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

[alsa-devel] [alsa-lib][PATCH 0/7] fix compiler warnings
by Takashi Sakamoto 11 Jul '16

11 Jul '16

Hi, This patchset is just for fixes of compiler warnings. Nothing others. Takashi Sakamoto (7): topology: remove unused variable in tplg_parse_fe_dai() topology: remove unused variable in get_tokens() topology: apply the same type to compared two variables topology: initialize variable for error output pcm: set error code at failure path mixer: remove assignment to unused variable pcm: add 'default' case for switch statement modules/mixer/simple/sbase.c | 2 -- src/pcm/pcm_direct.c | 2 ++ src/pcm/pcm_hw.c | 1 + src/topology/data.c | 10 +++++----- src/topology/pcm.c | 1 - 5 files changed, 8 insertions(+), 8 deletions(-) -- 2.7.4

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Re: [alsa-devel] ALSA sound core device deinit crash
by Takashi Iwai 11 Jul '16

11 Jul '16

(Please don't drop Cc to ML) On Fri, 08 Jul 2016 07:35:37 +0200, b_lkasam(a)codeaurora.org wrote: > > On 2016-07-08 10:58, Takashi Iwai wrote: > > On Fri, 08 Jul 2016 07:19:05 +0200, > > b_lkasam(a)codeaurora.org wrote: > >> > >> Hi Alsa team, > >> There is kernel crash observed when soundcard register failure case as > >> below -> > >> > >> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c old mode > >> 100644 new mode 100755 index 0495890..60a1eb0 > >> --- a/sound/soc/soc-core.c > >> +++ b/sound/soc/soc-core.c > >> @@ -1382,6 +1382,7 @@ static int soc_probe_link_dais(struct > >> snd_soc_card *card, int num, int order) > >> /* do machine specific initialization */ if (dai_link->init) { ret = > >> dai_link->init(rtd); > >> + ret = -ENODEV; // -> we can force error here to reproduce crash > >> easily. > >> if (ret < 0) { > >> dev_err(card->dev, "ASoC: failed to init %s: %d\n", dai_link->name, > >> ret); > >> > >> If sound card fails at initialize at above, it is crashing in > >> pcm_chmap_ctl_private_free(). > >> > >> <1>[ 40.646642] [01-01-2016 00:07:16 CPU:0x3] Unable to handle > >> kernel > >> paging request at virtual address ffffffc0da644b68 > >> <1>[ 40.646664] [01-01-2016 00:07:16 CPU:0x3] pgd = ffffffc001d28000 > >> <1>[ 40.646673] [01-01-2016 00:07:16 CPU:0x3] [ffffffc0da644b68] > >> *pgd=00000000857fc003, *pud=00000000857fc003, *pmd=000000017ddd8003, > >> *pte=00c000015a644793 > >> <0>[ 40.646697] [01-01-2016 00:07:16 CPU:0x3] Internal error: Oops: > >> 9600004f [#1] PREEMPT SMP > >> <6>[ 40.646708] [01-01-2016 00:07:16 CPU:0x3] Modules linked in: > >> brcm_bt_drv fm_drv brcm_hci_ldisc texfat(PO) > >> <6>[ 40.646735] [01-01-2016 00:07:16 CPU:0x3] CPU: 3 PID: 299 Comm: > >> kworker/u8:8 Tainted: P O 3.18.24-g41b2dda2-00002-gbe25a74 > >> #1 > >> <6>[ 40.646744] [01-01-2016 00:07:16 CPU:0x3] Hardware name: > >> Qualcomm > >> Technologies, Inc. MSM 8996 v3.x + PMI8996 MTP (DT) > >> <6>[ 40.646763] [01-01-2016 00:07:16 CPU:0x3] Workqueue: deferwq > >> deferred_probe_work_func > >> <6>[ 40.646773] [01-01-2016 00:07:16 CPU:0x3] task: ffffffc0e951c880 > >> ti: ffffffc03368c000 task.ti: ffffffc03368c000 > >> <6>[ 40.646787] [01-01-2016 00:07:16 CPU:0x3] PC is at > >> pcm_chmap_ctl_private_free+0x1c/0x2c > >> <6>[ 40.646798] [01-01-2016 00:07:16 CPU:0x3] LR is at > >> snd_ctl_free_one+0x20/0x34 > >> > >> > >> FIX: > >> > >> Can you look at the change below and share your comments on this? > >> diff --git a/sound/core/device.c b/sound/core/device.c old mode 100644 > >> new mode 100755 index 41bec30..eaffde1 > >> --- a/sound/core/device.c > >> +++ b/sound/core/device.c > >> @@ -219,6 +219,7 @@ void snd_device_free_all(struct snd_card *card) > >> > >> if (snd_BUG_ON(!card)) > >> return; > >> - list_for_each_entry_safe_reverse(dev, next, &card->devices, > >> list) > >> + list_for_each_entry_safe(dev, next, &card->devices, > >> list) > >> __snd_device_free(dev); } > >> > >> > >> Since control sound device has the lowest type value > >> (SNDRV_DEV_CONTROL), it will be the first entry linked in the > >> card->devices linked list head and will be the last one to be freed. > >> > >> This issue seems to be resolved by modifying