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

[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

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[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

2 1

[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

2 6

[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

2 19

[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

1 0

[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

2 10

[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

1 0

[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

2 1

[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

2 1

[alsa-devel] [PATCH] ASoC: simple-card-utils: remove strict limitation of bit/frame master
by Kuninori Morimoto 26 May '17

26 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current asoc_simple_card_parse_daifmt is keeping backward compatibility for bitmaster/framemaster which didn't use phandle. The keep compatibility, it is checking prefix length, but it is too strict. let's loosen it. Otherwise, OF-graph base sound card which doesn't have prefix can't detect daifmt. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card-utils.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index b5047e2..f5a3e32 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -21,14 +21,13 @@ int asoc_simple_card_parse_daifmt(struct device *dev, { struct device_node *bitclkmaster = NULL; struct device_node *framemaster = NULL; - int prefix_len = prefix ? strlen(prefix) : 0; unsigned int daifmt; daifmt = snd_soc_of_parse_daifmt(node, prefix, &bitclkmaster, &framemaster); daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; - if (prefix_len && !bitclkmaster && !framemaster) { + if (!bitclkmaster && !framemaster) { /* * No dai-link level and master setting was not found from * sound node level, revert back to legacy DT parsing and -- 1.9.1

2 1

[alsa-devel] 1.1.3 bug report: dmix reports inconsistent status
by Cheng Sun 26 May '17

26 May '17

Hi all, The following commit (released in v1.1.3) introduced an issue where dmix reports its state differently when snd_pcm_state is queried, compared to snd_pcm_status. 38a2d2eda880fda9ed5a4a219db498d9c0856d71 (I suspect faf53c197cabeb1eb0fd1f66cd001d1469ebac09 affects dshare similarly, but this is not tested) The sample program attached at the bottom of this email demonstrates this behaviour by printing the following on underrun: try 1: status_state=3, state=4 try 2: status_state=3, state=4 ALSA lib pcm.c:8306:(snd_pcm_recover) underrun occurred Cheers, Cheng --- #include <assert.h> #include <alsa/asoundlib.h> #include <stdint.h> #define FRAMES 100 int32_t buffer[FRAMES][2]; int main(int argc, char *argv[]) { const char *pcmname = argc == 2 ? argv[1] : "default"; snd_pcm_t *pcm; assert( snd_pcm_open(&pcm, pcmname, SND_PCM_STREAM_PLAYBACK, 0) == 0 ); assert( snd_pcm_set_params(pcm, SND_PCM_FORMAT_S32_LE, SND_PCM_ACCESS_RW_INTERLEAVED, 2, 48000, 1, 250) == 0 ); for (;;) { snd_pcm_sframes_t written = snd_pcm_writei(pcm, buffer, FRAMES); if (written <= 0) { snd_pcm_status_t *status; snd_pcm_status_alloca(&status); assert( snd_pcm_status(pcm, status) == 0 ); int status_state, state; status_state = snd_pcm_status_get_state(status); state = snd_pcm_state(pcm); printf("try 1: status_state=%d, state=%d\n", status_state, state); status_state = snd_pcm_status_get_state(status); state = snd_pcm_state(pcm); printf("try 2: status_state=%d, state=%d\n", status_state, state); assert( snd_pcm_recover(pcm,written,0) == 0 ); } } }

1 1

Re: [alsa-devel] 1.1.4 bug report: dmix fails when using both 32-bit and 64-bit clients
by Takashi Iwai 26 May '17

26 May '17

On Thu, 25 May 2017 22:17:45 +0200, Cheng Sun wrote: > > On 25 May 2017 at 19:55, Takashi Iwai <tiwai(a)suse.de> wrote: > > On Thu, 25 May 2017 20:34:23 +0200, > > Cheng Sun wrote: > >> > >> On 25 May 2017 at 19:17, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > On Thu, 25 May 2017 18:51:10 +0200, > >> > Cheng Sun wrote: > >> >> > >> >> On 25 May 2017 at 17:31, Takashi Iwai <tiwai(a)suse.de> wrote: > >> >> > On Thu, 25 May 2017 17:43:44 +0200, > >> >> > Cheng Sun wrote: > >> >> >> > >> >> >> On 25 May 2017 14:49, "Takashi Iwai" <tiwai(a)suse.de> wrote: > >> >> >> > > >> >> >> > On Wed, 24 May 2017 21:13:46 +0200, > >> >> >> > Cheng Sun wrote: > >> >> >> > > > >> >> >> > > Hi all, > >> >> >> > > > >> >> >> > > The following commit (released in v1.1.4) introduced an issue where a > >> >> >> > > client compiled against a 32-bit (x86) compile of alsa-lib cannot use > >> >> >> > > the same dmix as a client compiled against a 64-bit alsa-lib. > >> >> >> > > > >> >> >> > > 1a9bd0f0448106b917ae7f7bedccfcbf6ce84802 > >> >> >> > > > >> >> >> > > The error message is: > >> >> >> > > > >> >> >> > > ALSA lib pcm_dmix.c:1077:(snd_pcm_dmix_open) unable to create IPC > >> >> >> > > shm instance > >> >> >> > > > >> >> >> > > The following forum post (not mine) describes the symptoms of the > >> >> >> > > issue in more detail (although the git commit the poster specified is > >> >> >> > > not correct): > >> >> >> > > > >> >> >> > > > >> >> >> https://forums.gentoo.org/viewtopic-p-8072290.html?sid=c2fbca45102933ddf9ca… > >> >> >> > > >> >> >> > There is a known problem when you mix up two alsa-lib versions because > >> >> >> > of the changes in dmix shmem struct. The solution should upgrade all > >> >> >> > users. > >> >> >> > >> >> >> (Apologies for duplicate email --accidentally sent the first reply to the > >> >> >> wrong address.) > >> >> >> > >> >> >> Arch Linux has upgraded both its 32 and 64 bit packages to v1.1.4, and I'm > >> >> >> still experiencing this problem. > >> >> >> > >> >> >> While bisecting the bug I linked against two compiles of the same alsa-lib > >> >> >> versions each time. > >> >> >> > >> >> >> Let me know if I can provide any other information. > >> >> > > >> >> > OK, at first to see whether it's the issue of the struct size > >> >> > mismatch, check the following patch. If it is, you'll see the error > >> >> > message. To be sure, check both cases: 64bit -> 32bit, and 32bit -> > >> >> > 64bit. > >> >> > >> >> The behaviour is different for the two cases. > >> >> > >> >> 1) If 64bit client is started first, then the 32bit client prints the following: > >> >> > >> >> magic mismatch: shmptr=a15ad4f0, expected=a15ad4ec > >> >> ALSA lib pcm_dmix.c:1077:(snd_pcm_dmix_open) unable to create IPC shm instance > >> >> > >> >> 2) If 32bit client is started first, then the 64bit client only prints: > >> >> > >> >> ALSA lib pcm_dmix.c:1077:(snd_pcm_dmix_open) unable to create IPC shm instance > >> >> > >> >> I've added some further print statements and it turns out that this is > >> >> because the "return err;" on line 115 is triggered. > >> > > >> > Hm, so at least the struct size difference is a problem. > >> > Doese the patch below change the behavior? > >> > >> Your patch did not change the behaviour. > > > > Hmm, then it's 8 bytes alignment. > > > >> However, as an experiment I added "__attribute__((__packed__))" to the > >> struct, which fixed the problem. So this does seem to be caused by gcc > >> padding structs differently when compiling 32-bit vs 64-bit. > >> > >> I don't think "__attribute__((__packed__))" is necessarily the correct > >> solution here though. > > > > OK, that's at least one way to fix. Another way would be to put > > another 4 bytes integer as a padding. Could you give it a try, too? > > > > > > Takashi > > The following patch works. I'm a little worried that future > rearrangement/additions to the struct will cause it to break again > though... Yes, maybe the combination of both is the better way. That is, we should add packed attribute, but OTOH we should avoid the incompatibility again from 1.1.4 at least on x86-64. If a padding + packed doesn't change the struct size on x86-64 from the unpatched struct, we can go with it. Alternatively, we may move the new field into some of unused old fields. Below is an example. This will make it backward compatible with 1.1.3 and older versions. thanks, Takashi -- 8< -- --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -615,7 +615,7 @@ int snd_pcm_direct_slave_recover(snd_pcm } 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 */ 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 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) { --- 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 1

[alsa-devel] [PATCH 0/3] ALSA: Add rewind disable support
by Subhransu S. Prusty 26 May '17

26 May '17

Rewinds can be disabled when data written in ring buffer will never be validated. This allows for new HDaudio SPIB DMA functionality(allow fetch up to the application pointer, no rewind supported). Skylake driver changes using SPIB will be posted once the core changes are merged. Based on discussion on community, .ack callback is extended with new attribute argument, to allow fetch up to the application pointer. Also drop the mmap of status and control if SPIB functionality is reported by driver. There may be some slight performance downside, but it's likely very small. Verified the changes with tinyalsa and alsa-lib, with both mmap and without mmap support and it works fine. Pierre-Louis Bossart (3): ALSA: core: let low-level driver or userspace disable rewinds ALSA: core: modify .ack callback to take arguments for updating appl ptr ALSA: pcm: conditionally avoid mmap of control data include/sound/pcm-indirect.h | 4 ++-- include/sound/pcm.h | 9 +++++++- include/uapi/sound/asound.h | 2 ++ sound/core/pcm_lib.c | 6 ++++-- sound/core/pcm_native.c | 49 ++++++++++++++++++++++++++++++++++++++++++- sound/mips/hal2.c | 14 ++++++++++--- sound/pci/cs46xx/cs46xx_lib.c | 18 ++++++++++++---- sound/pci/emu10k1/emupcm.c | 8 +++++-- sound/pci/rme32.c | 15 ++++++++++--- 9 files changed, 107 insertions(+), 18 deletions(-) -- 1.9.1

5 30

[alsa-devel] Problems with the DP MST audio support patchset
by Pascal Wichmann 26 May '17

26 May '17

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 Hello, I am not sure whether alsa-devel is the right place for this kind of question. If it is not, please tell me a better place for them. I am building my kernel against the drm-tip freedesktop repository which includes the patches from December 2016 (message id <1480992519-108079-1-git-send-email-libin.yang(a)intel.com> and follow-ups on alsa-devel). I am using a ThinkPad X250 and the ThinkPad Ultra Dock with the HDMI output connected to my YAMAHA HTR-2866; this hdmi output is seen as display port and uses DP MST from the dock. In principle, everything is working quite well; however, I frequently recognize two weird behaviours: * First of all, my audio receiver has 5.1 support. When I configure my laptop to send 5.1 audio using alsa or pulseaudio, there is only silence and no audio signal reaches the receiver; however, when I choose 7.1 digital surround (which my receiver actually does not suppo rt natively), everything works well and the two additional speakers are split to the existing ones (I do not know whether alsa/the kernel or the receiver does this, probably it is the receiver). This problem is pretty much only esthetical, as everything works fine if I use the 7.1 "work-around" (Digital Stereo does work also, but obviously does not support the full 5.1 setup) * The other problem -- which is actually very annoying -- is that there frequently is silence though there is audio playing; this happens now and then, but when there has been *no* output for a while, it always happens a few seconds after starting the new output and lasts for two to three seconds. During these periods of silence, the icons on my receiver indicating that there is an audio source are also gone. This problem occurs with 7.1 configuration as well as with Digital Stereo. I am not sure what these problems are caused by and do not know what kin d of debug information you may require to get more hints on how to solve them. Any hints are appreciated. Thank you for your advice, Pascal Wichmann -----BEGIN PGP SIGNATURE----- iQIzBAEBCAAdFiEE/gJ8Tpn0OtJz8hxD6+YIa34W/zYFAljbj5cACgkQ6+YIa34W /zbIVg//QP4A0wUE2MyaEAenN8nUpNV5ASglRTZ0N5hXBjD+B/sxIMkJdtSynlcB eBYL90ZLuFHxdfXsltifeY29D7CaHfyqQCeJUfzV+RgUGrlv0wqk2vA9Dfc+RTIT PBORlyuknqIV14z8vxMGO/1+Po/S8AFplK/QJs9Y+irGAX+CAhBMlVMt8V5rbL+e oD8KRWgPIwuYShhd0f3tmgqdhRn9RUQgBaYOaq1pXdI/nNDHZuRA021zytkkV4rR NGG2zOsgrrXo82ac1AAeUXlNczqUCM7UXNLjpVYUQQotEMZd9d6oL5/Iq/bD7Z8i Z+bzOWymzaaThqarc11UL9ngJr6JsDNXgdDrLgLbkSE1r6Knq66qzOmzZL7qeUDI 1JeCNkmztn82VLovRogQTY8p8Ye7gXwVUPNDocCoeBKgRA6u2VGs+zJljCDOiyT9 cCA+QGMpRMz+KLt82Bt5/vE6TzngUEOZli34wNhf6DaHYo+0j4TpiCZT3roIVlp+ C+tPvDsbt9SfVLKvmoDCKAS8o5wp7+jESAsTEMzssBeN6LYKTWeAJeZHymf17ckb Y3oub8Wk5bEvZh0AO5q7wqdWMLuTc9BSmJ8eJA0jNV+m9sGKxaTB/qFMfNoJjfFD yvBBeYvic0btTtOiQAPEia1igNR9vt1hpj5wCI6xNQwSekM1bOQ= =/6Dp -----END PGP SIGNATURE-----

4 13

[alsa-devel] [PATCH 0/7] ALSA: Fix/improve PCM ack callback
by Takashi Iwai 25 May '17

25 May '17

Hi, this is a series of patches to fix the potential bugs in PCM ack callback using the pcm-indirect helper functions, and also to enhance the ack callback to be called properly in all appl_ptr updates, as similarly done by Pierre and Subhransu's patches. The changes in the pcm-indirect helper code and its users are fairly trivial, just passing the error code back. ARM/R-Pi people are Cc'ed because of bcm2835-audio driver. Takashi === Takashi Iwai (7): ALSA: pcm: Fix negative appl_ptr handling in pcm-indirect helpers ALSA: mips: Deliver indirect-PCM transfer error ALSA: cs46xx: Deliver indirect-PCM transfer error ALSA: emu10k1: Deliver indirect-PCM transfer error ALSA: rme32: Deliver indirect-PCM transfer error staging: bcm2835-audio: Deliver indirect-PCM transfer error ALSA: pcm: Call ack() whenever appl_ptr is updated .../vc04_services/bcm2835-audio/bcm2835-pcm.c | 5 +-- include/sound/pcm-indirect.h | 10 ++++- sound/core/pcm_native.c | 46 +++++++++++++++++----- sound/mips/hal2.c | 14 +++---- sound/pci/cs46xx/cs46xx_lib.c | 8 ++-- sound/pci/emu10k1/emupcm.c | 4 +- sound/pci/rme32.c | 10 ++--- 7 files changed, 63 insertions(+), 34 deletions(-) -- 2.13.0

3 14

[alsa-devel] 1.1.4 bug report: dmix fails when using both 32-bit and 64-bit clients
by Cheng Sun 25 May '17

25 May '17

Hi all, The following commit (released in v1.1.4) introduced an issue where a client compiled against a 32-bit (x86) compile of alsa-lib cannot use the same dmix as a client compiled against a 64-bit alsa-lib. 1a9bd0f0448106b917ae7f7bedccfcbf6ce84802 The error message is: ALSA lib pcm_dmix.c:1077:(snd_pcm_dmix_open) unable to create IPC shm instance The following forum post (not mine) describes the symptoms of the issue in more detail (although the git commit the poster specified is not correct): https://forums.gentoo.org/viewtopic-p-8072290.html?sid=c2fbca45102933ddf9ca… Cheers, Cheng

2 7

[alsa-devel] [PATCH RFC 00/11] ALSA: pcm: introduce copy_frame operation and SND_PCM_PROXY_DRIVER_SUPPORT
by Takashi Sakamoto 25 May '17

