST(AVCODEC_LIBS) AC_SUBST(AVCODEC_HEADER) +AC_ARG_ENABLE([avresample], + AS_HELP_STRING([--disable-avresample], [Do not build plugins depending on avcodec (lavrate)])) + +if test "x$enable_avresample" != "xno"; then + PKG_CHECK_MODULES(AVRESAMPLE, [libavresample libavutil], [HAVE_AVRESAMPLE=yes], [HAVE_AVRESAMPLE=no]) +fi + +AM_CONDITIONAL(HAVE_AVRESAMPLE, test x$HAVE_AVCODEC = xyes) +AC_SUBST(AVRESAMPLE_CFLAGS) +AC_SUBST(AVRESAMPLE_LIBS) +AC_SUBST(AVRESAMPLE_HEADER) + AC_ARG_ENABLE([speexdsp], AS_HELP_STRING([--disable-speexdsp], [Disable building of speexdsp plugin])) @@ -217,7 +233,7 @@ AC_OUTPUT([ mix/Makefile rate/Makefile a52/Makefile - rate-lavc/Makefile + rate-lavr/Makefile maemo/Makefile doc/Makefile usb_stream/Makefile diff --git a/rate-lavr/Makefile.am b/rate-lavr/Makefile.am new file mode 100644 index 0000000..a1dca35 --- /dev/null +++ b/rate-lavr/Makefile.am @@ -0,0 +1,22 @@ +asound_module_rate_lavr_LTLIBRARIES = libasound_module_rate_lavr.la + +asound_module_rate_lavrdir = @ALSA_PLUGIN_DIR@ + +AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ @AVRESAMPLE_CFLAGS@ +AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED) + +libasound_module_rate_lavr_la_SOURCES = rate_lavr.c +libasound_module_rate_lavr_la_LIBADD = @ALSA_LIBS@ @AVRESAMPLE_LIBS@ + + +install-exec-hook: + rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate*.so + $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate.so + $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_higher.so + $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_high.so + $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_fast.so + $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_faster.so + +uninstall-hook: + rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate*.so + rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavr*.so diff --git a/rate-lavr/rate_lavr.c b/rate-lavr/rate_lavr.c new file mode 100644 index 0000000..fe3bf4b --- /dev/null +++ b/rate-lavr/rate_lavr.c @@ -0,0 +1,227 @@ +/* + * Rate converter plugin using libavresample + * Copyright (c) 2014 by Anton Khirnov + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + */ + +#include <stdio.h> +#include <alsa/asoundlib.h> +#include <alsa/pcm_rate.h> + +#include <libavresample/avresample.h> +#include <libavutil/channel_layout.h> +#include <libavutil/opt.h> +#include <libavutil/mathematics.h> +#include <libavutil/samplefmt.h> + + +static int filter_size = 16; +static int phase_shift = 10; /* auto-adjusts */ +static double cutoff = 0; /* auto-adjusts */ + +struct rate_src { + AVAudioResampleContext *avr; + + int in_rate; + int out_rate; + unsigned int channels; +}; + +static snd_pcm_uframes_t input_frames(void *obj, snd_pcm_uframes_t frames) +{ + return frames; +} + +static snd_pcm_uframes_t output_frames(void *obj, snd_pcm_uframes_t frames) +{ + return frames; +} + +static void pcm_src_free(void *obj) +{ + struct rate_src *rate = obj; + avresample_free(&rate->avr); +} + +static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info) +{ + struct rate_src *rate = obj; + int i, ir, or; + + if (!rate->avr || rate->channels != info->channels) { + int ret; + + pcm_src_free(rate); + rate->channels = info->channels; + ir = rate->in_rate = info->in.rate; + or = rate->out_rate = info->out.rate; + i = av_gcd(or, ir); + if (or > ir) { + phase_shift = or/i; + } else { + phase_shift = ir/i; + } + if (cutoff <= 0.0) { + cutoff = 1.0 - 1.0/filter_size; + if (cutoff < 0.80) + cutoff = 0.80; + } + + rate->avr = avresample_alloc_context(); + if (!rate->avr) + return -ENOMEM; + + av_opt_set_int(rate->avr, "in_sample_rate", info->in.rate, 0); + av_opt_set_int(rate->avr, "out_sample_rate", info->out.rate, 0); + av_opt_set_int(rate->avr, "in_sample_format", AV_SAMPLE_FMT_S16, 0); + av_opt_set_int(rate->avr, "out_sample_format", AV_SAMPLE_FMT_S16, 