the sequence the sound > >> devices in the card->devices list are freed as shown below (from > >> “prev” > >> direction to “next” direction) but I’m not sure if this is a right > >> approach from ALSA perspective. > > > > This doesn't look correct. The strange thing is that this error > > shouldn't happen no matter which free loop direction is. The chmap > > ctl should have been already removed by the disconnection before > > freeing. It means that either the disconnection isn't done properly > > or something else is missing. > > > > Could you give the full stack trace? It's important to know which > > code path triggers it. > > > > > > thanks, > > > > Takashi > > > Hi Takashi, > > Here is full stack trace --> > > > [Callstack] > : (struct snd_pcm_chmap *)info = 0xFFFFFFC0DA6A5200 > : (struct snd_pcm *)pcm = 0xFFFFFFC0DA644A80 //part of buddy page, > read-only > : info->stream = 0 > : snd_card_free() was called > > -012|pcm_chmap_ctl_private_free() > //info->pcm->streams[info->stream].chmap_kctl = NULL > -013|snd_ctl_free_one() > -014|snd_ctl_remove() > -015|snd_ctl_dev_free() > -016|__snd_device_free() > -017|snd_device_free_all() > -018|snd_card_do_free(inline) > -018|release_card_device() > -019|device_release() > -020|kobject_cleanup(inline) > -020|kobject_release() > -021|kobject_put() > -022|put_device() > -023|snd_card_free_when_closed() > -024|snd_card_free() > -025|snd_soc_instantiate_card(inline) -> //Here instantiate card > failed for some reason, then triggers snd_card_free > -025|snd_soc_register_card() > -026|msm8996_asoc_machine_probe() > -027|platform_drv_probe() > -028|really_probe(inline) > -028|driver_probe_device() > -029|__device_attach() > -030|bus_for_each_drv() > -031|device_attach() > -032|bus_probe_device() > -033|deferred_probe_work_func() > -034|static_key_count(inline) > -034|static_key_false(inline) > -034|trace_workqueue_execute_end(inline) > -034|process_one_work() > -035|worker_thread() > -036|kthread() > -037|ret_from_fork(asm) > ---|end of frame > > > ----------------------- > trigger point in soc-core.c > in API snd_soc_instantiate_card(), > > card_probe_error: > if (card->remove) > card->remove(card); > > snd_card_free(card->snd_card); -> this is > snd_card_free which internally leads to above function call. > ------------------------------------------ OK, thanks. So this is the case where it frees before registering, and indeed there is a bug in the PCM chmap code. It's freed at disconnection but the disconnect is called only when it was registered. Below is a quick fix. Give it a try. thanks, Takashi --- diff --git a/sound/core/control.c b/sound/core/control.c index 9ff081cd03f4..fb096cb20a80 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -160,6 +160,8 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, if (snd_BUG_ON(!card || !id)) return; + if (card->shutdown) + return; read_lock(&card->ctl_files_rwlock); #if IS_ENABLED(CONFIG_SND_MIXER_OSS) card->mixer_oss_change_count++; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 308c9ecf73db..8e980aa678d0 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -849,6 +849,14 @@ int snd_pcm_new_internal(struct snd_card *card, const char *id, int device, } EXPORT_SYMBOL(snd_pcm_new_internal); +static void free_chmap(struct snd_pcm_str *pstr) +{ + if (pstr->chmap_kctl) { + snd_ctl_remove(pstr->pcm->card, pstr->chmap_kctl); + pstr->chmap_kctl = NULL; + } +} + static void snd_pcm_free_stream(struct snd_pcm_str * pstr) { struct snd_pcm_substream *substream, *substream_next; @@ -871,6 +879,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) kfree(setup); } #endif + free_chmap(pstr); if (pstr->substream_count) put_device(&pstr->dev); } @@ -1135,10 +1144,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) for (cidx = 0; cidx < 2; cidx++) { if (!pcm->internal) snd_unregister_device(&pcm->streams[cidx].dev); - if (pcm->streams[cidx].chmap_kctl) { - snd_ctl_remove(pcm->card, pcm->streams[cidx].chmap_kctl); - pcm->streams[cidx].chmap_kctl = NULL; - } + free_chmap(&pcm->streams[cidx]); } mutex_unlock(&pcm->open_mutex); mutex_unlock(&register_mutex);