25 May '17

Hi, This RFC patchset is my concept work of another solution against a series of issues in ALSA PCM core, for which Iwai Takashi work in his hasty proposals[2][3]. Included changes are also available in my personal repository on github.com[4]. The aim of this work is to declare existent issues and to propose solutions as moderated as possible. Code refactoring is done to assist it. Patch 01-05: code refactoring phase, with an alias of function prototype. Patch 06-09: driver API changing phase. The .copy_frames is introduced. Patch 10-11: integration phase for proxy drivers, to get new configuration. I need to do this work with a little time (half a day), thus changes are as minimal as possible, especially two drivers are just modified even if drivers in below list should be changed: * drivers/media/pci/solo6x10/solo6x10-g723.c * sound/drivers/dummy.c * sound/isa/gus/gus_pcm.c * sound/isa/sb/emu8000_pcm.c * sound/pci/es1938.c * sound/pci/korg1212/korg1212.c * sound/pci/nm256/nm256.c * sound/pci/rme32.c * sound/pci/rme96.c * sound/pci/rme9652/hdsp.c * sound/pci/rme9652/rme9652.c * sound/sh/sh_dac_audio.c * sound/soc/blackfin/bf5xx-ac97-pcm.c * sound/soc/blackfin/bf5xx-i2s-pcm.c Furthermore, I apologize that this is untested. I wish you to check the concept. [1] [alsa-devel] [PATCH RFC 00/26] Kill set_fs() in ALSA codes http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120542 [2] [alsa-devel] [PATCH 00/16] ALSA: Convert to new copy_silence PCM ops http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120895.html [3] [alsa-devel] [PATCH 0/4] ALSA: Extend PCM buffer-copy helpers http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120957.html [4] takaswie/sound https://github.com/takaswie/sound/tree/topic/add-pcm-frame-ops Takashi Sakamoto (11): ALSA: pcm: introduce an alias of prototype to copy PCM frames ALSA: pcm: use new function alias instead of local one ALSA: pcm: refactoring silence operation ALSA: pcm: add alternative calls ALSA: core: remove duplicated codes to handle PCM frames ALSA: pcm: add new operation; copy_frames ALSA: rme9652: a sample for drivers which support interleaved buffer ALSA: gus: a sample for drivers which support non-interleaved buffer ALSA: pcm: remove copy and silence callbacks ALSA: pcm: add client_space parameter to runtime of PCM substream for PCM proxy drivers ALSA: pcm: add new configuration for PCM proxy drivers drivers/usb/gadget/Kconfig | 3 +- include/sound/pcm.h | 13 +- sound/core/Kconfig | 15 +- sound/core/pcm_lib.c | 448 +++++++++++++++++++++++++++++---------------- sound/core/pcm_native.c | 10 + sound/isa/gus/gus_pcm.c | 100 +++++----- sound/pci/rme9652/hdsp.c | 59 +++--- 7 files changed, 398 insertions(+), 250 deletions(-) -- 2.11.0

2 15

[alsa-devel] [PATCH] ALSA: sb: remove needless evaluation in implementation for copy callback
by Takashi Sakamoto 25 May '17

25 May '17

In design of ALSA pcm core, 'struct snd_pcm_ops.copy' is expected to copy PCM frames, according to frame alignment on intermediate buffer for userspace and dedicated buffer for data transmission. In this callback, value of 'channel' argument depends on the frame alignment, which drivers registers to runtime of PCM substream. When target devices can handle non-interleaved buffer, this value has positive value, otherwise negative. ALSA driver for PCM component of EMU8000 chip is programmed with local macro to switch the frame alignment. The 'copy' operation in non-interleaved side has evaluation of the 'channel' argument (actually it's 'voice' argument). This is useless. This commit remove the evaluation. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/isa/sb/emu8000_pcm.c | 28 ++++------------------------ 1 file changed, 4 insertions(+), 24 deletions(-) diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index 32f234f494e5..bab796d75dbd 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -450,20 +450,8 @@ static int emu8k_pcm_copy(struct snd_pcm_substream *subs, struct snd_emu8000 *emu = rec->emu; snd_emu8000_write_wait(emu, 1); - if (voice == -1) { - unsigned short *buf = src; - int i, err; - count /= rec->voices; - for (i = 0; i < rec->voices; i++) { - err = emu8k_transfer_block(emu, pos + rec->loop_start[i], buf, count); - if (err < 0) - return err; - buf += count; - } - return 0; - } else { - return emu8k_transfer_block(emu, pos + rec->loop_start[voice], src, count); - } + return emu8k_transfer_block(emu, pos + rec->loop_start[voice], src, + count); } /* make a channel block silence */ @@ -487,17 +475,9 @@ static int emu8k_pcm_silence(struct snd_pcm_substream *subs, struct snd_emu8000 *emu = rec->emu; snd_emu8000_write_wait(emu, 1); - if (voice == -1 && rec->voices == 1) + if (rec->voices == 1) voice = 0; - if (voice == -1) { - int err; - err = emu8k_silence_block(emu, pos + rec->loop_start[0], count / 2); - if (err < 0) - return err; - return emu8k_silence_block(emu, pos + rec->loop_start[1], count / 2); - } else { - return emu8k_silence_block(emu, pos + rec->loop_start[voice], count); - } + return emu8k_silence_block(emu, pos + rec->loop_start[voice], count); } #else /* interleave */ -- 2.11.0

2 4

[alsa-devel] [alsa-lib][PATCH] test: add a test for list operation to user-defined element sets
by Takashi Sakamoto 25 May '17

25 May '17

Current implementation of test for user-defined element doesn't perform list operation. This commit adds easy test for the operation. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- test/user-ctl-element-set.c | 80 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 80 insertions(+) diff --git a/test/user-ctl-element-set.c b/test/user-ctl-element-set.c index 75083d21..8635c156 100644 --- a/test/user-ctl-element-set.c +++ b/test/user-ctl-element-set.c @@ -433,6 +433,78 @@ static int check_event(struct elem_set_trial *trial, unsigned int mask, return 0; } +static int check_elem_list(struct elem_set_trial *trial) +{ + snd_ctl_elem_list_t *list; + snd_ctl_elem_id_t *id; + int e; + unsigned int i; + int err; + + snd_ctl_elem_list_alloca(&list); + snd_ctl_elem_id_alloca(&id); + + err = snd_ctl_elem_list(trial->handle, list); + if (err < 0) + return err; + + /* Certainly some elements are already added. */ + if (snd_ctl_elem_list_get_count(list) == 0) + return -EIO; + + err = snd_ctl_elem_list_alloc_space(list, + snd_ctl_elem_list_get_count(list)); + if (err < 0) + return err; + + err = snd_ctl_elem_list(trial->handle, list); + if (err < 0) + goto end; + + if (trial->element_count > snd_ctl_elem_list_get_count(list)) { + err = -EIO; + goto end; + } + + i = 0; + for (e = 0; e < snd_ctl_elem_list_get_count(list); ++e) { + snd_ctl_elem_list_get_id(list, e, id); + + if (strcmp(snd_ctl_elem_id_get_name(id), + snd_ctl_elem_id_get_name(trial->id)) != 0) + continue; + if (snd_ctl_elem_id_get_interface(id) != + snd_ctl_elem_id_get_interface(trial->id)) + continue; + if (snd_ctl_elem_id_get_device(id) != + snd_ctl_elem_id_get_device(trial->id)) + continue; + if (snd_ctl_elem_id_get_subdevice(id) != + snd_ctl_elem_id_get_subdevice(trial->id)) + continue; + + /* + * Here, I expect the list includes element ID data in numerical + * order. Actually, it does. + */ + if (snd_ctl_elem_id_get_numid(id) != + snd_ctl_elem_id_get_numid(trial->id) + i) + continue; + if (snd_ctl_elem_id_get_index(id) != + snd_ctl_elem_id_get_index(trial->id) + i) + continue; + + ++i; + } + + if (i != trial->element_count) + err = -EIO; +end: + snd_ctl_elem_list_free_space(list); + + return err; +} + static int check_elem_set_props(struct elem_set_trial *trial) { snd_ctl_elem_id_t *id; @@ -701,6 +773,14 @@ int main(void) break; } + /* Check added elements are retrieved in a list. */ + err = check_elem_list(&trial); + if (err < 0) { + printf("Fail to list each element with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + /* Check properties of each element in this element set. */ err = check_elem_set_props(&trial); if (err < 0) { -- 2.11.0

2 1

[alsa-devel] [asoc:topic/dai-id 1/1] sound/soc/soc-core.c:4054:9: error: implicit declaration of function 'of_graph_get_port_parent'
by kbuild test robot 25 May '17

25 May '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/dai-id head: a180e8b988437b3e84a1b501ac4d073467602ca6 commit: a180e8b988437b3e84a1b501ac4d073467602ca6 [1/1] ASoC: add snd_soc_get_dai_id() function config: x86_64-randconfig-x010-201721 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout a180e8b988437b3e84a1b501ac4d073467602ca6 # save the attached .config to linux build tree make ARCH=x86_64 All error/warnings (new ones prefixed by >>): sound/soc/soc-core.c: In function 'snd_soc_get_dai_id': >> sound/soc/soc-core.c:4054:9: error: implicit declaration of function 'of_graph_get_port_parent' [-Werror=implicit-function-declaration] node = of_graph_get_port_parent(ep); ^~~~~~~~~~~~~~~~~~~~~~~~ >> sound/soc/soc-core.c:4054:7: warning: assignment makes pointer from integer without a cast [-Wint-conversion] node = of_graph_get_port_parent(ep); ^ cc1: some warnings being treated as errors vim +/of_graph_get_port_parent +4054 sound/soc/soc-core.c 4048 int snd_soc_get_dai_id(struct device_node *ep) 4049 { 4050 struct snd_soc_component *pos; 4051 struct device_node *node; 4052 int ret; 4053 > 4054 node = of_graph_get_port_parent(ep); 4055 4056 /* 4057 * For example HDMI case, HDMI has video/sound port, --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Fix type warning in sprintf
by Richard Fitzgerald 24 May '17

24 May '17

The shift member of struct soc_mixer_control is unsigned int. 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 20695b6..a7dc760 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2654,7 +2654,7 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; char preload[32]; - snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", mc->shift); + snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift); dsp->preloaded = ucontrol->value.integer.value[0]; -- 1.9.1

2 3

[alsa-devel] [PATCH] ASoC: Fix use-after-free at card unregistration
by Takashi Iwai 24 May '17

24 May '17

soc_cleanup_card_resources() call snd_card_free() at the last of its procedure. This turned out to lead to a use-after-free. PCM runtimes have been already removed via soc_remove_pcm_runtimes(), while it's dereferenced later in soc_pcm_free() called via snd_card_free(). The fix is simple: just move the snd_card_free() call to the beginning of the whole procedure. This also gives another benefit: it guarantees that all operations have been shut down before actually releasing the resources, which was racy until now. Reported-and-tested-by: Robert Jarzmik <robert.jarzmik(a)free.fr> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index aae099c0e502..754e3ef8d7ae 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2286,6 +2286,9 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) list_for_each_entry(rtd, &card->rtd_list, list) flush_delayed_work(&rtd->delayed_work); + /* free the ALSA card at first; this syncs with pending operations */ + snd_card_free(card->snd_card); + /* remove and free each DAI */ soc_remove_dai_links(card); soc_remove_pcm_runtimes(card); @@ -2300,9 +2303,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) if (card->remove) card->remove(card); - snd_card_free(card->snd_card); return 0; - } /* removes a socdev */ -- 2.13.0

2 1

[alsa-devel] Applied "MAINTAINERS: Update email address for patches to Wolfson parts" to the asoc tree
by Mark Brown 24 May '17

24 May '17

The patch MAINTAINERS: Update email address for patches to Wolfson parts has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3a1672bb0186e940fc8521d19069f57690e850c2 Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Date: Wed, 24 May 2017 11:01:32 +0100 Subject: [PATCH] MAINTAINERS: Update email address for patches to Wolfson parts A somewhat overdue update of the address for sending patches on Wolfson parts to since our acquision a couple of years ago by Cirrus Logic. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index f7d568b8f133..517829e3bd1d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13858,7 +13858,7 @@ S: Odd fixes F: drivers/net/wireless/wl3501* WOLFSON MICROELECTRONICS DRIVERS -L: patches(a)opensource.wolfsonmicro.com +L: patches(a)opensource.cirrus.com T: git https://github.com/CirrusLogic/linux-drivers.git W: https://github.com/CirrusLogic/linux-drivers/wiki S: Supported -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: rsnd: SSI PIO adjust to 24bit mode
by Kuninori Morimoto 24 May '17

24 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 90431eb49bff ("ASoC: rsnd: don't use PDTA bit for 24bit on SSI") fixups 24bit mode data alignment, but PIO was not cared. This patch fixes PIO mode 24bit data alignment Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Mark, this is for mark/fix/rcar branch, not mark/topic/rcar sound/soc/sh/rcar/ssi.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 57daac0..0822ae0b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -567,6 +567,13 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); u32 *buf = (u32 *)(runtime->dma_area + rsnd_dai_pointer_offset(io, 0)); + int shift = 0; + + switch (runtime->sample_bits) { + case 32: + shift = 8; + break; + } /* * 8/16/32 data can be assesse to TDR/RDR register @@ -574,9 +581,9 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, * see rsnd_ssi_init() */ if (rsnd_io_is_play(io)) - rsnd_mod_write(mod, SSITDR, *buf); + rsnd_mod_write(mod, SSITDR, (*buf) << shift); else - *buf = rsnd_mod_read(mod, SSIRDR); + *buf = (rsnd_mod_read(mod, SSIRDR) >> shift); elapsed = rsnd_dai_pointer_update(io, sizeof(*buf)); } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: cht_bsw_max98090_ti: Remove unused function cht_get_codec_dai()
by Matthias Kaehlcke 24 May '17

24 May '17

Looks like the function has never been used since it was added by commit 17119a465706 ("ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_max98090_ti"). Removing it fixes the following warning when building with clang: sound/soc/intel/boards/cht_bsw_max98090_ti.c:42:35: error: unused function 'cht_get_codec_dai' [-Werror,-Wunused-function] Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 742bc0d4e681..20755ecc7f9e 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -39,18 +39,6 @@ struct cht_mc_private { bool ts3a227e_present; }; -static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - - list_for_each_entry(rtd, &card->rtd_list, list) { - if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, - strlen(CHT_CODEC_DAI))) - return rtd->codec_dai; - } - return NULL; -} - static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), -- 2.13.0.219.gdb65acc882-goog

3 4

[alsa-devel] [PATCH] ASoC: cs4271: Remove unnecessary additional variable definition
by Charles Keepax 24 May '17

24 May '17

The function already defines a ret variable at the top and makes no particular use of the shadowed definition, as such remove the redundant definition. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4271.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index e78b5f0..d882477 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -674,8 +674,6 @@ static int cs4271_common_probe(struct device *dev, cs4271->gpio_nreset = cs4271plat->gpio_nreset; if (gpio_is_valid(cs4271->gpio_nreset)) { - int ret; - ret = devm_gpio_request(dev, cs4271->gpio_nreset, "CS4271 Reset"); if (ret < 0) -- 2.1.4

3 2

[alsa-devel] [PATCH 1/3] ASoC: Intel: Skylake: Fix IPC rx_list corruption
by Subhransu S. Prusty 24 May '17

24 May '17

From: Pardha Saradhi K <pardha.saradhi.kesapragada(a)intel.com> In SKL+ platforms, all IPC commands are serialised, i.e. the driver sends a new IPC to DSP, only after receiving a reply from the firmware for the current IPC. Hence it seems apparent that there is only a single modifier of the IPC RX List. However, during an IPC timeout case in a multithreaded environment, there is a possibility of the list element being deleted two times if not properly protected. So, use spin lock save/restore to prevent rx_list corruption. Signed-off-by: Pardha Saradhi K <pardha.saradhi.kesapragada(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- sound/soc/intel/skylake/skl-sst-ipc.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 58c525096a7c..498b15345b1a 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -413,8 +413,11 @@ static void skl_ipc_process_reply(struct sst_generic_ipc *ipc, u32 reply = header.primary & IPC_GLB_REPLY_STATUS_MASK; u64 *ipc_header = (u64 *)(&header); struct skl_sst *skl = container_of(ipc, struct skl_sst, ipc); + unsigned long flags; + spin_lock_irqsave(&ipc->dsp->spinlock, flags); msg = skl_ipc_reply_get_msg(ipc, *ipc_header); + spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); if (msg == NULL) { dev_dbg(ipc->dev, "ipc: rx list is empty\n"); return; @@ -456,8 +459,10 @@ static void skl_ipc_process_reply(struct sst_generic_ipc *ipc, } } + spin_lock_irqsave(&ipc->dsp->spinlock, flags); list_del(&msg->list); sst_ipc_tx_msg_reply_complete(ipc, msg); + spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); } irqreturn_t skl_dsp_irq_thread_handler(int irq, void *context) -- 1.9.1

3 6

[alsa-devel] [PATCH v3 0/7] ASoC: Intel: kabylake: Adds rt5663+max98927 machine driver
by Naveen M 24 May '17