0); + av_opt_set_int(rate->avr, "in_channel_layout", av_get_default_channel_layout(rate->channels), 0); + av_opt_set_int(rate->avr, "out_channel_layout", av_get_default_channel_layout(rate->channels), 0); + + av_opt_set_int(rate->avr, "filter_size", filter_size, 0); + av_opt_set_int(rate->avr, "phase_shift", phase_shift, 0); + av_opt_set_double(rate->avr, "cutoff", cutoff, 0); + + ret = avresample_open(rate->avr); + if (ret < 0) { + avresample_free(&rate->avr); + return -EINVAL; + } + } + + return 0; +} + +static int pcm_src_adjust_pitch(void *obj, snd_pcm_rate_info_t *info) +{ + struct rate_src *rate = obj; + + if (info->out.rate != rate->out_rate || info->in.rate != rate->in_rate) + pcm_src_init(obj, info); + return 0; +} + +static void pcm_src_reset(void *obj) +{ + struct rate_src *rate = obj; + + if (rate->avr) { + avresample_close(rate->avr); + avresample_open(rate->avr); + } +} + +static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int + dst_frames, const int16_t *src, unsigned int src_frames) +{ + struct rate_src *rate = obj; + int consumed = 0, chans=rate->channels, ret=0, i; + int total_in = avresample_get_delay(rate->avr) + src_frames; + + ret = avresample_convert(rate->avr, &dst, dst_frames * chans * 2, dst_frames, + &src, src_frames * chans * 2, src_frames); + + avresample_set_compensation(rate->avr, + total_in - src_frames > filter_size ? 0 : 1, src_frames); +} + +static void pcm_src_close(void *obj) +{ + pcm_src_free(obj); +} + +#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002 +static int get_supported_rates(void *obj, unsigned int *rate_min, + unsigned int *rate_max) +{ + *rate_min = *rate_max = 0; /* both unlimited */ + return 0; +} + +static void dump(void *obj, snd_output_t *out) +{ + snd_output_printf(out, "Converter: libavr\n"); +} +#endif + +static snd_pcm_rate_ops_t pcm_src_ops = { + .close = pcm_src_close, + .init = pcm_src_init, + .free = pcm_src_free, + .adjust_pitch = pcm_src_adjust_pitch, + .convert_s16 = pcm_src_convert_s16, + .input_frames = input_frames, + .output_frames = output_frames, +#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002 + .version = SND_PCM_RATE_PLUGIN_VERSION, + .get_supported_rates = get_supported_rates, + .dump = dump, +#endif +}; + +int pcm_src_open(unsigned int version, void **objp, snd_pcm_rate_ops_t *ops) + +{ + struct rate_src *rate; + +#if SND_PCM_RATE_PLUGIN_VERSION < 0x010002 + if (version != SND_PCM_RATE_PLUGIN_VERSION) { + fprintf(stderr, "Invalid rate plugin version %x\n", version); + return -EINVAL; + } +#endif + rate = calloc(1, sizeof(*rate)); + if (!rate) + return -ENOMEM; + + *objp = rate; + rate->avr = NULL; +#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002 + if (version == 0x010001) + memcpy(ops, &pcm_src_ops, sizeof(snd_pcm_rate_old_ops_t)); + else +#endif + *ops = pcm_src_ops; + return 0; +} + +int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate)(unsigned int version, void **objp, + snd_pcm_rate_ops_t *ops) +{ + return pcm_src_open(version, objp, ops); +} +int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_higher)(unsigned int version, + void **objp, snd_pcm_rate_ops_t *ops) +{ + filter_size = 64; + return pcm_src_open(version, objp, ops); +} +int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_high)(unsigned int version, + void **objp, snd_pcm_rate_ops_t *ops) +{ + filter_size = 32; + return pcm_src_open(version, objp, ops); +} +int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_fast)(unsigned int version, + void **objp, snd_pcm_rate_ops_t *ops) +{ + filter_size = 8; + return pcm_src_open(version, objp, ops); +} +int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_faster)(unsigned int version, + void **objp, snd_pcm_rate_ops_t *ops) +{ + filter_size = 4; + return pcm_src_open(version, objp, ops); +} + + -- 2.13.0