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[alsa-devel] [alsa-lib][PATCH] hctl: remove assertion from element event handler because removal event overwrites all of queued events
by Takashi Sakamoto 11 Jul '16

11 Jul '16

In ALSA control core, when several events occurs for an element, they're represented bit mask (struct snd_ctl_event.data.elem.mask). Thus userspace applications can handles these events separately in one I/O operation. There's an exception; removal event. This is represented by all of bits in the mask. Therefore, when a removal event occurs for an element, all of queued events for the element are overwritten. This works worse for a combination of applications which add/remove control elements and applications which keep a cache of elements. For example, let's assume a case that: 1.some elements are added by ioctl(SNDRV_CTL_IOCTL_ELEM_ADD). 2.ALSA control core handles the request and queues 'add' event. 3.applications don't read the event yet. 4.the elements are removed by ioctl(SNDRV_CTL_IOCTL_ELEM_REMOVE) 5.ALSA control core handles the request and queues 'removal' event by overwriting queued events. 6.applications read the event at last. 7.the applications cannot find removed elements in their cache data. Especially, when applications use hctl interface of ALSA userspace library, this situation occurs because the interface is designed to keep element cache implicitly inner the library. PulseAudio is such an application. When executing a test program of ALSA library (test/user-ctl-element-set.c), pulseaudio process aborts due to assertion at calls of snd_hctl_handle_events(). The rule to represent events is in ALSA kernel/userspace interfaces (UAPI header) and it's difficult to improve the rule. Therefore, developers should pay enough attention to the fact that userspace applications can receive removal events for elements which are not notified with 'add' event. For this reason, this commit removes assertion from event handler of ALSA hctl API. CC: pulseaudio-discuss(a)lists.freedesktop.org Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/hcontrol.c | 1 - 1 file changed, 1 deletion(-) diff --git a/src/control/hcontrol.c b/src/control/hcontrol.c index 7645c57..c40a807 100644 --- a/src/control/hcontrol.c +++ b/src/control/hcontrol.c @@ -742,7 +742,6 @@ static int snd_hctl_handle_event(snd_hctl_t *hctl, snd_ctl_event_t *event) if (event->data.elem.mask == SNDRV_CTL_EVENT_MASK_REMOVE) { int dir; res = _snd_hctl_find_elem(hctl, &event->data.elem.id, &dir); - assert(res >= 0 && dir == 0); if (res < 0 || dir != 0) return -ENOENT; snd_hctl_elem_remove(hctl, (unsigned int) res); -- 2.7.4

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[alsa-devel] [alsa-lib][PATCH] ctl: improve documentation of APIs to add an element set
by Takashi Sakamoto 11 Jul '16