24 May '17

Add support for multiple codecs in acpi matching of sst library using machine quirk for robust mach driver selection. Add kabylake machine driver & corresponding driver entry in machine table for following codecs: 1. rt5663 for headset configured to ssp1. 2. max98927 for speakers configured to ssp0. Changes in V2: Convert all machine data to C99 style. Changes in V3: Split machine data for respective drivers Naveen M (7): ASoC: Intel: Convert atom machine data to C99 style ASoC: Intel: Convert skl machine data to C99 style ASoC: Intel: Create a helper to search for matching machine ASoC: Move quirk to identify correct machine driver ASoC: Improve machine driver selection based on quirk data ASoC: Intel: Add Kabylake Realtek Maxim machine driver ASoC: Intel: Add Kabylake RT5663+MAX98927 machine driver entry sound/soc/intel/Kconfig | 15 + sound/soc/intel/atom/sst/sst_acpi.c | 211 ++++++-- sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 687 +++++++++++++++++++++++++++ sound/soc/intel/common/sst-acpi.h | 23 + sound/soc/intel/common/sst-match-acpi.c | 47 +- sound/soc/intel/skylake/skl.c | 84 +++- 7 files changed, 1003 insertions(+), 66 deletions(-) create mode 100644 sound/soc/intel/boards/kbl_rt5663_max98927.c -- 1.9.1

3 10

[alsa-devel] [PATCH 0/6] ASoC: Intel: kabylake: Adds rt5663+max98927 machine driver
by Naveen M 24 May '17

24 May '17

Add support for multiple codecs in acpi matching of sst library using machine quirk for robust mach driver selection. Add kabylake machine driver & corresponding driver entry in machine table for following codecs: 1. rt5663 for headset configured to ssp1. 2. max98927 for speakers configured to ssp0. Naveen M (6): ASoC: Intel: Create a helper to search for matching machine ASoC: Move quirk to identify correct machine driver ASoC: Add quirk data to support multiple codecs ASoC: Improve machine driver selection based on quirk data ASoC: Intel: Add Kabylake Realtek Maxim machine driver ASoC: Intel: Add Kabylake RT5663+MAX98927 machine driver entry sound/soc/intel/Kconfig | 15 + sound/soc/intel/atom/sst/sst_acpi.c | 48 +- sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 687 +++++++++++++++++++++++++++ sound/soc/intel/common/sst-acpi.h | 23 + sound/soc/intel/common/sst-match-acpi.c | 47 +- sound/soc/intel/skylake/skl.c | 41 +- 7 files changed, 821 insertions(+), 42 deletions(-) create mode 100644 sound/soc/intel/boards/kbl_rt5663_max98927.c -- 1.9.1

4 11

[alsa-devel] [PATCH v2 0/6] ASoC: Intel: kabylake: Adds rt5663+max98927 machine driver
by Naveen M 24 May '17

24 May '17

Convert all machine data to C99 style initialization. Add support for multiple codecs in acpi matching of sst library using machine quirk for robust mach driver selection. Add kabylake machine driver & corresponding driver entry in machine table for following codecs: 1. rt5663 for headset configured to ssp1. 2. max98927 for speakers configured to ssp0. Changes in V2: Convert all machine data to C99 style. Naveen M (6): ASoC: Intel: Convert all machine data to C99 style ASoC: Intel: Create a helper to search for matching machine ASoC: Move quirk to identify correct machine driver ASoC: Improve machine driver selection based on quirk data ASoC: Intel: Add Kabylake Realtek Maxim machine driver ASoC: Intel: Add Kabylake RT5663+MAX98927 machine driver entry sound/soc/intel/Kconfig | 15 + sound/soc/intel/atom/sst/sst_acpi.c | 211 ++++++-- sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 687 +++++++++++++++++++++++++++ sound/soc/intel/common/sst-acpi.h | 23 + sound/soc/intel/common/sst-match-acpi.c | 47 +- sound/soc/intel/skylake/skl.c | 84 +++- 7 files changed, 1003 insertions(+), 66 deletions(-) create mode 100644 sound/soc/intel/boards/kbl_rt5663_max98927.c -- 1.9.1

3 10

[alsa-devel] [PATCH] ASoC: simple-card: Fix misleading error message
by Julian Scheel 24 May '17

24 May '17

In case cpu could not be found the error message would always refer to /codec/ not being found in DT. Fix this by catching the cpu node not found case explicitly. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- sound/soc/generic/simple-card.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7ed84f811205..38d7b601546f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -233,13 +233,19 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, snprintf(prop, sizeof(prop), "%scpu", prefix); cpu = of_get_child_by_name(node, prop); + if (!cpu) { + ret = -EINVAL; + dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); + goto dai_link_of_err; + } + snprintf(prop, sizeof(prop), "%splat", prefix); plat = of_get_child_by_name(node, prop); snprintf(prop, sizeof(prop), "%scodec", prefix); codec = of_get_child_by_name(node, prop); - if (!cpu || !codec) { + if (!codec) { ret = -EINVAL; dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); goto dai_link_of_err; -- 2.12.2

2 1

[alsa-devel] Applied "ASoC: sun8i-codec-analog: split out mbias" to the asoc tree
by Mark Brown 24 May '17

24 May '17

The patch ASoC: sun8i-codec-analog: split out mbias has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b08a20f58d2efcd88bf5276e34cd4020028accb7 Mon Sep 17 00:00:00 2001 From: Icenowy Zheng <icenowy(a)aosc.xyz> Date: Wed, 24 May 2017 18:05:59 +0800 Subject: [PATCH] ASoC: sun8i-codec-analog: split out mbias Allwinner V3s features an analog codec without MBIAS pin. Split out this part, in order to prepare for the V3s analog codec. Signed-off-by: Icenowy Zheng <icenowy(a)aosc.xyz> Reviewed-by: Chen-Yu Tsai <wens(a)csie.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/sun8i-codec-analog.c | 35 ++++++++++++++++++++++++++++++----- 1 file changed, 30 insertions(+), 5 deletions(-) diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c index 6c17c99c2c8d..edcc3eb7cd9a 100644 --- a/sound/soc/sunxi/sun8i-codec-analog.c +++ b/sound/soc/sunxi/sun8i-codec-analog.c @@ -289,11 +289,6 @@ static const struct snd_soc_dapm_widget sun8i_codec_common_widgets[] = { /* Microphone input */ SND_SOC_DAPM_INPUT("MIC1"), - /* Microphone Bias */ - SND_SOC_DAPM_SUPPLY("MBIAS", SUN8I_ADDA_MIC1G_MICBIAS_CTRL, - SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MMICBIASEN, - 0, NULL, 0), - /* Mic input path */ SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN8I_ADDA_MIC1G_MICBIAS_CTRL, SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1AMPEN, 0, NULL, 0), @@ -453,6 +448,27 @@ static int sun8i_codec_add_headphone(struct snd_soc_component *cmpnt) return 0; } +/* mbias specific widget */ +static const struct snd_soc_dapm_widget sun8i_codec_mbias_widgets[] = { + SND_SOC_DAPM_SUPPLY("MBIAS", SUN8I_ADDA_MIC1G_MICBIAS_CTRL, + SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MMICBIASEN, + 0, NULL, 0), +}; + +static int sun8i_codec_add_mbias(struct snd_soc_component *cmpnt) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt); + struct device *dev = cmpnt->dev; + int ret; + + ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_mbias_widgets, + ARRAY_SIZE(sun8i_codec_mbias_widgets)); + if (ret) + dev_err(dev, "Failed to add MBIAS DAPM widgets: %d\n", ret); + + return ret; +} + /* hmic specific widget */ static const struct snd_soc_dapm_widget sun8i_codec_hmic_widgets[] = { SND_SOC_DAPM_SUPPLY("HBIAS", SUN8I_ADDA_MIC1G_MICBIAS_CTRL, @@ -679,6 +695,7 @@ struct sun8i_codec_analog_quirks { bool has_hmic; bool has_linein; bool has_lineout; + bool has_mbias; bool has_mic2; }; @@ -686,12 +703,14 @@ static const struct sun8i_codec_analog_quirks sun8i_a23_quirks = { .has_headphone = true, .has_hmic = true, .has_linein = true, + .has_mbias = true, .has_mic2 = true, }; static const struct sun8i_codec_analog_quirks sun8i_h3_quirks = { .has_linein = true, .has_lineout = true, + .has_mbias = true, .has_mic2 = true, }; @@ -734,6 +753,12 @@ static int sun8i_codec_analog_cmpnt_probe(struct snd_soc_component *cmpnt) return ret; } + if (quirks->has_mbias) { + ret = sun8i_codec_add_mbias(cmpnt); + if (ret) + return ret; + } + if (quirks->has_mic2) { ret = sun8i_codec_add_mic2(cmpnt); if (ret) -- 2.11.0

1 0

[alsa-devel] [PATCH] ALSA: control: remove entry limitation for list operation
by Takashi Sakamoto 24 May '17

24 May '17

In current implementation of ALSA control core, list operation has a limitation to handle 16384 entries at once. This seems due to allocation in kernel space to copy data from user space. With a commit 53e7bf452584 ("ALSA: control: Simplify snd_ctl_elem_list() implementation"), for the operation, ALSA control core copies data into user space directly. No need to care of kernel spaces anymore. This commit purges the limitation. --- sound/core/control.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 47080da8451a..ecd358213b83 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -757,9 +757,7 @@ static int snd_ctl_elem_list(struct snd_card *card, return -EFAULT; offset = list.offset; space = list.space; - /* try limit maximum space */ - if (space > 16384) - return -ENOMEM; + down_read(&card->controls_rwsem); list.count = card->controls_count; list.used = 0; -- 2.11.0

2 3

[alsa-devel] BUG: double free of PCM structure upon module removal
by Robert Jarzmik 24 May '17