2 1

[alsa-devel] [PATCH] ASoC: zx-i2s: fix harsh noise with 16-bit audio
by Shawn Guo 29 May '17

29 May '17

From: Shawn Guo <shawn.guo(a)linaro.org> The audio parameter setup in zx_i2s_hw_params() works fine with 32-bit samples, but for 16-bit ones, all we can hear is harsh noises. As suggested by vendor driver code, DMA burst size and TS width should always be 32 bits, no matter audio sample is 16-bit or 32-bit. Follow the suggestion above to fix the harsh noise issue seen with 16-bit audio samples. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/zte/zx-i2s.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c index a865f37c2a56..a7f7a56e0a2d 100644 --- a/sound/soc/zte/zx-i2s.c +++ b/sound/soc/zte/zx-i2s.c @@ -226,11 +226,12 @@ static int zx_i2s_hw_params(struct snd_pcm_substream *substream, struct zx_i2s_info *i2s = snd_soc_dai_get_drvdata(socdai); struct snd_dmaengine_dai_dma_data *dma_data; unsigned int lane, ch_num, len, ret = 0; + unsigned int ts_width = 32; unsigned long val; unsigned long chn_cfg; dma_data = snd_soc_dai_get_dma_data(socdai, substream); - dma_data->addr_width = params_width(params) >> 3; + dma_data->addr_width = ts_width >> 3; val = readl_relaxed(i2s->reg_base + ZX_I2S_TIMING_CTRL); val &= ~(ZX_I2S_TIMING_TS_WIDTH_MASK | ZX_I2S_TIMING_DATA_SIZE_MASK | @@ -251,7 +252,7 @@ static int zx_i2s_hw_params(struct snd_pcm_substream *substream, dev_err(socdai->dev, "Unknown data format\n"); return -EINVAL; } - val |= ZX_I2S_TIMING_TS_WIDTH(len) | ZX_I2S_TIMING_DATA_SIZE(len); + val |= ZX_I2S_TIMING_TS_WIDTH(ts_width) | ZX_I2S_TIMING_DATA_SIZE(len); ch_num = params_channels(params); switch (ch_num) { -- 1.9.1

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[alsa-devel] [PATCH 0/3] ASoC: add audio-graph-scu-card
by Kuninori Morimoto 29 May '17

29 May '17

Hi Mark These adds audio-graph-scu-card which is same as simple-scu-card + OF-graph. 1) patch cleanups existing simple-scu-card document Kuninori Morimoto (3): 1) ASoC: simple-scu-card: cleanup documentation 2) ASoC: add audio-graph-scu-card document 3) ASoC: add audio-graph-scu-card support .../bindings/sound/audio-graph-scu-card.txt | 72 ++++ .../devicetree/bindings/sound/simple-scu-card.txt | 65 ++-- sound/soc/generic/Kconfig | 9 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-scu-card.c | 416 +++++++++++++++++++++ 5 files changed, 523 insertions(+), 41 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt create mode 100644 sound/soc/generic/audio-graph-scu-card.c -- 1.9.1

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[alsa-devel] [PATCH 0/4] ASoC: tidyup return method from probe()
by Kuninori Morimoto 29 May '17

29 May '17

Hi Mark As Dan reported, current simple / simple-scu / audio-graph / audio-grach-scu(it is now posting) are using strange return method on probe() These patches fix these up. Kuninori Morimoto (4): ASoC: simple-card: tidyup return method from probe() ASoC: simple-scu-card: tidyup return method from probe() ASoC: audio-graph-card: tidyup return method from probe() ASoC: audio-graph-scu-card: tidyup return method from probe() sound/soc/generic/audio-graph-card.c | 6 ++++-- sound/soc/generic/audio-graph-scu-card.c | 6 ++++-- sound/soc/generic/simple-card.c | 6 ++++-- sound/soc/generic/simple-scu-card.c | 6 ++++-- 4 files changed, 16 insertions(+), 8 deletions(-) -- 1.9.1

3 9

[alsa-devel] [PATCH 00/14] Revised patchset for PCM in-kernel copy support
by Takashi Iwai 26 May '17