11 Jul '16

Currently, documentation of APIs to add an element set describes causes of errors partly. But it gives an opinion to describe them thoroughly. This is not good for users. This commit changes the documentation so that it partly describes the causes. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index 85485aa..9bb8021 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -358,7 +358,8 @@ static bool validate_element_member_dimension(snd_ctl_elem_info_t *info) * <dt>-EBUSY * <dd>A element with ID \a id already exists. * <dt>-EINVAL - * <dd>ID has no name, or the number of members is not between 1 to 127. + * <dd>Some arguments include invalid value; i.e. ID field in \a info has no + * name, or the number of members is not between 1 to 127. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user elements. * <dt>-ENXIO @@ -446,7 +447,8 @@ int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * <dt>-EBUSY * <dd>A element with ID \a id already exists. * <dt>-EINVAL - * <dd>ID has no name, or the number of members is not between 1 to 127. + * <dd>Some arguments include invalid value; i.e. ID has no name, or the number + * of members is not between 1 to 127. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user elements. * <dt>-ENXIO @@ -530,7 +532,8 @@ int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * <dt>-EBUSY * <dd>A element with ID \a id already exists. * <dt>-EINVAL - * <dd>ID has no name, or the number of members is not between 1 to 127. + * <dd>Some parameters include invalid value; i.e. ID has no name, or the number + * of members is not between 1 to 127. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user elements. * <dt>-ENXIO @@ -591,9 +594,10 @@ int snd_ctl_elem_add_boolean_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * <dt>-EBUSY * <dd>A control element with ID \a id already exists. * <dt>-EINVAL - * <dd>\a element_count is not between 1 to 127, or \a items is not at least - * one, or a string in \a labels is empty, or longer than 63 bytes, or total - * length of the labels requires more than 64 KiB storage. + * <dd>Some arguments include invalid value; i.e. \a element_count is not + * between 1 to 127, or \a items is not at least one, or a string in \a + * labels is empty, or longer than 63 bytes, or total length of the labels + * requires more than 64 KiB storage. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user control elements. * <dt>-ENXIO @@ -677,7 +681,8 @@ int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * <dt>-EBUSY * <dd>A element with ID \a id already exists. * <dt>-EINVAL - * <dd>ID has no name, or the number of members is not between 1 to 511. + * <dd>Some arguments include invalid value; i.e. ID has no name, or the number + * of members is not between 1 to 511. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user elements. * <dt>-ENXIO -- 2.7.4

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[alsa-devel] [PATCH v6] ASoC: rockchip: Add machine driver for RK3399 GRU Boards
by Xing Zheng 11 Jul '16