24 May '17

Hi Mark, I've encoutered a double free on my platform with the last mainline kernel. The devm_card_release() call makes a double free in my platform configuration. This double release is a PCM free, ie. a soc_free_pcm_runtime() is called and then soc_pcm_free() is called on the same PCM. The call stack as far as I could tell is : - devm_card_release() snd_soc_unregister_card() soc_remove_pcm_runtimes() soc_free_pcm_runtime() ===> first release ... snd_card_free() snd_card_free_when_closed() put_device() release_card_device() snd_device_free_all() __snd_device_free() snd_pcm_dev_free() snd_pcm_free() ===> second release The issue is seen on the PXA platform, where the PCM is allocated in : sound/soc/pxa/pxa2xx-pcm.c, pxa2xx_soc_pcm_new() Did you notice the same behavior on other platforms, and if not would you have a hint why it happens to me ? Cheers. -- Robert PS: The stacks I captured (with added WARN_ON() calls) : ================================================================ =============== Added warning for first pcm free =============== ================================================================ ------------[ cut here ]------------ [ 4033.143750] RJK: soc_new_pcm_runtime() rtd=c307c000, rtd->card=bf025a08 [ 4033.153523] RJK: soc_new_pcm_runtime() rtd=c39cb800, rtd->card=bf025a08 [ 4036.674226] WARNING: CPU: 0 PID: 969 at sound/soc/soc-core.c:578 soc_free_pcm_runtime+0x28/0x6c [ 4036.680894] Modules linked in: snd_soc_mioa701(-) snd_soc_pxa2xx snd_soc_pxa2xx_ac97 snd_soc_wm9713 snd_pxa2xx_lib snd_pcm_dmaengine regmap_ac97 [ 4036.691540] CPU: 0 PID: 969 Comm: rmmod Not tainted 4.12.0-rc2+ #35 [ 4036.698136] Hardware name: MIO A701 [ 4036.701883] [<c00125fc>] (unwind_backtrace) from [<c000eb58>] (show_stack+0x20/0x24) [ 4036.708352] [<c000eb58>] (show_stack) from [<c0229274>] (dump_stack+0x20/0x28) [ 4036.714974] [<c0229274>] (dump_stack) from [<c001d354>] (__warn+0xe8/0x114) [ 4036.721680] [<c001d354>] (__warn) from [<c001d450>] (warn_slowpath_null+0x30/0x38) [ 4036.728132] [<c001d450>] (warn_slowpath_null) from [<c03d0590>] (soc_free_pcm_runtime+0x28/0x6c) [ 4036.734727] [<c03d0590>] (soc_free_pcm_runtime) from [<c03d0968>] (soc_remove_pcm_runtimes+0x70/0x94) [ 4036.741435] [<c03d0968>] (soc_remove_pcm_runtimes) from [<c03d1cc0>] (snd_soc_unregister_card+0x5c/0xdc) [ 4036.747840] [<c03d1cc0>] (snd_soc_unregister_card) from [<c03e1c84>] (devm_card_release+0x1c/0x20) [ 4036.754415] [<c03e1c84>] (devm_card_release) from [<c02cfe18>] (release_nodes+0x1f0/0x228) [ 4036.761125] [<c02cfe18>] (release_nodes) from [<c02d075c>] (devres_release_all+0x68/0x70) [ 4036.767522] [<c02d075c>] (devres_release_all) from [<c02cc24c>] (device_release_driver_internal+0x170/0x204) [ 4036.774064] [<c02cc24c>] (device_release_driver_internal) from [<c02cc3b4>] (driver_detach+0xb0/0xd0) [ 4036.780703] [<c02cc3b4>] (driver_detach) from [<c02cb0a0>] (bus_remove_driver+0x9c/0xdc) [ 4036.786995] [<c02cb0a0>] (bus_remove_driver) from [<c02ccbdc>] (driver_unregister+0x4c/0x58) [ 4036.793449] [<c02ccbdc>] (driver_unregister) from [<c02ce270>] (platform_driver_unregister+0x1c/0x20) [ 4036.799763] [<c02ce270>] (platform_driver_unregister) from [<bf025174>] (mioa701_wm9713_driver_exit+0x14/0x1c [snd_soc_mioa701]) [ 4036.809548] [<bf025174>] (mioa701_wm9713_driver_exit [snd_soc_mioa701]) from [<c0088a74>] (SyS_delete_module+0x194/0x220) [ 4036.819840] [<c0088a74>] (SyS_delete_module) from [<c000a960>] (ret_fast_syscall+0x0/0x1c) [ 4036.826896] ---[ end trace 6aefa683095edac5 ]--- [ 4036.830510] RJK: soc_free_pcm_runtime() rtd=c307c000, rtd->card=bf025a08 [ 4036.837138] ------------[ cut here ]------------ [ 4036.840936] WARNING: CPU: 0 PID: 969 at sound/soc/soc-core.c:578 soc_free_pcm_runtime+0x28/0x6c [ 4036.847717] Modules linked in: snd_soc_mioa701(-) snd_soc_pxa2xx snd_soc_pxa2xx_ac97 snd_soc_wm9713 snd_pxa2xx_lib snd_pcm_dmaengine regmap_ac97 [ 4036.857917] CPU: 0 PID: 969 Comm: rmmod Tainted: G W 4.12.0-rc2+ #35 [ 4036.864741] Hardware name: MIO A701 [ 4036.868164] [<c00125fc>] (unwind_backtrace) from [<c000eb58>] (show_stack+0x20/0x24) [ 4036.875069] [<c000eb58>] (show_stack) from [<c0229274>] (dump_stack+0x20/0x28) [ 4036.881999] [<c0229274>] (dump_stack) from [<c001d354>] (__warn+0xe8/0x114) [ 4036.888630] [<c001d354>] (__warn) from [<c001d450>] (warn_slowpath_null+0x30/0x38) [ 4036.895458] [<c001d450>] (warn_slowpath_null) from [<c03d0590>] (soc_free_pcm_runtime+0x28/0x6c) [ 4036.902422] [<c03d0590>] (soc_free_pcm_runtime) from [<c03d0968>] (soc_remove_pcm_runtimes+0x70/0x94) [ 4036.909100] [<c03d0968>] (soc_remove_pcm_runtimes) from [<c03d1cc0>] (snd_soc_unregister_card+0x5c/0xdc) [ 4036.915941] [<c03d1cc0>] (snd_soc_unregister_card) from [<c03e1c84>] (devm_card_release+0x1c/0x20) [ 4036.922900] [<c03e1c84>] (devm_card_release) from [<c02cfe18>] (release_nodes+0x1f0/0x228) [ 4036.929603] [<c02cfe18>] (release_nodes) from [<c02d075c>] (devres_release_all+0x68/0x70) [ 4036.936447] [<c02d075c>] (devres_release_all) from [<c02cc24c>] (device_release_driver_internal+0x170/0x204) [ 4036.943404] [<c02cc24c>] (device_release_driver_internal) from [<c02cc3b4>] (driver_detach+0xb0/0xd0) [ 4036.950059] [<c02cc3b4>] (driver_detach) from [<c02cb0a0>] (bus_remove_driver+0x9c/0xdc) [ 4036.956876] [<c02cb0a0>] (bus_remove_driver) from [<c02ccbdc>] (driver_unregister+0x4c/0x58) [ 4036.963793] [<c02ccbdc>] (driver_unregister) from [<c02ce270>] (platform_driver_unregister+0x1c/0x20) [ 4036.970766] [<c02ce270>] (platform_driver_unregister) from [<bf025174>] (mioa701_wm9713_driver_exit+0x14/0x1c [snd_soc_mioa701]) [ 4036.981054] [<bf025174>] (mioa701_wm9713_driver_exit [snd_soc_mioa701]) from [<c0088a74>] (SyS_delete_module+0x194/0x220) [ 4036.991323] [<c0088a74>] (SyS_delete_module) from [<c000a960>] (ret_fast_syscall+0x0/0x1c) [ 4036.998052] ---[ end trace 6aefa683095edac6 ]--- [ 4037.001597] RJK: soc_free_pcm_runtime() rtd=c39cb800, rtd->card=bf025a08 ------------[ cut here ]------------ ================================================================ ====================== Here comes the BUG ====================== ================================================================ [ 4037.309613] RJK: soc_pcm_free() rtd=c39cb800, rtd->card=35343720 [ 4037.327993] Unable to handle kernel paging request at virtual address 35343848 [ 4037.334248] pgd = c3184000 [ 4037.337047] [35343848] *pgd=00000000 [ 4037.339722] Internal error: Oops: f5 [#1] PREEMPT ARM [ 4037.342391] Modules linked in: snd_soc_mioa701(-) snd_soc_pxa2xx snd_soc_pxa2xx_ac97 snd_soc_wm9713 snd_pxa2xx_lib snd_pcm_dmaengine regmap_ac97 [ 4037.350342] CPU: 0 PID: 969 Comm: rmmod Tainted: G W 4.12.0-rc2+ #35 [ 4037.355660] Hardware name: MIO A701 [ 4037.358383] task: c2871ae0 task.stack: c31f4000 [ 4037.361184] PC is at soc_pcm_free+0x38/0x80 [ 4037.364008] LR is at preempt_count_sub+0x3c/0x58 [ 4037.366807] pc : [<c03dcea8>] lr : [<c0043494>] psr: 60000013 [ 4037.366807] sp : c31f5cf8 ip : 00000000 fp : c31f5d14 [ 4037.374932] r10: 00000000 r9 : c02cf494 r8 : c287fa60 [ 4037.377654] r7 : c287f960 r6 : c2811000 r5 : c39cb800 r4 : c072b394 [ 4037.383037] r3 : 35343720 r2 : 35343720 r1 : c0059460 r0 : 00000034 [ 4037.388622] Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none [ 4037.394328] Control: 0000397f Table: a3184000 DAC: 00000051 [ 4037.400110] Process rmmod (pid: 969, stack limit = 0xc31f4198) [ 4037.406089] Stack: (0xc31f5cf8 to 0xc31f6000) [ 4037.409177] 5ce0: c03dce70 c072b394 [ 4037.415346] 5d00: c2811000 c072b3a0 c31f5d34 c31f5d18 c03b0c7c c03dce7c c03b0e68 c287f120 [ 4037.421501] 5d20: c39cb190 c3181960 c31f5d44 c31f5d38 c03b0e84 c03b0c14 c31f5d5c c31f5d48 [ 4037.427649] 5d40: c03a9d28 c03b0e74 c03a5160 c336e060 c31f5d74 c31f5d60 c03a9f00 c03a9ce0 [ 4037.433829] 5d60: c39cb358 c39cb000 c31f5d8c c31f5d78 c03a5184 c03a9ed8 c39cb360 c39cb358 [ 4037.440010] 5d80: c31f5dac c31f5d90 c02c6264 c03a516c c02c61f8 c39cb360 c0721cd4 c07382f0 [ 4037.446202] 5da0: c31f5dd4 c31f5db0 c022ab8c c02c6204 c05cbbc0 c31f5dac c39cb358 c39cb000 [ 4037.452401] 5dc0: bf025ad0 c287f980 c31f5dec c31f5dd8 c02c70cc c022aa28 00020000 00000000 [ 4037.458590] 5de0: c31f5e04 c31f5df0 c03a57d4 c02c70a0 c31f5e0c c39cb000 c31f5e2c c31f5e08 [ 4037.464923] 5e00: c03a594c c03a57b4 bf025518 00000000 c31f5e10 c31f5e10 bf025518 bf025a08 [ 4037.471327] 5e20: c31f5e4c c31f5e30 c03d1d0c c03a5914 c03e1c68 c31f5e60 00000003 c06ef270 [ 4037.477748] 5e40: c31f5e5c c31f5e50 c03e1c84 c03d1c70 c31f5e94 c31f5e60 c02cfe18 c03e1c74 [ 4037.484180] 5e60: c287f9c0 c287f980 a0000013 c06ef270 a0000013 c0721f80 bf025bcc c0721fb4 [ 4037.490631] 5e80: c31f4000 00020000 c31f5eac c31f5e98 c02d075c c02cfc34 c06ef270 c06ef2a4 [ 4037.497192] 5ea0: c31f5ed4 c31f5eb0 c02cc24c c02d0700 00020000 c06ef270 bf025bcc c31f4000 [ 4037.503789] 5ec0: 00000081 c000ab28 c31f5ef4 c31f5ed8 c02cc3b4 c02cc0e8 00000000 bf025bcc [ 4037.510424] 5ee0: beebd4a8 00000880 c31f5f0c c31f5ef8 c02cb0a0 c02cc310 bf025160 bf025bcc [ 4037.517107] 5f00: c31f5f24 c31f5f10 c02ccbdc c02cb010 bf025160 bf025d40 c31f5f34 c31f5f28 [ 4037.523869] 5f20: c02ce270 c02ccb9c c31f5f44 c31f5f38 bf025174 c02ce260 c31f5fa4 c31f5f48 [ 4037.530706] 5f40: c0088a74 bf02516c 00001000 5f646e73 5f636f73 616f696d 00313037 c31f5f68 [ 4037.537553] 5f60: c00ed5cc c00514e4 c31f5f68 c31f5f68 c31f4000 007fd008 00000000 beebd4d0 [ 4037.544443] 5f80: beebd4a8 00000081 00000000 00000000 beebd4d0 beebd4a8 00000000 c31f5fa8 [ 4037.551358] 5fa0: c000a960 c00888ec 00000000 beebd4d0 beebd4a8 00000880 00000000 00000000 [ 4037.558280] 5fc0: 00000000 beebd4d0 beebd4a8 00000081 beebfcd0 beebfd10 00000880 beebfd44 [ 4037.565213] 5fe0: 00000000 beebd4a0 00009088 b6f2fbcc 60000010 beebd4a8 00000000 00000000 [ 4037.572210] [<c03dcea8>] (soc_pcm_free) from [<c03b0c7c>] (snd_pcm_free+0x74/0xa0) [ 4037.579179] [<c03b0c7c>] (snd_pcm_free) from [<c03b0e84>] (snd_pcm_dev_free+0x1c/0x20) [ 4037.586210] [<c03b0e84>] (snd_pcm_dev_free) from [<c03a9d28>] (__snd_device_free+0x54/0x7c) [ 4037.593253] [<c03a9d28>] (__snd_device_free) from [<c03a9f00>] (snd_device_free_all+0x34/0x48) [ 4037.600252] [<c03a9f00>] (snd_device_free_all) from [<c03a5184>] (release_card_device+0x24/0x7c) [ 4037.607279] [<c03a5184>] (release_card_device) from [<c02c6264>] (device_release+0x6c/0xa4) [ 4037.614304] [<c02c6264>] (device_release) from [<c022ab8c>] (kobject_put+0x170/0x1e0) [ 4037.621295] [<c022ab8c>] (kobject_put) from [<c02c70cc>] (put_device+0x38/0x44) [ 4037.628237] [<c02c70cc>] (put_device) from [<c03a57d4>] (snd_card_free_when_closed+0x2c/0x34) [ 4037.635196] [<c03a57d4>] (snd_card_free_when_closed) from [<c03a594c>] (snd_card_free+0x44/0x68) [ 4037.642125] [<c03a594c>] (snd_card_free) from [<c03d1d0c>] (snd_soc_unregister_card+0xa8/0xdc) [ 4037.649044] [<c03d1d0c>] (snd_soc_unregister_card) from [<c03e1c84>] (devm_card_release+0x1c/0x20) [ 4037.655995] [<c03e1c84>] (devm_card_release) from [<c02cfe18>] (release_nodes+0x1f0/0x228) [ 4037.662917] [<c02cfe18>] (release_nodes) from [<c02d075c>] (devres_release_all+0x68/0x70) [ 4037.669805] [<c02d075c>] (devres_release_all) from [<c02cc24c>] (device_release_driver_internal+0x170/0x204) [ 4037.676678] [<c02cc24c>] (device_release_driver_internal) from [<c02cc3b4>] (driver_detach+0xb0/0xd0) [ 4037.683522] [<c02cc3b4>] (driver_detach) from [<c02cb0a0>] (bus_remove_driver+0x9c/0xdc) [ 4037.690308] [<c02cb0a0>] (bus_remove_driver) from [<c02ccbdc>] (driver_unregister+0x4c/0x58) [ 4037.697092] [<c02ccbdc>] (driver_unregister) from [<c02ce270>] (platform_driver_unregister+0x1c/0x20) [ 4037.703885] [<c02ce270>] (platform_driver_unregister) from [<bf025174>] (mioa701_wm9713_driver_exit+0x14/0x1c [snd_soc_mioa701]) [ 4037.714014] [<bf025174>] (mioa701_wm9713_driver_exit [snd_soc_mioa701]) from [<c0088a74>] (SyS_delete_module+0x194/0x220) [ 4037.724128] [<c0088a74>] (SyS_delete_module) from [<c000a960>] (ret_fast_syscall+0x0/0x1c) [ 4037.730848] Code: e59f104c ebf37450 e5952004 e1a03002 (e5b34128) [ 4037.738251] ---[ end trace 6aefa683095edac7 ]--- -- Robert