26 May '17

Hi, this is a sort of RFC, a revised patchset of the previous two patches for killing set_fs() usages in PCM. Instead of converting to the merged copy_silence ops, this adds a new copy_kernel ops instead. At the same time, copy/silence are changed to receive the position and size in bytes instead of frames. This allows us to simplify the PCM core code. As a result, a good amount of code could be removed from pcm_lib.c. Note that this patchset isn't complete, containing only two driver conversions (nm256 for interleaved and rme9652 for non-interleaved) as a demonstration. Takashi === Takashi Iwai (14): ALSA: pcm: Introduce copy_user, copy_kernel and fill_silence ops ALSA: nm256: Convert to new PCM copy ops ALSA: rme9652: Convert to the new PCM ops ALSA: pcm: Drop the old copy and silence ops ALSA: pcm: Check PCM state by a common helper function ALSA: pcm: Shuffle codes ALSA: pcm: Call directly the common read/write helpers ALSA: pcm: More unification of PCM transfer codes ALSA: pcm: Unify read/write loop ALSA: pcm: Simplify snd_pcm_playback_silence() ALSA: pcm: Direct in-kernel read/write support usb: gadget: u_uac1: Kill set_fs() usage ALSA: pcm: Kill set_fs() in PCM OSS layer ALSA: pcm: Build OSS writev/readv helpers conditionally drivers/usb/gadget/function/u_uac1.c | 7 +- include/sound/pcm.h | 80 ++++- sound/core/oss/io.c | 4 +- sound/core/oss/pcm_oss.c | 81 +---- sound/core/oss/pcm_plugin.h | 6 +- sound/core/pcm_lib.c | 551 +++++++++++++---------------------- sound/pci/nm256/nm256.c | 57 ++-- sound/pci/rme9652/rme9652.c | 71 +++-- sound/soc/soc-pcm.c | 5 +- 9 files changed, 386 insertions(+), 476 deletions(-) -- 2.13.0

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[alsa-devel] Steinberg MIDEX8 driver attempt
by Hedde Bosman 26 May '17

26 May '17

Dear Alsa developers, I've owned a MIDEX8 for some years now and wanted to do something with it in Linux , so I started some analysis and coding. The current result can be found at https://github.com/sgorpi/midex8 it looks a bit like the standard midi class driver, but it requires some periodic (i guess timer) messages on endpoint 0x82. Therefore, I've took quite some inspiration from the sound/usb/midi.c driver. In general, it seems to work now for simple cases, but I'm having a few issues, that might be due to my lack of knowledge on the USB subsystem, or something else. In the (simplified) libusb test in the above repository, I do not seem to have these issues. Issue 1: Not all (interrupt) urbs that are submitted will complete (within 25 ms at least). Wireshark doesn't show the packets being sent out, and the completeion handler never gets called for those urbs, but also no errors are given by usb_submit_urb. This happens mainly with the timer messages, and my current solution is to unlink the urbs that haven't completed within 25ms. However, I feel there is something going wrong there. If I do not send urbs on the midi-in endpoint, I do not see any missing urbs... Do I have to mutex anything for a shared urb queue or so? Issue 2: Sometimes after sending a few midi messages, an empty urb shows up in wireshark. However, in code, there's an if-statement that should allow only messages of length > 0 to be submitted (line 568, if (num_read > 0)). Might this have to do with issue 1? I intend to improve on the driver some more, but if you could give me any advice on the above issues that would be highly appreciated. With kind regards, Hedde

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[alsa-devel] [PATCH 0/2] ALSA: pcm: add local header for module-local functions
by Takashi Sakamoto 26 May '17

26 May '17

Hi, This patchset is a supplement for merged patchset[1]. As I suggested[2], an added helper function is useful to some parts of PCM core module. Although it's better to replace existent code block with it, there's an issue to share it with several objects, because there is no header files for external symbols not exported to kernel space. This patchset adds a header file for the module-local symbols, then adds calls of the function in several parts of the module. [1] [alsa-devel] [PATCH 0/7] ALSA: Fix/improve PCM ack callback http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120886.html [2] [alsa-devel] [PATCH 0/7] ALSA: Fix/improve PCM ack callback http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120910.html Takashi Sakamoto (2): ALSA: pcm: add local header file for snd-pcm module ALSA: pcm: add arrangement for applying appl_ptr include/sound/pcm.h | 32 ----------------------------- sound/core/pcm.c | 2 ++ sound/core/pcm_lib.c | 14 +++++++------ sound/core/pcm_local.h | 54 +++++++++++++++++++++++++++++++++++++++++++++++++ sound/core/pcm_misc.c | 3 +++ sound/core/pcm_native.c | 13 +++++++----- 6 files changed, 75 insertions(+), 43 deletions(-) create mode 100644 sound/core/pcm_local.h -- 2.11.0

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[alsa-devel] [PATCH] pcm: dmix: Workaround for binary incompatibility
by Takashi Iwai 26 May '17