11 Jul '16

Because we need to support the multiple codecs (MAX98357A/RT5514/DA7219) on the RK3399 GRU boards, this patch can help us to support these codecs. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- Changes in v6: - remove enable/disable dapm pins, and to switch codecs via user space - fix "Speaker" dapm route for max98357a - add SND_JACK_LINEOUT flag for da7219 jack detection Changes in v5: - fix the build warning for missing I2C/CLKDEV_LOOKUP dependency Changes in v4: - the compatible name needs to be based on a single reference design, and the name of the board is called "GRU" - use one i2sX card "rockchip,cpu = <&i2s0>" instead of array that is based on GRU - rename DOC to rockchip,rk3399-gru-sound.txt - rename compatible to rockchip,rk3399-gru-sound - rename source code to rk3399_gru_sound.c Changes in v3: - rename DOC to rockchip,rk3399-max98357a-rt5514-da7219.txt - rename compatible to rockchip,rk3399-max98357a-rt5514-da7219 - rename source code to rk3399_max98357a_rt5514_da7219.c Changes in v2: - use the FS 256 to set mclks of the max98357a and rt5514 danamically - add more sample rate for da7219 .../bindings/sound/rockchip,rk3399-gru-sound.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 322 ++++++++++++++++++++ 4 files changed, 350 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt create mode 100644 sound/soc/rockchip/rk3399_gru_sound.c diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt new file mode 100644 index 0000000..f19b6c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs on GRU boards + +Required properties: +- compatible: "rockchip,rk3399-gru-sound" +- rockchip,cpu: The phandle of the Rockchip I2S controller that's + connected to the codecs +- rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs + +Example: + +sound { + compatible = "rockchip,rk3399-gru-sound"; + rockchip,cpu = <&i2s0>; + rockchip,codec = <&max98357a &rt5514 &da7219>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..6d39032 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -41,3 +41,14 @@ config SND_SOC_ROCKCHIP_RT5645 help Say Y or M here if you want to add support for SoC audio on Rockchip boards using the RT5645/RT5650 codec, such as Veyron. + +config SND_SOC_RK3399_GRU_SOUND + tristate "ASoC support multiple codecs for Rockchip RK3399 GRU boards" + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98357A + select SND_SOC_RT5514 + select SND_SOC_DA7219 + help + Say Y or M here if you want to add support multiple codecs for SoC + audio on Rockchip RK3399 GRU boards. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..84e5c7c 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rk3399-gru-sound-objs := rk3399_gru_sound.o obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o +obj-$(CONFIG_SND_SOC_RK3399_GRU_SOUND) += snd-soc-rk3399-gru-sound.o diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c new file mode 100644 index 0000000..43a537c --- /dev/null +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -0,0 +1,322 @@ +/* + * Rockchip machine ASoC driver for boards using MAX98357A/RT5514/DA7219 + * + * Copyright (c) 2016, ROCKCHIP CORPORATION. All rights reserved. + * + * 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. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/delay.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "rockchip_i2s.h" +#include "../codecs/da7219.h" +#include "../codecs/da7219-aad.h" +#include "../codecs/rt5514.h" + +#define DRV_NAME "rk3399-gru-sound" + +#define SOUND_FS 256 + +static struct snd_soc_jack rockchip_sound_jack; + +static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { + /* Input Lines */ + {"MIC", NULL, "Headset Mic"}, + {"DMIC1L", NULL, "Int Mic"}, + {"DMIC1R", NULL, "Int Mic"}, + + /* Output Lines */ + {"Headphones", NULL, "HPL"}, + {"Headphones", NULL, "HPR"}, + {"Speakers", NULL, "Speaker"}, +}; + +static const struct snd_kcontrol_new rockchip_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), +}; + +static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int mclk; + int ret; + + /* max98357a supports these sample rates */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = params_rate(params) * SOUND_FS; + break; + default: + dev_err(rtd->card->dev, "%s() doesn't support this sample rate: %d\n", + __func__, params_rate(params)); + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, mclk, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int mclk; + int ret; + + mclk = params_rate(params) * SOUND_FS; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(rtd->card->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5514_SCLK_S_MCLK, + mclk, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, params_rate(params) * 512, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int mclk, ret; + + /* in bypass mode, the mclk has to be one of the frequencies below */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; + int ret; + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &rockchip_sound_jack, NULL, 0); + + if (ret) { + dev_err(rtd->card->dev, "New Headset Jack failed! (%d)\n", ret); + return ret; + } + + da7219_aad_jack_det(codec, &rockchip_sound_jack); + + return 0; +} + +static struct snd_soc_ops rockchip_sound_max98357a_ops = { + .hw_params = rockchip_sound_max98357a_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_rt5514_ops = { + .hw_params = rockchip_sound_rt5514_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_da7219_ops = { + .hw_params = rockchip_sound_da7219_hw_params, +}; + +enum { + DAILINK_MAX98357A, + DAILINK_RT5514, + DAILINK_DA7219, +}; + +static struct snd_soc_dai_link rockchip_dailinks[] = { + [DAILINK_MAX98357A] = { + .name = "MAX98357A", + .stream_name = "MAX98357A PCM", + .codec_dai_name = "HiFi", + .ops = &rockchip_sound_max98357a_ops, + /* set max98357a as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_RT5514] = { + .name = "RT5514", + .stream_name = "RT5514 PCM", + .codec_dai_name = "rt5514-aif1", + .ops = &rockchip_sound_rt5514_ops, + /* set rt5514 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_DA7219] = { + .name = "DA7219", + .stream_name = "DA7219 PCM", + .codec_dai_name = "da7219-hifi", + .init = rockchip_sound_da7219_init, + .ops = &rockchip_sound_da7219_ops, + /* set da7219 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card rockchip_sound_card = { + .name = "rk3399-gru-sound", + .owner = THIS_MODULE, + .dai_link = rockchip_dailinks, + .num_links = ARRAY_SIZE(rockchip_dailinks), + .dapm_widgets = rockchip_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets), + .dapm_routes = rockchip_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes), + .controls = rockchip_controls, + .num_controls = ARRAY_SIZE(rockchip_controls), +}; + +static int rockchip_sound_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &rockchip_sound_card; + struct device_node *cpu_node; + int i, ret; + + cpu_node = of_parse_phandle(pdev->dev.of_node, "rockchip,cpu", 0); + if (!cpu_node) { + dev_err(&pdev->dev, "Property 'rockchip,cpu' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + rockchip_dailinks[i].platform_of_node = cpu_node; + rockchip_dailinks[i].cpu_of_node = cpu_node; + + rockchip_dailinks[i].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "rockchip,codec", i); + if (!rockchip_dailinks[i].codec_of_node) { + dev_err(&pdev->dev, + "Property[%d] 'rockchip,codec' missing or invalid\n", i); + return -EINVAL; + } + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + + return ret; +} + +static const struct of_device_id rockchip_sound_of_match[] = { + { .compatible = "rockchip,rk3399-gru-sound", }, + {}, +}; + +static struct platform_driver rockchip_sound_driver = { + .probe = rockchip_sound_probe, + .driver = { + .name = DRV_NAME, + .of_match_table = rockchip_sound_of_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, +}; + +module_platform_driver(rockchip_sound_driver); + +MODULE_AUTHOR("Xing Zheng <zhengxing(a)rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_sound_of_match); -- 1.7.9.5