3 5

[alsa-devel] [PATCH] ASoC: Add pcm9211 driver
by Julian Scheel 24 May '17

24 May '17

This adds a driver for the TI PCM9211 digital audio interfaces transceiver. The driver currently only handles the receiver aspect of the chip and supports only control of the primary I2S (MAIN) output port. Support for AUX output is trivial to add, but wasn't done due to missing hardware for testing. The same goes for the transmitter functionality. The driver uses some extra device-tree fields to allow configuration of pin functions directly out of this driver. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- .../devicetree/bindings/sound/pcm9211.txt | 118 ++ MAINTAINERS | 6 + include/dt-bindings/sound/pcm9211.h | 55 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/pcm9211-i2c.c | 65 + sound/soc/codecs/pcm9211.c | 1357 ++++++++++++++++++++ sound/soc/codecs/pcm9211.h | 206 +++ 8 files changed, 1821 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm9211.txt create mode 100644 include/dt-bindings/sound/pcm9211.h create mode 100644 sound/soc/codecs/pcm9211-i2c.c create mode 100644 sound/soc/codecs/pcm9211.c create mode 100644 sound/soc/codecs/pcm9211.h diff --git a/Documentation/devicetree/bindings/sound/pcm9211.txt b/Documentation/devicetree/bindings/sound/pcm9211.txt new file mode 100644 index 000000000000..bfc29889caf6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm9211.txt @@ -0,0 +1,118 @@ +PCM9211 audio CODEC + +This device support both I2C and SPI (configured with pin strapping +on the board). The driver is currently implementing i2c only. + +Required properties: + + - compatible : "ti,pcm9211" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + + - VCCAD-supply, VCC-supply, VDDRX-supply and AVDD-supply : + power supplies for the device, as covered in bindings/regulator/regulator.txt + +Optional properties: + + - clocks : A clock specifier for the clock connected to XTI. While the + device could work without it, this driver currently relies on it being + available + + - clock-names : Must speficy "xti" to match the requested clock. + + - reset-gpios : GPIO wired to the reset pin, if not supplied a soft-reset is + used as fallback + + - int0-gpios : GPIO connected to interrupt0 output of the PCM9211. The + line can be configured to any of MPO0,1 and MPIOA,B,C0-4 and error/int0. + If this is specified it is used as an interrupt to notify alsa controls + about changes of the incoming sampling rate on DIR (spdif) as well as NPCM + status. + + - ti,group-function: An array with size 3 specifying function for pingroups + A, B and C. Possible values are defined in dt-bindings/sound/pcm9211.h + + - ti,mpio-a-flags-gpio: An array with size 4 specifying flags out or gpio + mode per pin, when function DIR_FLAGS_GPIO is selected for group a. + Possible values are defined in dt-bindings/sound/pcm9211.h + + - ti,mpio-b-flags-gpio: An array with size 4 specifying flags out or gpio + mode per pin, when function DIR_FLAGS_GPIO is selected for group b. + Possible values are defined in dt-bindings/sound/pcm9211.h + + - ti,mpio-c-flags-gpio: An array with size 4 specifying flags out or gpio + mode per pin, when function DIR_FLAGS_GPIO is selected for group c. + Possible values are defined in dt-bindings/sound/pcm9211.h + + - ti,mpio-a-flag: An array with size 4 specifying flag assinged per pin, + when function DIR_FLAGS_GPIO is selected for group a and pin is set to + flags mode. + Possible values are defined in dt-bindings/sound/pcm9211.h + + - ti,mpio-b-flag: An array with size 4 specifying flag assinged per pin, + when function DIR_FLAGS_GPIO is selected for group b and pin is set to + flags mode. + Possible values are defined in dt-bindings/sound/pcm9211.h + + - ti,mpio-c-flag: An array with size 4 specifying flag assinged per pin, + when function DIR_FLAGS_GPIO is selected for group c and pin is set to + flags mode. + Possible values are defined in dt-bindings/sound/pcm9211.h + + - ti,mpo-function: An array with size 2 specifying flag assinged per mpo + pin. + Possible values are defined in dt-bindings/sound/pcm9211.h + + - ti,int0-function: Selects error/int0 pin function. + Possible values are defined in dt-bindings/sound/pcm9211.h + + - ti,int1-function: Selects error/int1 pin function. + Possible values are defined in dt-bindings/sound/pcm9211.h + +Examples: + + pcm9211: pcm9211@43 { + compatible = "ti,pcm9211"; + reg = <0x43>; + + VCCAD-supply = <&reg_5v0_analog>; + VCC-supply = <&reg_3v3_pll_analog>; + VDDRX-supply = <&reg_3v3>; + DVDD-supply = <&reg_3v3>; + + clocks = <&xti_clk>; + clock-names = "xti"; + }; + + + pcm9211: pcm9211@43 { + compatible = "ti,pcm9211"; + reg = <0x43>; + + VCCAD-supply = <&reg_5v0_analog>; + VCC-supply = <&reg_3v3_pll_analog>; + VDDRX-supply = <&reg_3v3>; + DVDD-supply = <&reg_3v3>; + + clocks = <&xti_clk>; + clock-names = "xti"; + + reset-gpios = <&gpio1 0 GPIO_ACTIVE_HIGH>; + int-gpios = <&gpio1 0 GPIO_ACTIVE_HIGH>; + + ti,group-function = /bits/ 8 + <PCM9211_MPIO_A_GROUP_BIPHASE_INPUT, + PCM9211_MPIO_B_GROUP_DIR_FLAGS_GPIO, + PCM9211_MPIO_C_GROUP_AUXIN1>; + ti,mpio-b-flags-gpio = /bits/ 8 + <PCM9211_MPIO_DIR_FLAGS, + PCM9211_MPIO_DIR_FLAGS, + PCM9211_MPIO_DIR_FLAGS, + PCM9211_MPIO_DIR_FLAGS>; + ti,mpio-b-flag = /bits/ 8 + <PCM9211_MPIO_FLAG_INT0, + PCM9211_MPIO_FLAG_CLKST, + PCM9211_MPIO_FLAG_CLKST, + PCM9211_MPIO_FLAG_CLKST>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index c2ffc96ccf4c..1f3b2e98dd9d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -9985,6 +9985,12 @@ S: Supported F: Documentation/devicetree/bindings/pci/pci-thunder-* F: drivers/pci/host/pci-thunder-* +PCM9211 AUDIO CODEC DRIVER +M: Julian Scheel <julian(a)jusst.de> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: sound/soc/codecs/pcm9211* + PCMCIA SUBSYSTEM P: Linux PCMCIA Team L: linux-pcmcia(a)lists.infradead.org diff --git a/include/dt-bindings/sound/pcm9211.h b/include/dt-bindings/sound/pcm9211.h new file mode 100644 index 000000000000..6d1367f21d25 --- /dev/null +++ b/include/dt-bindings/sound/pcm9211.h @@ -0,0 +1,55 @@ +#ifndef __DT_PCM9211_H +#define __DT_PCM9211_H + +/* INT0/ERROR select */ +#define PCM9211_ERROR_INT0_ERROR 0 +#define PCM9211_ERROR_INT0_INT0 1 + +/* INT1/NPCM select */ +#define PCM9211_NPCM_INT1_NPCM 0 +#define PCM9211_NPCM_INT1_INT1 1 + +/* MPIO group functions */ +#define PCM9211_MPIO_A_GROUP_BIPHASE_INPUT 0 +#define PCM9211_MPIO_A_GROUP_CLKST_VOUT_XMCK_INT0 1 +#define PCM9211_MPIO_A_GROUP_SEC_BCK_LRCK_XMCKO_INT0 2 +#define PCM9211_MPIO_A_GROUP_DIR_FLAGS_GPIO 3 + +#define PCM9211_MPIO_B_GROUP_AUXIN2 0 +#define PCM9211_MPIO_B_GROUP_AUXOUT 1 +#define PCM9211_MPIO_B_GROUP_SAMPLING_FREQ_RES 2 +#define PCM9211_MPIO_B_GROUP_DIR_FLAGS_GPIO 3 +#define PCM9211_MPIO_B_GROUP_DIR_BCUV_BFRAME_VUC 4 +#define PCM9211_MPIO_B_GROUP_EXT_ADC_IN 5 +#define PCM9211_MPIO_B_GROUP_TEST_MODE 7 + +#define PCM9211_MPIO_C_GROUP_AUXIN1 0 +#define PCM9211_MPIO_C_GROUP_ADC_CLOCK_DATA 1 +#define PCM9211_MPIO_C_GROUP_SAMPLING_FREQ_RES 2 +#define PCM9211_MPIO_C_GROUP_DIR_FLAGS_GPIO 3 +#define PCM9211_MPIO_C_GROUP_DIR_BCUV_BFRAME_VUC 4 +#define PCM9211_MPIO_C_GROUP_DIT_CLOCK_DATA 5 + +/* MPIO flags/gpio select */ +#define PCM9211_MPIO_DIR_FLAGS 0 +#define PCM9211_MPIO_GPIO 1 + +/* MPIO flags */ +#define PCM9211_MPIO_FLAG_CLKST 0 +#define PCM9211_MPIO_FLAG_EMPH 1 +#define PCM9211_MPIO_FLAG_BPSYNC 2 +#define PCM9211_MPIO_FLAG_DTSCD 3 +#define PCM9211_MPIO_FLAG_PARITY 4 +#define PCM9211_MPIO_FLAG_LOCK 5 +#define PCM9211_MPIO_FLAG_VOUT 6 +#define PCM9211_MPIO_FLAG_UOUT 7 +#define PCM9211_MPIO_FLAG_COUT 8 +#define PCM9211_MPIO_FLAG_BFRAME 9 +#define PCM9211_MPIO_FLAG_FSOUT0 10 +#define PCM9211_MPIO_FLAG_FSOUT1 11 +#define PCM9211_MPIO_FLAG_FSOUT2 12 +#define PCM9211_MPIO_FLAG_FSOUT3 13 +#define PCM9211_MPIO_FLAG_INT0 14 +#define PCM9211_MPIO_FLAG_INT1 15 + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 883ed4c8a551..d7cf85ba020f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -113,6 +113,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM5102A select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_PCM9211_I2C if I2C select SND_SOC_RT286 if I2C select SND_SOC_RT298 if I2C select SND_SOC_RT5514 if I2C @@ -684,6 +685,15 @@ config SND_SOC_PCM512x_SPI select SND_SOC_PCM512x select REGMAP_SPI +config SND_SOC_PCM9211 + tristate + +config SND_SOC_PCM9211_I2C + tristate "Texas Instruments PCM9211 CODEC - I2C" + depends on I2C + select SND_SOC_PCM9211 + select REGMAP_I2C + config SND_SOC_RL6231 tristate default y if SND_SOC_RT5514=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 28a63fdaf982..a8eecee5fcfc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -110,6 +110,8 @@ snd-soc-pcm5102a-objs := pcm5102a.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o +snd-soc-pcm9211-objs := pcm9211.o +snd-soc-pcm9211-i2c-objs := pcm9211-i2c.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt286-objs := rt286.o @@ -344,6 +346,8 @@ obj-$(CONFIG_SND_SOC_PCM5102A) += snd-soc-pcm5102a.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o +obj-$(CONFIG_SND_SOC_PCM9211) += snd-soc-pcm9211.o +obj-$(CONFIG_SND_SOC_PCM9211_I2C) += snd-soc-pcm9211-i2c.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o diff --git a/sound/soc/codecs/pcm9211-i2c.c b/sound/soc/codecs/pcm9211-i2c.c new file mode 100644 index 000000000000..e0678b280f11 --- /dev/null +++ b/sound/soc/codecs/pcm9211-i2c.c @@ -0,0 +1,65 @@ +/* + * PCM9211 codec i2c driver + * + * Copyright (C) 2017 jusst technologies GmbH / jusst.engineering + * + * Author: Julian Scheel <julian(a)jusst.de> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + */ + +#include <linux/i2c.h> +#include <linux/module.h> + +#include <sound/soc.h> + +#include "pcm9211.h" + +static int pcm9211_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &pcm9211_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return pcm9211_probe(&i2c->dev, regmap); +} + +static int pcm9211_i2c_remove(struct i2c_client *i2c) +{ + pcm9211_remove(&i2c->dev); + + return 0; +} + +static const struct i2c_device_id pcm9211_i2c_id[] = { + { "pcm9211", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, pcm9211_i2c_id); + +static const struct of_device_id pcm9211_of_match[] = { + { .compatible = "ti,pcm9211", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm9211_of_match); + +static struct i2c_driver pcm9211_i2c_driver = { + .probe = pcm9211_i2c_probe, + .remove = pcm9211_i2c_remove, + .id_table = pcm9211_i2c_id, + .driver = { + .name = "pcm9211", + .of_match_table = pcm9211_of_match, + .pm = &pcm9211_pm_ops, + }, +}; +module_i2c_driver(pcm9211_i2c_driver); + +MODULE_DESCRIPTION("PCM9211 I2C codec driver"); +MODULE_AUTHOR("Julian Scheel <julian(a)jusst.de>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm9211.c b/sound/soc/codecs/pcm9211.c new file mode 100644 index 000000000000..0ab4ad551df8 --- /dev/null +++ b/sound/soc/codecs/pcm9211.c @@ -0,0 +1,1357 @@ +/* + * PCM9211 codec driver + * + * Copyright (C) 2017 jusst technologies GmbH / jusst.engineering + * + * Author; Julian Scheel <julian(a)jusst.de> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + */ + +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/pm_runtime.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/delay.h> +#include <linux/clk.h> + +#include <sound/control.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "pcm9211.h" + +#define PCM9211_MAX_SYSCLK 24576000 + +#define PCM9211_SUPPLIES 4 +static const char *const pcm9211_supply_names[PCM9211_SUPPLIES] = { + "VCCAD", + "VCC", + "VDDRX", + "DVDD", +}; + +struct pcm9211_priv { + struct regulator_bulk_data supplies[PCM9211_SUPPLIES]; + struct snd_pcm_hw_constraint_list rate_constraints; + struct delayed_work npcm_clear_work; + struct snd_kcontrol *preamble_ctl; + struct snd_kcontrol *npcm_ctl; + struct snd_kcontrol *rate_ctl; + struct snd_kcontrol *dts_ctl; + struct snd_soc_codec *codec; + struct gpio_desc *reset; + struct gpio_desc *int0; + struct regmap *regmap; + struct device *dev; + struct clk *xti; + + unsigned int dir_rate; + unsigned long sysclk; + u8 burst_preamble[4]; + u8 npcm_state; +}; + +static const struct regmap_range pcm9211_reg_rd_range[] = { + regmap_reg_range(PCM9211_ERR_OUT, PCM9211_PD_BUF1), + regmap_reg_range(PCM9211_SYS_RESET, PCM9211_SYS_RESET), + regmap_reg_range(PCM9211_ADC_CTRL1, PCM9211_ADC_CTRL1), + regmap_reg_range(PCM9211_ADC_L_CH_ATT, PCM9211_ADC_CTRL3), + regmap_reg_range(PCM9211_DIR_STATUS1, PCM9211_DIT_STATUS6), + regmap_reg_range(PCM9211_MAIN_AUX_MUTE, PCM9211_MPIO_C_DATA_IN), +}; + +static const struct regmap_access_table pcm9211_reg_rd_table = { + .yes_ranges = pcm9211_reg_rd_range, + .n_yes_ranges = ARRAY_SIZE(pcm9211_reg_rd_range), +}; + +static const struct regmap_range pcm9211_reg_wr_range[] = { + regmap_reg_range(PCM9211_ERR_OUT, PCM9211_INT1_CAUSE), + regmap_reg_range(PCM9211_INT_POLARITY, PCM9211_FS_CALC_TARGET), + regmap_reg_range(PCM9211_SYS_RESET, PCM9211_SYS_RESET), + regmap_reg_range(PCM9211_ADC_CTRL1, PCM9211_ADC_CTRL1), + regmap_reg_range(PCM9211_ADC_L_CH_ATT, PCM9211_ADC_CTRL3), + regmap_reg_range(PCM9211_DIT_CTRL1, PCM9211_DIT_STATUS6), + regmap_reg_range(PCM9211_MAIN_AUX_MUTE, PCM9211_MPIO_C_DATA_OUT), +}; + +static const struct regmap_access_table pcm9211_reg_wr_table = { + .yes_ranges = pcm9211_reg_wr_range, + .n_yes_ranges = ARRAY_SIZE(pcm9211_reg_wr_range), +}; + +static const struct regmap_range pcm9211_reg_volatile_range[] = { + regmap_reg_range(PCM9211_INT0_OUT, PCM9211_INT1_OUT), + regmap_reg_range(PCM9211_BIPHASE_INFO, PCM9211_PD_BUF1), + regmap_reg_range(PCM9211_DIR_STATUS1, PCM9211_DIR_STATUS6), +}; + +static const struct regmap_access_table pcm9211_reg_volatile_table = { + .yes_ranges = pcm9211_reg_volatile_range, + .n_yes_ranges = ARRAY_SIZE(pcm9211_reg_volatile_range), +}; + +static const struct reg_default pcm9211_reg_defaults[] = { + { PCM9211_ERR_OUT, 0x00 }, + { PCM9211_DIR_INITIAL1, 0x00 }, + { PCM9211_DIR_INITIAL2, 0x01 }, + { PCM9211_DIR_INITIAL3, 0x04 }, + { PCM9211_OSC_CTRL, 0x00 }, + { PCM9211_ERR_CAUSE, 0x01 }, + { PCM9211_AUTO_SEL_CAUSE, 0x01 }, + { PCM9211_DIR_FS_RANGE, 0x00 }, + { PCM9211_NON_PCM_DEF, 0x03 }, + { PCM9211_DTS_CD_LD, 0x0c }, + { PCM9211_INT0_CAUSE, 0xff }, + { PCM9211_INT1_CAUSE, 0xff }, + { PCM9211_INT0_OUT, 0x00 }, + { PCM9211_INT1_OUT, 0x00 }, + { PCM9211_INT_POLARITY, 0x00 }, + { PCM9211_DIR_OUT_FMT, 0x04 }, + { PCM9211_DIR_RSCLK_RATIO, 0x02 }, + { PCM9211_XTI_SCLK_FREQ, 0x1a }, + { PCM9211_DIR_SOURCE_BIT2, 0x22 }, + { PCM9211_XTI_SOURCE_BIT2, 0x22 }, + { PCM9211_DIR_INP_BIPHASE, 0xc2 }, + { PCM9211_RECOUT0_BIPHASE, 0x02 }, + { PCM9211_RECOUT1_BIPHASE, 0x02 }, + { PCM9211_FS_CALC_TARGET, 0x00 }, + { PCM9211_FS_CALC_RESULT, 0x08 }, + { PCM9211_BIPHASE_INFO, 0x08 }, + { PCM9211_PC_BUF0, 0x01 }, + { PCM9211_PC_BUF1, 0x00 }, + { PCM9211_PD_BUF0, 0x20 }, + { PCM9211_PD_BUF1, 0x57 }, + { PCM9211_SYS_RESET, 0x40 }, + { PCM9211_ADC_CTRL1, 0x02 }, + { PCM9211_ADC_L_CH_ATT, 0xd7 }, + { PCM9211_ADC_R_CH_ATT, 0xd7 }, + { PCM9211_ADC_CTRL2, 0x00 }, + { PCM9211_ADC_CTRL3, 0x00 }, + { PCM9211_DIR_STATUS1, 0x04 }, + { PCM9211_DIR_STATUS2, 0x00 }, + { PCM9211_DIR_STATUS3, 0x00 }, + { PCM9211_DIR_STATUS4, 0x00 }, + { PCM9211_DIR_STATUS5, 0x00 }, + { PCM9211_DIR_STATUS6, 0x00 }, + { PCM9211_DIT_CTRL1, 0x44 }, + { PCM9211_DIT_CTRL2, 0x10 }, + { PCM9211_DIT_CTRL3, 0x00 }, + { PCM9211_DIT_STATUS1, 0x00 }, + { PCM9211_DIT_STATUS2, 0x00 }, + { PCM9211_DIT_STATUS3, 0x00 }, + { PCM9211_DIT_STATUS4, 0x00 }, + { PCM9211_DIT_STATUS5, 0x00 }, + { PCM9211_DIT_STATUS6, 0x00 }, + { PCM9211_MAIN_AUX_MUTE, 0x00 }, + { PCM9211_MAIN_OUT_SOURCE, 0x00 }, + { PCM9211_AUX_OUT_SOURCE, 0x00 }, + { PCM9211_MPIO_B_MAIN_HIZ, 0x00 }, + { PCM9211_MPIO_C_MPIO_A_HIZ, 0x0f }, + { PCM9211_MPIO_GROUP, 0x40 }, + { PCM9211_MPIO_A_FLAGS, 0x00 }, + { PCM9211_MPIO_B_MPIO_C_FLAGS, 0x00 }, + { PCM9211_MPIO_A1_A0_OUT_FLAG, 0x00 }, + { PCM9211_MPIO_A3_A2_OUT_FLAG, 0x00 }, + { PCM9211_MPIO_B1_B0_OUT_FLAG, 0x00 }, + { PCM9211_MPIO_B3_B2_OUT_FLAG, 0x00 }, + { PCM9211_MPIO_C1_C0_OUT_FLAG, 0x00 }, + { PCM9211_MPIO_C3_C2_OUT_FLAG, 0x00 }, + { PCM9211_MPO_1_0_FUNC, 0x3d }, + { PCM9211_MPIO_A_B_DIR, 0x00 }, + { PCM9211_MPIO_C_DIR, 0x00 }, + { PCM9211_MPIO_A_B_DATA_OUT, 0x00 }, + { PCM9211_MPIO_C_DATA_OUT, 0x00 }, + { PCM9211_MPIO_A_B_DATA_IN, 0x00 }, + { PCM9211_MPIO_C_DATA_IN, 0x02 }, +}; + +const struct regmap_config pcm9211_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = PCM9211_MPIO_C_DATA_IN, + .wr_table = &pcm9211_reg_wr_table, + .rd_table = &pcm9211_reg_rd_table, + .volatile_table = &pcm9211_reg_volatile_table, + .reg_defaults = pcm9211_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm9211_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(pcm9211_regmap); + +static const u32 adc_rates[] = { 48000, 92000 }; +static const struct snd_pcm_hw_constraint_list adc_rate_constraints = { + .count = ARRAY_SIZE(adc_rates), + .list = adc_rates, +}; + +static const int biphase_rates[] = { + 0, 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, + 48000, 64000, 88200, 96000, 128000, 176400, 192000 +}; + +static const unsigned int pcm9211_sck_ratios[] = { 1, 2, 4, 8 }; +static const unsigned int pcm9211_bck_ratios[] = { 2, 4, 8, 16 }; +static const unsigned int pcm9211_lrck_ratios[] = { 128, 256, 512, 1024 }; +#define PCM9211_NUM_SCK_RATIOS ARRAY_SIZE(pcm9211_sck_ratios) +#define PCM9211_NUM_BCK_RATIOS ARRAY_SIZE(pcm9211_bck_ratios) +#define PCM9211_NUM_LRCK_RATIOS ARRAY_SIZE(pcm9211_lrck_ratios) + +static int pcm9211_main_output_port(struct device *dev) +{ + struct pcm9211_priv *priv = dev_get_drvdata(dev); + unsigned int val; + unsigned int port; + int ret; + + ret = regmap_read(priv->regmap, PCM9211_MAIN_OUT_SOURCE, &val); + if (ret) { + dev_err(dev, "Failed to read selected source: %d\n", ret); + return