26 May '17

The commit 1a9bd0f04481 ("pcm: direct: Fix for sync issue on xrun recover") introduced a new field "recoveries" in snd_pcm_direct_share_t. Unfortunately this caused two issues: - It changed the size of the struct which is used as the magic key - The struct size differs between 32bit and 64bit due to alignment The former brought the incompatibility with the older alsa-lib, e.g. when you run an app with an older alsa-lib via LD_PRELOAD, it doesn't work any longer. The latter is more serious, it disallows running 32bit apps dmix with 64bit together. As a workaround, put recoveries field to the unused field "s.xfer_align", so that the struct is in an old form. This is a one-time workaround, and we may need to reconsider more about a breakage in future... Reported-and-tested-by: Cheng Sun <chengsun9(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 8 ++++---- src/pcm/pcm_direct.h | 3 +-- 2 files changed, 5 insertions(+), 6 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 89468d3537d0..43702601b0f2 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -615,7 +615,7 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) } return ret; } - direct->shmptr->recoveries++; + direct->shmptr->s.recoveries++; semerr = snd_pcm_direct_semaphore_up(direct, DIRECT_IPC_SEM_CLIENT); if (semerr < 0) { @@ -631,11 +631,11 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) */ int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) { - if (direct->shmptr->recoveries != direct->recoveries) { + if (direct->shmptr->s.recoveries != direct->recoveries) { /* no matter how many xruns we missed - * so don't increment but just update to actual counter */ - direct->recoveries = direct->shmptr->recoveries; + direct->recoveries = direct->shmptr->s.recoveries; pcm->fast_ops->drop(pcm); /* trigger_tstamp update is missing in drop callbacks */ gettimestamp(&direct->trigger_tstamp, pcm->tstamp_type); @@ -1539,7 +1539,7 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm dmix->slave_buffer_size = spcm->buffer_size; dmix->slave_period_size = dmix->shmptr->s.period_size; dmix->slave_boundary = spcm->boundary; - dmix->recoveries = dmix->shmptr->recoveries; + dmix->recoveries = dmix->shmptr->s.recoveries; ret = snd_pcm_mmap(spcm); if (ret < 0) { diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 24e85f096310..582f055a5451 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -66,7 +66,6 @@ typedef struct { char socket_name[256]; /* name of communication socket */ snd_pcm_type_t type; /* PCM type (currently only hw) */ int use_server; - unsigned int recoveries; /* no of executed recoveries on slave*/ struct { unsigned int format; snd_interval_t rate; @@ -95,7 +94,7 @@ typedef struct { unsigned int stop_threshold; unsigned int silence_threshold; unsigned int silence_size; - unsigned int xfer_align; /* not used */ + unsigned int recoveries; /* no of executed recoveries on slave*/ unsigned long long boundary; unsigned int info; unsigned int msbits; -- 2.13.0

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[alsa-devel] [PATCH] ASoC: wm_adsp: Fix typo in algorithm list warning message
by Richard Fitzgerald 26 May '17

26 May '17

The list terminator is 0xbedead but the message warning if it wasn't found was showing that 0xbeadead was expected. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5aff83b..65c059b5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1888,7 +1888,7 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, } if (be32_to_cpu(val) != 0xbedead) - adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbeadead\n", + adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbedead\n", pos + len, be32_to_cpu(val)); alg = kzalloc(len * 2, GFP_KERNEL | GFP_DMA); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: fixup parent_clk_name of AUDIO_CLKOUTx
by Kuninori Morimoto 26 May '17

26 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 25165f79adc7 ("ASoC: rsnd: enable clock-frequency for both 44.1kHz/48kHz") supported both 44.1kHz/48kHz for AUDIO_CLKOUTx, but it didn't care its parent clock name. This patch fixes it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 36bece8..4a72fd7 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -510,7 +510,8 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, rbga = rbgx; adg->rbga_rate_for_441khz = rate / div; ckr |= brg_table[i] << 20; - if (req_441kHz_rate) + if (req_441kHz_rate && + !(adg_mode_flags(adg) & AUDIO_OUT_48)) parent_clk_name = __clk_get_name(clk); } } @@ -525,7 +526,8 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, rbgb = rbgx; adg->rbgb_rate_for_48khz = rate / div; ckr |= brg_table[i] << 16; - if (req_48kHz_rate) + if (req_48kHz_rate && + (adg_mode_flags(adg) & AUDIO_OUT_48)) parent_clk_name = __clk_get_name(clk); } } -- 1.9.1

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