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[alsa-devel] [GIT PULL] skylake audio firmware update
by Vinod Koul 09 Jul '16

09 Jul '16

Hello, Please PULL to receive the updated audio firmware for SKL which contains update for supporting multiple languages. The following changes since commit 3ef7857d511ce6a91c5ce609da76c4702651cfa5: amdgpu: update polaris ucode (2016-06-28 14:31:11 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to d4274de33ed9448e91b970225ce9afdb74d5f13e: linux-firmware: intel: Update Skylake audio firmware (2016-07-01 13:10:07 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Update Skylake audio firmware WHENCE | 4 ++-- intel/dsp_fw_release.bin | 2 +- intel/dsp_fw_release_v951.bin | Bin 0 -> 221184 bytes 3 files changed, 3 insertions(+), 3 deletions(-) create mode 100644 intel/dsp_fw_release_v951.bin Thanks -- ~Vinod

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[alsa-devel] [PATCH 1/2] ASoC: Intel: cht: fix uninit variable warning
by Vinod Koul 08 Jul '16

08 Jul '16

Kbuild bot reports that we might use dai_index uninitialized. sound/soc/intel/boards/cht_bsw_rt5645.c:391:37: warning: 'dai_index' may be used uninitialized in this function [-Wmaybe-uninitialized] Since it is theoretically possible, set it while initializing. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index f26c7b8545ae..56056ed7fcfd 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -357,7 +357,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) char codec_name[16]; struct sst_acpi_mach *mach; const char *i2c_name = NULL; - int dai_index; + int dai_index = 0; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); if (!drv) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix to use the actual size for TLV control
by Vinod Koul 08 Jul '16

08 Jul '16

From: Dharageswari R <dharageswari.r(a)intel.com> DSP expects the actual length of parameters that is set through TLV to be passed in large config set, so pass the actual size received in tlv_control_set() instead of max size. Signed-off-by: Dharageswari R <dharageswari.r(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 13 +++++++++---- sound/soc/intel/skylake/skl-topology.h | 1 + 2 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index b284b3cf5f94..cc0150fc2601 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -411,7 +411,7 @@ static int skl_tplg_set_module_params(struct snd_soc_dapm_widget *w, if (bc->set_params == SKL_PARAM_SET) { ret = skl_set_module_params(ctx, - (u32 *)bc->params, bc->max, + (u32 *)bc->params, bc->size, bc->param_id, mconfig); if (ret < 0) return ret; @@ -446,7 +446,7 @@ static int skl_tplg_set_module_init_data(struct snd_soc_dapm_widget *w) continue; mconfig->formats_config.caps = (u32 *)&bc->params; - mconfig->formats_config.caps_size = bc->max; + mconfig->formats_config.caps_size = bc->size; break; } @@ -1054,7 +1054,7 @@ static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol, if (w->power) skl_get_module_params(skl->skl_sst, (u32 *)bc->params, - bc->max, bc->param_id, mconfig); + bc->size, bc->param_id, mconfig); /* decrement size for TLV header */ size -= 2 * sizeof(u32); @@ -1088,6 +1088,10 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol, struct skl *skl = get_skl_ctx(w->dapm->dev); if (ac->params) { + if (size > ac->max) + return -EINVAL; + + ac->size = size; /* * if the param_is is of type Vendor, firmware expects actual * parameter id and size from the control. @@ -1103,7 +1107,7 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol, if (w->power) return skl_set_module_params(skl->skl_sst, - (u32 *)ac->params, ac->max, + (u32 *)ac->params, ac->size, ac->param_id, mconfig); } @@ -1719,6 +1723,7 @@ static int skl_init_algo_data(struct device *dev, struct soc_bytes_ext *be, ac->max = dfw_ac->max; ac->param_id = dfw_ac->param_id; ac->set_params = dfw_ac->set_params; + ac->size = dfw_ac->max; if (ac->max) { ac->params = (char *) devm_kzalloc(dev, ac->max, GFP_KERNEL); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 170d68b4e12d..22d3ef83817d 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -321,6 +321,7 @@ struct skl_algo_data { u32 param_id; u32 set_params; u32 max; + u32 size; char *params; }; -- 1.9.1