ret; + } + + port = (val & PCM9211_MOPSRC_MASK) >> PCM9211_MOPSRC_SHIFT; + if (port == PCM9211_MOSRC_AUTO) { + ret = regmap_read(priv->regmap, PCM9211_BIPHASE_INFO, &val); + if (ret) { + dev_err(dev, "Failed to read biphase information: %d\n", + ret); + return ret; + } + + /* Assumes that Sampling Frequency Status calculation + * corresponds with DIR Lock, which seems to to be exposed to + * any register directly + */ + if ((val & PCM9211_BIPHASE_SFSST_MASK) == 0) + port = PCM9211_MOSRC_DIR; + else + port = PCM9211_MOSRC_ADC; + } + + return port; +} + +static int pcm9211_dir_rate(struct device *dev) +{ + struct pcm9211_priv *priv = dev_get_drvdata(dev); + unsigned int val; + int ret; + + ret = regmap_read(priv->regmap, PCM9211_BIPHASE_INFO, &val); + if (ret) { + dev_err(dev, "Failed to read biphase information: %d\n", ret); + return ret; + } + + if ((val & PCM9211_BIPHASE_SFSST_MASK)) { + dev_dbg(dev, "Biphase Fs calculation not locked\n"); + return 0; + } + + return biphase_rates[(val & PCM9211_BIPHASE_SFSOUT_MASK) + >> PCM9211_BIPHASE_SFSOUT_SHIFT]; +} + +static int pcm9211_read_burst_preamble(struct device *dev) +{ + struct pcm9211_priv *priv = dev_get_drvdata(dev); + int ret; + + ret = regmap_raw_read(priv->regmap, PCM9211_PC_BUF0, + priv->burst_preamble, 4); + if (ret) { + dev_err(dev, "Failed to read burst preamble: %d\n", ret); + return ret; + } + + dev_dbg(dev, "Burst preamble: 0x%02x 0x%02x 0x%02x 0x%02x\n", + priv->burst_preamble[0], priv->burst_preamble[1], + priv->burst_preamble[2], priv->burst_preamble[3]); + + return 0; +} + +static int pcm9211_dir_rate_kctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 8000; + uinfo->value.integer.min = 96000; + + return 0; +} + +static int pcm9211_dir_rate_kctl(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct pcm9211_priv *priv = snd_soc_component_get_drvdata(component); + struct device *dev = priv->dev; + + /* If we have an interrupt connected dir_rate is up-to-date */ + if (!priv->int0) + priv->dir_rate = pcm9211_dir_rate(dev); + + ucontrol->value.integer.value[0] = priv->dir_rate; + + return 0; +} + +#define pcm9211_dir_npcm_kctl_info snd_ctl_boolean_mono_info +#define pcm9211_dir_npcm_kctl pcm9211_int0_kctl + +#define pcm9211_dir_dtscd_kctl_info snd_ctl_boolean_mono_info +#define pcm9211_dir_dtscd_kctl pcm9211_int0_kctl + +static int pcm9211_int0_kctl(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct pcm9211_priv *priv = snd_soc_component_get_drvdata(component); + u8 mask = kcontrol->private_value & 0xff; + struct device *dev = priv->dev; + unsigned int cause; + int ret; + + /* If interrupt line is not connected read the last interrupt state */ + if (!priv->int0) { + ret = regmap_read(priv->regmap, PCM9211_INT0_OUT, &cause); + if (ret) { + dev_err(dev, "Failed to read int0 cause: %d\n", ret); + return IRQ_HANDLED; + } + priv->npcm_state = cause & (PCM9211_INT0_MNPCM0_MASK | + PCM9211_INT0_MDTSCD0_MASK); + } + + ucontrol->value.integer.value[0] = (priv->npcm_state & mask) == mask; + + return 0; +} + +static int pcm9211_dir_preamble_kctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = 4; + + return 0; +} + +static int pcm9211_dir_preamble_kctl(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct pcm9211_priv *priv = snd_soc_component_get_drvdata(component); + struct device *dev = priv->dev; + + /* If we have an interrupt connected preamble is up-to-date */ + if (!priv->int0) + priv->dir_rate = pcm9211_read_burst_preamble(dev); + + memcpy(ucontrol->value.bytes.data, priv->burst_preamble, 4); + + return 0; +} + +static struct snd_kcontrol* pcm9211_get_ctl(struct device *dev, const char *name) +{ + struct pcm9211_priv *priv = dev_get_drvdata(dev); + struct snd_ctl_elem_id elem_id; + + memset(&elem_id, 0, sizeof(elem_id)); + elem_id.iface = SNDRV_CTL_ELEM_IFACE_PCM; + strcpy(elem_id.name, name); + return snd_ctl_find_id(priv->codec->component.card->snd_card, &elem_id); +} + +static struct snd_kcontrol* pcm9211_get_rate_ctl(struct device *dev) +{ + struct pcm9211_priv *priv = dev_get_drvdata(dev); + + if (!priv->rate_ctl) + priv->rate_ctl = pcm9211_get_ctl(dev, "DIR Sample Rate"); + + return priv->rate_ctl; +} + +static struct snd_kcontrol* pcm9211_get_npcm_ctl(struct device *dev) +{ + struct pcm9211_priv *priv = dev_get_drvdata(dev); + + if (!priv->npcm_ctl) + priv->npcm_ctl = pcm9211_get_ctl(dev, "DIR Non-PCM Bitstream"); + + return priv->npcm_ctl; +} + +static struct snd_kcontrol* pcm9211_get_dtscd_ctl(struct device *dev) +{ + struct pcm9211_priv *priv = dev_get_drvdata(dev); + + if (!priv->dts_ctl) + priv->dts_ctl = pcm9211_get_ctl(dev, "DIR DTS Bitstream"); + + return priv->dts_ctl; +} + +static struct snd_kcontrol* pcm9211_get_burst_preamble_ctl(struct device *dev) +{ + struct pcm9211_priv *priv = dev_get_drvdata(dev); + + if (!priv->preamble_ctl) + priv->preamble_ctl = pcm9211_get_ctl(dev, "DIR Burst Preamble"); + + return priv->preamble_ctl; +} + +static irqreturn_t pcm9211_interrupt(int irq, void *data) +{ + struct pcm9211_priv *priv = data; + struct device *dev = priv->dev; + + unsigned int cause; + int rate; + int ret; + + ret = regmap_read(priv->regmap, PCM9211_INT0_OUT, &cause); + if (ret) { + dev_err(dev, "Failed to read int0 cause: %d\n", ret); + return IRQ_HANDLED; + } + + if (cause & PCM9211_INT0_MFSCHG0_MASK) { + /* Interrupt is generated before the Fs calculation has + * finished. Give it time to settle. + */ + usleep_range(15000, 16000); + rate = pcm9211_dir_rate(dev); + + if (rate < 0) { + dev_err(dev, "Failed to retrieve DIR rate: %d\n", rate); + goto preamble; + } + + if (rate == priv->dir_rate) + goto preamble; + + priv->dir_rate = rate; + dev_dbg(dev, "DIR sampling rate changed to: %d\n", rate); + + if (priv->codec == NULL || pcm9211_get_rate_ctl(dev) == NULL) + goto preamble; + + snd_ctl_notify(priv->codec->component.card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, + &pcm9211_get_rate_ctl(dev)->id); + } + +preamble: + if ((cause & PCM9211_INT0_MPCRNW0_MASK)) { + if (pcm9211_read_burst_preamble(dev) < 0) + goto npcm; + + if (priv->codec == NULL || + pcm9211_get_burst_preamble_ctl(dev) == NULL) + goto dts; + + snd_ctl_notify(priv->codec->component.card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, + &pcm9211_get_burst_preamble_ctl(dev)->id); + } + +npcm: + if ((cause & PCM9211_INT0_MNPCM0_MASK)) { + /* PCM9211 does not generate an interrupt for NPCM0 1->0 + * transition, but continously generates interrupts as long as + * NPCM0 is high, so use a timeout to clear */ + cancel_delayed_work_sync(&priv->npcm_clear_work); + queue_delayed_work(system_wq, &priv->npcm_clear_work, + msecs_to_jiffies(100)); + + + if ((cause & PCM9211_INT0_MNPCM0_MASK) != + (priv->npcm_state & PCM9211_INT0_MNPCM0_MASK)) + dev_dbg(dev, "NPCM status on interrupt: %d\n", + (cause & PCM9211_INT0_MNPCM0_MASK) == + PCM9211_INT0_MNPCM0_MASK); + + priv->npcm_state = (priv->npcm_state & + ~PCM9211_INT0_MNPCM0_MASK) | + (cause & PCM9211_INT0_MNPCM0_MASK); + + if (priv->codec == NULL || pcm9211_get_npcm_ctl(dev) == NULL) + goto dts; + + snd_ctl_notify(priv->codec->component.card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, + &pcm9211_get_npcm_ctl(dev)->id); + } + +dts: + if (cause & PCM9211_INT0_MDTSCD0_MASK) { + dev_dbg(dev, "DTSCD status on interrupt: %d\n", + (cause & PCM9211_INT0_MDTSCD0_MASK) == + PCM9211_INT0_MDTSCD0_MASK); + priv->npcm_state |= PCM9211_INT0_MDTSCD0_MASK; + + if (priv->codec == NULL || pcm9211_get_dtscd_ctl(dev) == NULL) + return IRQ_HANDLED; + + snd_ctl_notify(priv->codec->component.card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, + &pcm9211_get_dtscd_ctl(dev)->id); + } + + return IRQ_HANDLED; +} + +static int pcm9211_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm9211_priv *priv = snd_soc_codec_get_drvdata(codec); + struct device *dev = codec->dev; + int port; + + port = pcm9211_main_output_port(dev); + if (port < 0) { + dev_err(dev, "Failed to read selected port: %d\n", port); + return port; + } + + if (port == PCM9211_MOSRC_ADC) + return snd_pcm_hw_constraint_list(substream->runtime, + 0, SNDRV_PCM_HW_PARAM_RATE, + &adc_rate_constraints); + + priv->dir_rate = pcm9211_dir_rate(dev); + priv->rate_constraints.count = 1; + priv->rate_constraints.list = &priv->dir_rate; + priv->rate_constraints.mask = 0; + + dev_dbg(dev, "Detected biphase rate is %d Hz\n", priv->dir_rate); + + return snd_pcm_hw_constraint_list(substream->runtime, + 0, SNDRV_PCM_HW_PARAM_RATE, &priv->rate_constraints); +} + +static int pcm9211_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm9211_priv *priv = snd_soc_codec_get_drvdata(codec); + struct device *dev = codec->dev; + + int ret; + + if (freq > PCM9211_MAX_SYSCLK) { + dev_err(dev, "System clock greater %d is not supported\n", + PCM9211_MAX_SYSCLK); + return -EINVAL; + } + + ret = clk_set_rate(priv->xti, freq); + if (ret) + return ret; + + priv->sysclk = freq; + + return 0; +} + +static int pcm9211_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int format) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm9211_priv *priv = snd_soc_codec_get_drvdata(codec); + struct device *dev = codec->dev; + u32 adfmt, dirfmt; + int ret; + + /* Configure format for ADC and DIR block, if main output source is + * set to AUTO the output port may switch between them at any time + */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + adfmt = PCM9211_ADFMT_I2S; + dirfmt = PCM9211_DIR_FMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + adfmt = PCM9211_ADFMT_RIGHT_J; + dirfmt = PCM9211_DIR_FMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + adfmt = PCM9211_ADFMT_LEFT_J; + dirfmt = PCM9211_DIR_FMT_LEFT_J; + break; + default: + dev_err(dev, "Unsupported DAI format\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, PCM9211_ADC_CTRL2, + PCM9211_ADFMT_MASK, adfmt << PCM9211_ADFMT_SHIFT); + if (ret) { + dev_err(dev, "Failed to update ADC format: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(priv->regmap, PCM9211_DIR_OUT_FMT, + PCM9211_DIR_FMT_MASK, dirfmt << PCM9211_DIR_FMT_SHIFT); + if (ret) { + dev_err(dev, "Failed to update ADC format: %d\n", ret); + return ret; + } + + return 0; +} + +static int pcm9211_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm9211_priv *priv = snd_soc_codec_get_drvdata(codec); + struct device *dev = codec->dev; + unsigned int sclk = 1; + unsigned int rate; + unsigned int bck; + unsigned int ratio; + unsigned int port; + int ret; + int i; + + rate = params_rate(params); + bck = rate * 64; + + port = pcm9211_main_output_port(dev); + if (port == PCM9211_MOSRC_ADC) { + switch (rate) { + case 48000: + sclk = 12288000; + break; + case 96000: + sclk = 24576000; + break; + default: + dev_err(dev, "Rate %d unsupported.\n", rate); + return -EINVAL; + } + + /* Systemclock setup */ + ratio = priv->sysclk / sclk; + for (i = 0; i < PCM9211_NUM_SCK_RATIOS; i++) { + if (pcm9211_sck_ratios[i] == ratio) + break; + } + if (i == PCM9211_NUM_SCK_RATIOS) { + dev_err(dev, "SCK divider %d is not supported\n", + ratio); + return -EINVAL; + } + ret = regmap_update_bits(priv->regmap, PCM9211_XTI_SCLK_FREQ, + PCM9211_XTI_XSCK_MASK, + i << PCM9211_XTI_XSCK_SHIFT); + + if (ret) { + dev_err(dev, "Failed to configure SCK divider: %d\n", + ret); + return ret; + } + + /* Bitclock setup */ + ratio = priv->sysclk / bck; + for (i = 0; i < PCM9211_NUM_BCK_RATIOS; i++) { + if (pcm9211_bck_ratios[i] == ratio) + break; + } + if (i == PCM9211_NUM_BCK_RATIOS) { + dev_err(dev, "BCK divider %d is not supported\n", + ratio); + return -EINVAL; + } + ret = regmap_update_bits(priv->regmap, PCM9211_XTI_SCLK_FREQ, + PCM9211_XTI_BCK_MASK, + i << PCM9211_XTI_BCK_SHIFT); + if (ret) { + dev_err(dev, "Failed to configure BCK divider: %d\n", + ret); + return ret; + } + + /* Frameclock setup */ + ratio = priv->sysclk / rate; + for (i = 0; i < PCM9211_NUM_LRCK_RATIOS; i++) { + if (pcm9211_lrck_ratios[i] == ratio) + break; + } + if (i == PCM9211_NUM_LRCK_RATIOS) { + dev_err(dev, "LRCK divider %d is not supported\n", + ratio); + return -EINVAL; + } + ret = regmap_update_bits(priv->regmap, PCM9211_XTI_SCLK_FREQ, + PCM9211_XTI_LRCK_MASK, + i << PCM9211_XTI_LRCK_SHIFT); + if (ret) { + dev_err(dev, "Failed to configure LRCK divider: %d\n", + ret); + return ret; + } + } + + return 0; +} + +static int pcm9211_reset(struct device *dev) +{ + struct pcm9211_priv *priv = dev_get_drvdata(dev); + int ret; + + /* Use reset gpio if available, otherwise soft-reset */ + if (priv->reset) { + gpiod_set_value_cansleep(priv->reset, 0); + usleep_range(500, 1000); + gpiod_set_value_cansleep(priv->reset, 1); + } else { + ret = regmap_update_bits(priv->regmap, PCM9211_SYS_RESET, + PCM9211_SYS_RESET_MRST, 0); + if (ret) { + dev_err(dev, "Could not reset device: %d\n", ret); + return ret; + } + usleep_range(10000, 15000); + } + + regcache_mark_dirty(priv->regmap); + + return 0; +} + +static int pcm9211_write_pinconfig(struct device *dev) +{ + struct pcm9211_priv *priv = dev_get_drvdata(dev); + u8 values[4]; + int val; + int ret; + int i; + + ret = of_property_read_u8_array(dev->of_node, "ti,group-function", + values, 3); + if (!ret) { + val = (values[0] & PCM9211_MPASEL_MASK) << + PCM9211_MPASEL_SHIFT | + (values[1] & PCM9211_MPBSEL_MASK) << + PCM9211_MPBSEL_SHIFT | + (values[2] & PCM9211_MPCSEL_MASK) << + PCM9211_MPCSEL_SHIFT; + ret = regmap_write(priv->regmap, PCM9211_MPIO_GROUP, val); + if (ret) { + dev_err(dev, "Failed to write mpio group functions: %d\n", + ret); + return ret; + } + } + + ret = of_property_read_u8_array(dev->of_node, "ti,mpio-a-flags-gpio", + values, 4); + if (!ret) { + /* Write MPIO A flags/gpio selection */ + for (i = 0, val = 0; i < 4; i++) + val |= (values[i] << PCM9211_MPAxSEL_SHIFT(i)) & + PCM9211_MPAxSEL_MASK(i); + + ret = regmap_update_bits(priv->regmap, PCM9211_MPIO_A_FLAGS, + PCM9211_MPAxSEL_MASK(0) | PCM9211_MPAxSEL_MASK(1) | + PCM9211_MPAxSEL_MASK(2), val); + if (ret) { + dev_err(dev, "Failed to update mpio_a flags: %d\n", ret); + return ret; + } + } + + ret = of_property_read_u8_array(dev->of_node, "ti,mpio-b-flags-gpio", + values, 4); + if (!ret) { + /* Write MPIO B flags/gpio selection */ + for (i = 0, val = 0; i < 4; i++) + val |= (values[i] << PCM9211_MPBxSEL_SHIFT(i)) & + PCM9211_MPBxSEL_MASK(i); + + ret = regmap_update_bits(priv->regmap, + PCM9211_MPIO_B_MPIO_C_FLAGS, + PCM9211_MPBxSEL_MASK(0) | + PCM9211_MPBxSEL_MASK(1) | + PCM9211_MPBxSEL_MASK(2), val); + if (ret) { + dev_err(dev, "Failed to update mpio_a flags: %d\n", + ret); + return ret; + } + } + + ret = of_property_read_u8_array(dev->of_node, "ti,mpio-c-flags-gpio", + values, 4); + if (!ret) { + /* Write MPIO B flags/gpio selection */ + for (i = 0, val = 0; i < 4; i++) + val |= (values[i] << PCM9211_MPCxSEL_SHIFT(i)) & + PCM9211_MPCxSEL_MASK(i); + + ret = regmap_update_bits(priv->regmap, + PCM9211_MPIO_B_MPIO_C_FLAGS, + PCM9211_MPCxSEL_MASK(0) | + PCM9211_MPCxSEL_MASK(1) | + PCM9211_MPCxSEL_MASK(2), val); + if (ret) { + dev_err(dev, "Failed to update mpio_a flags: %d\n", + ret); + return ret; + } + } + + ret = of_property_read_u8_array(dev->of_node, "ti,mpio-a-flag", + values, 4); + if (!ret) { + /* Write MPIO A flag selection */ + for (i = 0, val = 0; i < 2; i++) + val |= (values[i] << + PCM9211_MPIO_ABCx_FLAG_SHIFT(i)) & + PCM9211_MPIO_ABCx_FLAG_MASK(i); + ret = regmap_write(priv->regmap, PCM9211_MPIO_A1_A0_OUT_FLAG, + val); + if (ret) { + dev_err(dev, "Failed to update mpio_a1/0 flags: %d\n", + ret); + return ret; + } + + for (i = 2, val = 0; i < 4; i++) + val |= (values[i] << + PCM9211_MPIO_ABCx_FLAG_SHIFT(i)) & + PCM9211_MPIO_ABCx_FLAG_MASK(i); + ret = regmap_write(priv->regmap, PCM9211_MPIO_A3_A2_OUT_FLAG, + val); + if (ret) { + dev_err(dev, "Failed to update mpio_a3/2 flags: %d\n", + ret); + return ret; + } + } + + ret = of_property_read_u8_array(dev->of_node, "ti,mpio-b-flag", + values, 4); + if (!ret) { + /* Write MPIO B flag selection */ + for (i = 0, val = 0; i < 2; i++) + val |= (values[i] << PCM9211_MPIO_ABCx_FLAG_SHIFT(i)) & + PCM9211_MPIO_ABCx_FLAG_MASK(i); + ret = regmap_write(priv->regmap, PCM9211_MPIO_B1_B0_OUT_FLAG, + val); + if (ret) { + dev_err(dev, "Failed to update mpio_b1/0 flags: %d\n", + ret); + return ret; + } + + for (i = 2, val = 0; i < 4; i++) + val |= (values[i] << PCM9211_MPIO_ABCx_FLAG_SHIFT(i)) & + PCM9211_MPIO_ABCx_FLAG_MASK(i); + ret = regmap_write(priv->regmap, PCM9211_MPIO_B3_B2_OUT_FLAG, + val); + if (ret) { + dev_err(dev, "Failed to update mpio_b3/2 flags: %d\n", + ret); + return ret; + } + } + + ret = of_property_read_u8_array(dev->of_node, "ti,mpio-c-flag", + values, 4); + if (!ret) { + /* Write MPIO C flag selection */ + for (i = 0, val = 0; i < 2; i++) + val |= (values[i] << PCM9211_MPIO_ABCx_FLAG_SHIFT(i)) & + PCM9211_MPIO_ABCx_FLAG_MASK(i); + ret = regmap_write(priv->regmap, PCM9211_MPIO_C1_C0_OUT_FLAG, + val); + if (ret) { + dev_err(dev, "Failed to update mpio_c1/0 flags: %d\n", + ret); + return ret; + } + + for (i = 2, val = 0; i < 4; i++) + val |= (values[i] << PCM9211_MPIO_ABCx_FLAG_SHIFT(i)) & + PCM9211_MPIO_ABCx_FLAG_MASK(i); + ret = regmap_write(priv->regmap, PCM9211_MPIO_C3_C2_OUT_FLAG, + val); + if (ret) { + dev_err(dev, "Failed to update mpio_c3/2 flags: %d\n", + ret); + return ret; + } + } + + ret = of_property_read_u8_array(dev->of_node, "ti,mpo-function", + values, 2); + if (!ret) { + /* Write MPO function selection */ + for (i = 0, val = 0; i < 2; i++) + val |= (values[i] << PCM9211_MPOxOUT_SHIFT(i)) & + PCM9211_MPOxOUT_MASK(i); + ret = regmap_write(priv->regmap, PCM9211_MPO_1_0_FUNC, val); + if (ret) { + dev_err(dev, "Failed to update mpo function selection: %d\n", + ret); + return ret; + } + } + + ret = of_property_read_u8(dev->of_node, "ti,int0-function", values); + if (!ret) { + val = values[0] ? PCM9211_ERROR_INT0_MASK : 0; + ret = regmap_update_bits(priv->regmap, PCM9211_ERR_OUT, + PCM9211_ERROR_INT0_MASK, val); + if (ret) { + dev_err(dev, "Failed to update int0 function selection: %d\n", + ret); + return ret; + } + } + + ret = of_property_read_u8(dev->of_node, "ti,int1-function", values); + if (!ret) { + val = values[0] ? PCM9211_NPCM_INT1_MASK : 0; + ret = regmap_update_bits(priv->regmap, PCM9211_ERR_OUT, + PCM9211_NPCM_INT1_MASK, val); + if (ret) { + dev_err(dev, "Failed to update int1 function selection: %d\n", + ret); + return ret; + } + } + + return 0; +} + +static void pcm9211_npcm_clear_work(struct work_struct *work) +{ + struct pcm9211_priv *priv = container_of(work, struct pcm9211_priv, + npcm_clear_work.work); + u8 old_state = priv->npcm_state; + struct device *dev = priv->dev; + + /* Clear NPCM & DTSCD, as DTSCD is only valid as long as NPCM is */ + priv->npcm_state &= ~(PCM9211_INT0_MNPCM0_MASK | + PCM9211_INT0_MDTSCD0_MASK); + + dev_dbg(dev, "Clear NPCM flag after timeout\n"); + + if (priv->codec == NULL || pcm9211_get_dtscd_ctl(dev) == NULL || + pcm9211_get_npcm_ctl(dev) == NULL) + return; + + if (old_state & PCM9211_INT0_MNPCM0_MASK) + snd_ctl_notify(priv->codec->component.card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, + &pcm9211_get_npcm_ctl(dev)->id); + + if (old_state & PCM9211_INT0_MDTSCD0_MASK) + snd_ctl_notify(priv->codec->component.card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, + &pcm9211_get_dtscd_ctl(dev)->id); +} + +static int pcm9211_soc_probe(struct snd_soc_codec *codec) +{ + struct pcm9211_priv *priv = snd_soc_codec_get_drvdata(codec); + + priv->codec = codec; + + return 0; +} + +/* Simple Controls */ +static const DECLARE_TLV_DB_SCALE(pcm9211_adc_tlv, -10050, 50, 1); +static const char *const pcm9211_main_outputs[] = { "AUTO", "DIR", "ADC", + "AUXIN0", "AUXIN1", "AUXIN2" }; +static const struct soc_enum pcm9211_main_sclk_enum = + SOC_ENUM_SINGLE(PCM9211_MAIN_OUT_SOURCE, 4, 4, pcm9211_main_outputs); + +static const struct snd_kcontrol_new pcm9211_snd_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("ADC Attenuation", + PCM9211_ADC_L_CH_ATT, + PCM9211_ADC_R_CH_ATT, + 0, 14, 255, 0, pcm9211_adc_tlv), + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "DIR Sample Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = pcm9211_dir_rate_kctl_info, + .get = pcm9211_dir_rate_kctl, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "DIR Non-PCM Bitstream", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = pcm9211_dir_npcm_kctl_info, + .get = pcm9211_dir_npcm_kctl, + .private_value = PCM9211_INT0_MNPCM0_MASK, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "DIR DTS Bitstream", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = pcm9211_dir_dtscd_kctl_info, + .get = pcm9211_dir_dtscd_kctl, + .private_value = PCM9211_INT0_MDTSCD0_MASK, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "DIR Burst Preamble", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = pcm9211_dir_preamble_kctl_info, + .get = pcm9211_dir_preamble_kctl, + }, + SOC_ENUM("MAIN SCLK Output Select", pcm9211_main_sclk_enum), +}; + +/* DAPM Controls */ +static const char *const pcm9211_dir_inputs[] = { "RXIN0", "RXIN1", "RXIN2", + "RXIN3", "RXIN4", "RXIN5", "RXIN6", "RXIN7" }; +static const struct soc_enum pcm9211_dir_mux_enum = + SOC_ENUM_SINGLE(PCM9211_DIR_INP_BIPHASE, 0, 8, pcm9211_dir_inputs); +static const struct snd_kcontrol_new pcm9211_dir_mux_control = + SOC_DAPM_ENUM("DIR Input Select", pcm9211_dir_mux_enum); + +static const struct soc_enum pcm9211_main_out_enum = + SOC_ENUM_SINGLE(PCM9211_MAIN_OUT_SOURCE, 0, 4, pcm9211_main_outputs); +static const struct snd_kcontrol_new pcm9211_main_out_control = + SOC_DAPM_ENUM("MAIN Output Select", pcm9211_main_out_enum); + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget pcm9211_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("RXIN0"), + SND_SOC_DAPM_INPUT("RXIN1"), + SND_SOC_DAPM_INPUT("RXIN2"), + SND_SOC_DAPM_INPUT("RXIN3"), + SND_SOC_DAPM_INPUT("RXIN4"), + SND_SOC_DAPM_INPUT("RXIN5"), + SND_SOC_DAPM_INPUT("RXIN6"), + SND_SOC_DAPM_INPUT("RXIN7"), + SND_SOC_DAPM_INPUT("VINL"), + SND_SOC_DAPM_INPUT("VINR"), + + SND_SOC_DAPM_ADC("ADC", NULL, PCM9211_SYS_RESET, + PCM9211_SYS_RESET_ADDIS_SHIFT, 1), + + /* Processing */ + SND_SOC_DAPM_AIF_IN("DIR", NULL, 0, PCM9211_SYS_RESET, + PCM9211_SYS_RESET_RXDIS_SHIFT, 1), + SND_SOC_DAPM_MIXER("AUTO", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Internal routing */ + SND_SOC_DAPM_MUX("DIR Input Mux", SND_SOC_NOPM, 0, 0, + &pcm9211_dir_mux_control), + SND_SOC_DAPM_MUX("MAIN Output Mux", SND_SOC_NOPM, 0, 0, + &pcm9211_main_out_control), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("MAIN"), +}; + +/* DAPM Routing */ +static const struct snd_soc_dapm_route pcm9211_dapm_routes[] = { + { "DIR Input Mux", "RXIN0", "RXIN0" }, + { "DIR Input Mux", "RXIN1", "RXIN1" }, + { "DIR Input Mux", "RXIN2", "RXIN2" }, + { "DIR Input Mux", "RXIN3", "RXIN3" }, + { "DIR Input Mux", "RXIN4", "RXIN4" }, + { "DIR Input Mux", "RXIN5", "RXIN5" }, + { "DIR Input Mux", "RXIN6", "RXIN6" }, + { "DIR Input Mux", "RXIN7", "RXIN7" }, + + { "ADC", NULL, "VINL" }, + { "ADC", NULL, "VINR" }, + + { "DIR", NULL, "DIR Input Mux" }, + { "AUTO", NULL, "DIR" }, + { "AUTO", NULL, "ADC" }, + + { "MAIN Output Mux", "DIR", "DIR" }, + { "MAIN Output Mux", "ADC", "ADC" }, + { "MAIN Output Mux", "AUTO", "AUTO" }, + + { "MAIN", NULL, "MAIN Output Mux" }, +}; + +static struct snd_soc_dai_ops pcm9211_dai_ops = { + .startup = pcm9211_startup, + .hw_params = pcm9211_hw_params, + .set_sysclk = pcm9211_set_dai_sysclk, + .set_fmt = pcm9211_set_dai_fmt, +}; + +/* BCLK is always 64 * FS == 32 bit/channel */ +#define PCM9211_FORMATS SNDRV_PCM_FMTBIT_S32_LE +struct snd_soc_dai_driver pcm9211_dai = { + .name = "pcm9211-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = PCM9211_FORMATS, + }, + .ops = &pcm9211_dai_ops, +}; + +static const struct snd_soc_codec_driver pcm9211_driver = { + .probe = pcm9211_soc_probe, + .component_driver = { + .controls = pcm9211_snd_controls, + .num_controls = ARRAY_SIZE(pcm9211_snd_controls), + .dapm_widgets = pcm9211_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm9211_dapm_widgets), + .dapm_routes = pcm9211_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm9211_dapm_routes), + }, +}; + +int pcm9211_probe(struct device *dev, struct regmap *regmap) +{ + struct pcm9211_priv *priv; + unsigned int cause; + int ret; + int i; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + dev_set_drvdata(dev, priv); + priv->dev = dev; + priv->regmap = regmap; + + priv->xti = devm_clk_get(dev, "xti"); + if (IS_ERR(priv->xti)) { + ret = PTR_ERR(priv->xti); + if (ret != -EPROBE_DEFER) + dev_err(dev, "Failed to get clock 'xti': %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(priv->xti); + if (ret) { + dev_err(dev, "Failed to enable xti clock: %d\n", ret); + return ret; + } + + priv->sysclk = clk_get_rate(priv->xti); + if (priv->sysclk > PCM9211_MAX_SYSCLK) { + dev_err(dev, "xti clock rate (%lu) exceeds supported max %u\n", + priv->sysclk, PCM9211_MAX_SYSCLK); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(priv->supplies); i++) + priv->supplies[i].supply = pcm9211_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "Failed to get supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + priv->reset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_HIGH); + if (IS_ERR(priv->reset)) { + ret = PTR_ERR(priv->reset); + dev_err(dev, "Failed to get reset gpio: %d\n", ret); + return ret; + } + + pcm9211_reset(dev); + + priv->int0 = devm_gpiod_get_optional(dev, "int0", GPIOD_IN); + if (IS_ERR(priv->reset)) { + ret = PTR_ERR(priv->reset); + dev_err(dev, "Failed to get reset gpio: %d\n", ret); + return ret; + } + + if (priv->int0) { + int irq = gpiod_to_irq(priv->int0); + + if (irq < 0) { + dev_err(dev, "Configured 'int0' gpio cannot be used as IRQ: %d\n", + irq); + return irq; + } + + INIT_DELAYED_WORK(&priv->npcm_clear_work, + pcm9211_npcm_clear_work); + ret = devm_request_threaded_irq(dev, irq, NULL, + pcm9211_interrupt, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "pcm9211", priv); + if (ret) { + dev_err(dev, "Failed to request irq: %d\n", ret); + return ret; + } + + /* Set interrupt to use positive polarity */ + ret = regmap_update_bits(priv->regmap, PCM9211_INT_POLARITY, + PCM9211_INT0_POLARITY_POS_MASK, + PCM9211_INT0_POLARITY_POS_MASK); + if (ret) { + dev_err(dev, "Failed to configure int0 polaroty: %d\n", + ret); + return ret; + } + } + + ret = pcm9211_write_pinconfig(dev); + if (ret) + return ret; + + /* Unmap NPCM, DTS, Burst Preamble and Fs change interrupt */ + ret = regmap_update_bits(priv->regmap, PCM9211_INT0_CAUSE, + PCM9211_INT0_MNPCM0_MASK | PCM9211_INT0_MDTSCD0_MASK | + PCM9211_INT0_MPCRNW0_MASK | PCM9211_INT0_MFSCHG0_MASK, + 0); + if (ret) { + dev_err(dev, "Failed to unmask interrupt causes: %d\n", ret); + return ret; + } + + /* Enable DTSCD detection */ + ret = regmap_update_bits(priv->regmap, PCM9211_NON_PCM_DEF, + PCM9211_NON_PCM_DTS_CD_DET_MASK, + PCM9211_NON_PCM_DTS_CD_DET_MASK); + if (ret) { + dev_err(dev, "Failed to enable DTSCD detection: %d\n", ret); + return ret; + } + + /* Read initial sampling rate and npcm state */ + priv->dir_rate = pcm9211_dir_rate(dev); + ret = regmap_read(priv->regmap, PCM9211_INT0_OUT, &cause); + if (ret) { + dev_err(dev, "Failed to read int0 cause: %d\n", ret); + return IRQ_HANDLED; + } + priv->npcm_state = cause; + + ret = snd_soc_register_codec(dev, &pcm9211_driver, &pcm9211_dai, 1); + if (ret) { + dev_err(dev, "Failed to register codec: %d\n", ret); + return ret; + } + + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + pm_runtime_idle(dev); + + return 0; +} +EXPORT_SYMBOL_GPL(pcm9211_probe); + +void pcm9211_remove(struct device *dev) +{ + struct pcm9211_priv *priv = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + pm_runtime_disable(dev); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + clk_disable_unprepare(priv->xti); +} +EXPORT_SYMBOL_GPL(pcm9211_remove); + +#ifdef CONFIG_PM +static int pcm9211_runtime_resume(struct device *dev) +{ + struct pcm9211_priv *priv = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(priv->xti); + if (ret) { + dev_err(dev, "Failed to enable xti clock: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies); + if (ret) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + goto err_reg; + } + + ret = pcm9211_reset(dev); + if (ret) { + dev_err(dev, "Failed to reset device: %d\n", ret); + goto err; + } + + regcache_cache_only(priv->regmap, false); + regcache_mark_dirty(priv->regmap); + + ret = regcache_sync(priv->regmap); + if (ret) { + dev_err(dev, "Failed to sync regmap: %d\n", ret); + goto err; + } + + return 0; + +err: + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); +err_reg: + clk_disable_unprepare(priv->xti); + + return ret; +} + +static int pcm9211_runtime_suspend(struct device *dev) +{ + struct pcm9211_priv *priv = dev_get_drvdata(dev); + + regcache_cache_only(priv->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); + clk_disable_unprepare(priv->xti); + + return 0; +} +#endif + +const struct dev_pm_ops pcm9211_pm_ops = { + SET_RUNTIME_PM_OPS(pcm9211_runtime_suspend, pcm9211_runtime_resume, + NULL) +}; +EXPORT_SYMBOL_GPL(pcm9211_pm_ops); + +MODULE_DESCRIPTION("PCM9211 codec driver"); +MODULE_AUTHOR("Julian Scheel <julian(a)jusst.de>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm9211.h b/sound/soc/codecs/pcm9211.h new file mode 100644 index 000000000000..3ad9e9e9419a --- /dev/null +++ b/sound/soc/codecs/pcm9211.h @@ -0,0 +1,206 @@ +/* + * PCM9211 codec driver header + * + * Copyright (C) 2017 jusst technologies GmbH / jusst.engineering + * + * Author: Julian Scheel <julian(a)jusst.de> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + */ + +#ifndef __PCM9211_H_ +#define __PCM9211_H_ + +extern const struct dev_pm_ops pcm9211_pm_ops; +extern const struct regmap_config pcm9211_regmap; + +extern int pcm9211_probe(struct device *dev, struct regmap *regmap); +extern void pcm9211_remove(struct device *dev); + +/* Register definitions */ +#define PCM9211_ERR_OUT 0x20 +#define PCM9211_DIR_INITIAL1 0x21 +#define PCM9211_DIR_INITIAL2 0x22 +#define PCM9211_DIR_INITIAL3 0x23 +#define PCM9211_OSC_CTRL 0x24 +#define PCM9211_ERR_CAUSE 0x25 +#define PCM9211_AUTO_SEL_CAUSE 0x26 +#define PCM9211_DIR_FS_RANGE 0x27 +#define PCM9211_NON_PCM_DEF 0x28 +#define PCM9211_DTS_CD_LD 0x29 +#define PCM9211_INT0_CAUSE 0x2a +#define PCM9211_INT1_CAUSE 0x2b +#define PCM9211_INT0_OUT 0x2c +#define PCM9211_INT1_OUT 0x2d +#define PCM9211_INT_POLARITY 0x2e +#define PCM9211_DIR_OUT_FMT 0x2f +#define PCM9211_DIR_RSCLK_RATIO 0x30 +#define PCM9211_XTI_SCLK_FREQ 0x31 +#define PCM9211_DIR_SOURCE_BIT2 0x32 +#define PCM9211_XTI_SOURCE_BIT2 0x33 +#define PCM9211_DIR_INP_BIPHASE 0x34 +#define PCM9211_RECOUT0_BIPHASE 0x35 +#define PCM9211_RECOUT1_BIPHASE 0x36 +#define PCM9211_FS_CALC_TARGET 0x37 +#define PCM9211_FS_CALC_RESULT 0x38 +#define PCM9211_BIPHASE_INFO 0x39 +#define PCM9211_PC_BUF0 0x3a +#define PCM9211_PC_BUF1 0x3b +#define PCM9211_PD_BUF0 0x3c +#define PCM9211_PD_BUF1 0x3d + +#define PCM9211_SYS_RESET 0x40 + +#define PCM9211_ADC_CTRL1 0x42 + +#define PCM9211_ADC_L_CH_ATT 0x46 +#define PCM9211_ADC_R_CH_ATT 0x47 +#define PCM9211_ADC_CTRL2 0x48 +#define PCM9211_ADC_CTRL3 0x49 + +#define PCM9211_DIR_STATUS1 0x5a +#define PCM9211_DIR_STATUS2 0x5b +#define PCM9211_DIR_STATUS3 0x5c +#define PCM9211_DIR_STATUS4 0x5d +#define PCM9211_DIR_STATUS5 0x5e +#define PCM9211_DIR_STATUS6 0x5f +#define PCM9211_DIT_CTRL1 0x60 +#define PCM9211_DIT_CTRL2 0x61 +#define PCM9211_DIT_CTRL3 0x62 +#define PCM9211_DIT_STATUS1 0x63 +#define PCM9211_DIT_STATUS2 0x64 +#define PCM9211_DIT_STATUS3 0x65 +#define PCM9211_DIT_STATUS4 0x66 +#define PCM9211_DIT_STATUS5 0x67 +#define PCM9211_DIT_STATUS6 0x68 + +#define PCM9211_MAIN_AUX_MUTE 0x6a +#define PCM9211_MAIN_OUT_SOURCE 0x6b +#define PCM9211_AUX_OUT_SOURCE 0x6c +#define PCM9211_MPIO_B_MAIN_HIZ 0x6d +#define PCM9211_MPIO_C_MPIO_A_HIZ 0x6e +#define PCM9211_MPIO_GROUP 0x6f +#define PCM9211_MPIO_A_FLAGS 0x70 +#define PCM9211_MPIO_B_MPIO_C_FLAGS 0x71 +#define PCM9211_MPIO_A1_A0_OUT_FLAG 0x72 +#define PCM9211_MPIO_A3_A2_OUT_FLAG 0x73 +#define PCM9211_MPIO_B1_B0_OUT_FLAG 0x74 +#define PCM9211_MPIO_B3_B2_OUT_FLAG 0x75 +#define PCM9211_MPIO_C1_C0_OUT_FLAG 0x76 +#define PCM9211_MPIO_C3_C2_OUT_FLAG 0x77 +#define PCM9211_MPO_1_0_FUNC 0x78 +#define PCM9211_MPIO_A_B_DIR 0x79 +#define PCM9211_MPIO_C_DIR 0x7a +#define PCM9211_MPIO_A_B_DATA_OUT 0x7b +#define PCM9211_MPIO_C_DATA_OUT 0x7c +#define PCM9211_MPIO_A_B_DATA_IN 0x7d +#define PCM9211_MPIO_C_DATA_IN 0x7e + + +/* Register field values (only used ones) */ + +/* PCM9211_ERR_OUT */ +#define PCM9211_ERROR_INT0_MASK BIT(2) +#define PCM9211_NPCM_INT1_MASK BIT(0) + +/* PCM9211_NON_PCM_DEF */ +#define PCM9211_NON_PCM_DTS_CD_DET_MASK BIT(2) + +/* PCM9211_INT0_CAUSE */ +#define PCM9211_INT0_MNPCM0_MASK BIT(6) +#define PCM9211_INT0_MDTSCD0_MASK BIT(4) +#define PCM9211_INT0_MPCRNW0_MASK BIT(2) +#define PCM9211_INT0_MFSCHG0_MASK BIT(1) + +/* PCM9211_INT_POLARITY */ +#define PCM9211_INT0_POLARITY_POS_MASK BIT(2) + +/* PCM9211_DIR_OUT_FMT */ +#define PCM9211_DIR_FMT_MASK 0x7 +#define PCM9211_DIR_FMT_SHIFT 0 + +#define PCM9211_DIR_FMT_I2S 4 +#define PCM9211_DIR_FMT_LEFT_J 5 +#define PCM9211_DIR_FMT_RIGHT_J 0 + +/* PCM9211_XTI_SCLK_FREQ */ +#define PCM9211_XTI_XSCK_SHIFT 4 +#define PCM9211_XTI_XSCK_MASK (0x3 << PCM9211_XTI_XSCK_SHIFT) +#define PCM9211_XTI_BCK_SHIFT 2 +#define PCM9211_XTI_BCK_MASK (0x3 << PCM9211_XTI_BCK_SHIFT) +#define PCM9211_XTI_LRCK_SHIFT 0 +#define PCM9211_XTI_LRCK_MASK (0x3 << PCM9211_XTI_LRCK_SHIFT) + + +/* PCM9211_BIPHASE_INFO */ +#define PCM9211_BIPHASE_SFSST_SHIFT 7 +#define PCM9211_BIPHASE_SFSST_MASK BIT(PCM9211_BIPHASE_SFSST_SHIFT) +#define PCM9211_BIPHASE_SFSOUT_SHIFT 0 +#define PCM9211_BIPHASE_SFSOUT_MASK (0xf << PCM9211_BIPHASE_SFSOUT_SHIFT) + + +/* PCM9211_SYS_RESET */ +#define PCM9211_SYS_RESET_MRST BIT(7) +#define PCM9211_SYS_RESET_RXDIS_SHIFT 4 +#define PCM9211_SYS_RESET_ADDIS_SHIFT 5 + + +/* PCM9211_ADC_CTRL2 */ +#define PCM9211_ADFMT_MASK 0x3 +#define PCM9211_ADFMT_SHIFT 0 + +#define PCM9211_ADFMT_I2S 0 +#define PCM9211_ADFMT_LEFT_J 1 +#define PCM9211_ADFMT_RIGHT_J 2 + + +/* PCM9211_MAIN_OUT_SOURCE */ +#define PCM9211_MOSSRC_SHIFT 4 +#define PCM9211_MOPSRC_SHIFT 0 +#define PCM9211_MOSRC_MASK 0x7 +#define PCM9211_MOSSRC_MASK \ + (PCM9211_MOSRC_MASK << PCM9211_MOSSRC_SHIFT) +#define PCM9211_MOPSRC_MASK \ + (PCM9211_MOSRC_MASK << PCM9211_MOPSRC_SHIFT) + +#define PCM9211_MOSRC_AUTO 0 +#define PCM9211_MOSRC_DIR 1 +#define PCM9211_MOSRC_ADC 2 +#define PCM9211_MOSRC_AUXIN0 3 +#define PCM9211_MOSRC_AUXIN1 4 +#define PCM9211_MOSRC_AUXIN2 5 + + +/* PCM9211_MPIO_GROUP */ +#define PCM9211_MPASEL_SHIFT 6 +#define PCM9211_MPASEL_MASK (0x2 << PCM9211_MPASEL_SHIFT) +#define PCM9211_MPBSEL_SHIFT 3 +#define PCM9211_MPBSEL_MASK (0x7 << PCM9211_MPBSEL_SHIFT) +#define PCM9211_MPCSEL_SHIFT 0 +#define PCM9211_MPCSEL_MASK (0x7 << PCM9211_MPCSEL_SHIFT) + + +/* PCM9211_MPIO_A_FLAGS */ +#define PCM9211_MPAxSEL_SHIFT(x) (x) +#define PCM9211_MPAxSEL_MASK(x) BIT(PCM9211_MPAxSEL_SHIFT(x)) + + +/* PCM9211_MPIO_B_MPIO_C_FLAGS */ +#define PCM9211_MPBxSEL_SHIFT(x) ((x) + 4) +#define PCM9211_MPBxSEL_MASK(x) BIT(PCM9211_MPBxSEL_SHIFT(x)) +#define PCM9211_MPCxSEL_SHIFT(x) (x) +#define PCM9211_MPCxSEL_MASK(x) BIT(PCM9211_MPCxSEL_SHIFT(x)) + + +/* PCM9211_MPIO_A1_A0_OUT_FLAG..PCM9211_MPIO_C3_C2_OUT_FLAG */ +#define PCM9211_MPIO_ABCx_FLAG_SHIFT(x) (((x) % 2) * 4) +#define PCM9211_MPIO_ABCx_FLAG_MASK(x) (0xf << PCM9211_MPIO_ABCx_FLAG_SHIFT(x)) + + +/* PCM9211_MPO_1_0_FUNC */ +#define PCM9211_MPOxOUT_SHIFT(x) ((x) * 4) +#define PCM9211_MPOxOUT_MASK(x) (0xf << PCM9211_MPOxOUT_SHIFT(x)) + +#endif -- 2.12.2