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[alsa-devel] [PATCH 0/3] Add audio support for Microsoft Surface 3 tablet
by Stephen Just 08 Jul '16

08 Jul '16

This series adds audio support for the Surface 3 tablet. This tablet identifes an rt5645 audio codec using the ACPI identifer 10EC5640, which is already associated with the rt5640, thus a quirk is required to have the correct drivers loaded. Additionally, in order for jack-detection and the tablet's internal microphone to work, the Surface 3 must be configured in the audio codec driver similarly to other devices. Resolves: https://bugzilla.kernel.org/show_bug.cgi?id=98001 UCM configuration to be added to alsa-lib is included in patch 3, not included in the diffstat below. Stephen Just (2): ASoC: rt5645: Add support for Surface 3 tablet ASoC: Intel: Atom: add quirk for CHT w/ RT5645 sound/soc/codecs/rt5645.c | 7 +++++++ sound/soc/intel/atom/sst/sst_acpi.c | 28 +++++++++++++++++++++++++++- sound/soc/intel/boards/cht_bsw_rt5645.c | 1 + 3 files changed, 35 insertions(+), 1 deletion(-) -- 2.7.4

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[alsa-devel] [RFC 00/11] ASoC: hdac: Add hdac generic driver
by Subhransu S. Prusty 08 Jul '16

08 Jul '16

HDA devices generically can be modelled with DAPM in ASoC. This series adds a framework in ASoC to model HDA devices. HDA widgets are enumerated and one or multiple DAPM widget(s) are created. Connection list is queried for each widget to identify the connection between two endpoints and modelled using DAPM graph. Set of event handlers are defined for each widget type. Based on DAPM events required verbs are sent to program codec. Finally a function is exported to query for the device endpoint configuration to create machine controls. Hardik T Shah (1): ASoC: Add dai_ops to set the stream tag. Subhransu S. Prusty (10): ALSA: hdac: Add codec helper library ALSA: hda - Add macro to test pin widget's input capability ASoC: hdac: Add a generic hdac driver framework ASoC: hdac: Create DAPM model for HDA widgets ASoC: dapm: Create API to add a single route element ASoC: hdac: Build DAPM graph by querying through widget connection list ASoC: hdac: Register widget event handlers ALSA: hda - macro to get default config device of pin widgets ASoC: dapm: Export snd_soc_dapm_new_control ASoC: hdac: Export API to create machine controls include/sound/hdaudio.h | 1 + include/sound/soc-dai.h | 12 + include/sound/soc-dapm.h | 5 + sound/hda/ext/Makefile | 3 +- sound/hda/ext/hdac_codec.c | 188 +++++ sound/hda/ext/hdac_codec.h | 52 ++ sound/hda/local.h | 13 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_generic.c | 1561 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_generic.h | 31 + sound/soc/soc-core.c | 35 + sound/soc/soc-dapm.c | 22 + 13 files changed, 1929 insertions(+), 1 deletion(-) create mode 100644 sound/hda/ext/hdac_codec.c create mode 100644 sound/hda/ext/hdac_codec.h create mode 100644 sound/soc/codecs/hdac_generic.c create mode 100644 sound/soc/codecs/hdac_generic.h -- 1.9.1

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