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[alsa-devel] HR timer 1000Hz limit since kernel 4.4.0-77
by Tim 24 May '17

24 May '17

Hello, ever since kernel 4.4.0-77, there seems to be a limit on the ALSA HR timer, I can't set it above 1000Hz (snd_timer_params() fails with -22 invalid argument). I used to be able to set the rate much higher than that. Been observed in two completely different distros, one with a much newer kernel. How can I overcome this limit? Thanks.

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[alsa-devel] [PATCH v4 0/3] ASoC: Add STM32 I2S support
by olivier moysan 23 May '17

23 May '17

This patch-set handles the SPI/I2S IP on STM32 platforms. It applies to STM32 platforms implementing version 2 of SPI/I2S IP like STM32H7 series. The SPI/I2S block is a serial peripheral interface (SPI), which can also be configured to work on I2S/PCM mode. ASoC STM32 I2S driver only supports this I2S/PCM mode. The I2S/PCM interface can either support full duplex or half-duplex communication (transmitter or receiver only). The change of mode requires to stop the interface. This means that the two audio paths cannot be basically managed independently. The driver supports playback and capture through a single bidirectionnal DAI. In the first version of the driver, playback and capture paths are made mutually exclusive. A second patch introduces support of full duplex mode. In this implementation the interface is systematically configured in full-duplex mode from hardware point of view. So, when managing a single audio path, the interface is not configured as transmit or receive only, as it should be. This restriction is handled by masking underrun/overrun errors from the unused path. The I2S/PCM interface supports four audio standards: I2S Philips standard, MSB justified standard, LSB justified standard, PCM standard PCM standard is declined in two versions, PCM short and long, according frame synchronization duration. These standards can be mapped to ASoC standards as follows: - I2S: i2s - MSB justified standard: left_j - LSB justified standard: right_j - PCM short: dsp_a PCM long format is not exposed as it does not match supported ASoC standards. v3 -> v4 - update example in dt bindings olivier moysan (3): dt-bindings: Document STM32 I2S bindings ASoC: stm32: Add I2S driver ASoC: stm32: Add full duplex support to i2s .../devicetree/bindings/sound/st,stm32-i2s.txt | 62 ++ sound/soc/stm/Kconfig | 2 +- sound/soc/stm/Makefile | 4 + sound/soc/stm/stm32_i2s.c | 946 +++++++++++++++++++++ 4 files changed, 1013 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.txt create mode 100644 sound/soc/stm/stm32_i2s.c -- 1.9.1

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