September 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/2] ASoC: for_each_component_xx() macro
by Kuninori Morimoto 21 Sep '18

21 Sep '18

Hi Mark These are v2 of for_each_component_xx() macro. v1 -> v2 is renamed comment_all to comment. Kuninori Morimoto (2): ASoC: add for_each_component() macro ASoC: add for_each_component_dais() macro include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 35 ++++++++++++++++++++--------------- 2 files changed, 25 insertions(+), 15 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rsnd: fixup SSI clock during suspend/resume modes
by Kuninori Morimoto 21 Sep '18

21 Sep '18

From: Dmytro Prokopchuk <dmytro.prokopchuk(a)globallogic.com> Prepare <-> Cleanup functions pair has balanced calls. But in case of suspend mode no call to rsnd_soc_dai_shutdown() function, so cleanup isn't called. OTOH during resume mode function rsnd_soc_dai_prepare() is called, but calling rsnd_ssi_prepare() is skipped (rsnd_status_update() returns zero, bacause was not cleanup before). We need to call rsnd_ssi_prepare(), because it enables SSI clocks by calling rsnd_ssi_master_clk_start(). This patch allows to call prepare/cleanup functions always. Signed-off-by: Dmytro Prokopchuk <dmytro.prokopchuk(a)globallogic.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> [kuninori: adjusted to upstream] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 7 +++---- sound/soc/sh/rcar/rsnd.h | 14 +++++++------- 2 files changed, 10 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 0bbc4b0..6d19475 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -134,10 +134,9 @@ static int rsnd_dmaen_prepare(struct rsnd_mod *mod, struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); struct device *dev = rsnd_priv_to_dev(priv); - if (dmaen->chan) { - dev_err(dev, "it already has dma channel\n"); - return -EIO; - } + /* maybe suspended */ + if (dmaen->chan) + return 0; /* * DMAEngine request uses mutex lock. diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index e857311..4464d1d 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -318,9 +318,8 @@ struct rsnd_mod { /* * status * - * 0xH0000CBA + * 0xH0000CB0 * - * A 0: prepare 1: cleanup * B 0: init 1: quit * C 0: start 1: stop * @@ -331,9 +330,8 @@ struct rsnd_mod { * H 0: hw_params * H 0: pointer * H 0: prepare + * H 0: cleanup */ -#define __rsnd_mod_shift_prepare 0 -#define __rsnd_mod_shift_cleanup 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 @@ -345,11 +343,13 @@ struct rsnd_mod { #define __rsnd_mod_shift_fallback 28 /* always called */ #define __rsnd_mod_shift_hw_params 28 /* always called */ #define __rsnd_mod_shift_pointer 28 /* always called */ +#define __rsnd_mod_shift_prepare 28 /* always called */ +#define __rsnd_mod_shift_cleanup 28 /* always called */ #define __rsnd_mod_add_probe 0 #define __rsnd_mod_add_remove 0 -#define __rsnd_mod_add_prepare 1 -#define __rsnd_mod_add_cleanup -1 +#define __rsnd_mod_add_prepare 0 +#define __rsnd_mod_add_cleanup 0 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -363,7 +363,7 @@ struct rsnd_mod { #define __rsnd_mod_call_probe 0 #define __rsnd_mod_call_remove 0 #define __rsnd_mod_call_prepare 0 -#define __rsnd_mod_call_cleanup 1 +#define __rsnd_mod_call_cleanup 0 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 -- 2.7.4

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[alsa-devel] [PATCH] ASoC: tidyup for_each_card_prelinks() dai_link
by Kuninori Morimoto 21 Sep '18

21 Sep '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 7fe072b4df5d0 ("ASoC: add for_each_card_prelinks() macro") added new for_each_card_prelinks() macro, but it had typo. This patch fixup it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 8 ++++---- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 8 ++++---- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 6 +++--- 3 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 875f846..97f9f38 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -311,9 +311,9 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->platform_name) + if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } card->dev = dev; @@ -326,9 +326,9 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } codec_node_bt_mrg = of_parse_phandle(pdev->dev.of_node, diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index c67f629..6bc1d3d 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -107,9 +107,9 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->platform_name) + if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } card->dev = &pdev->dev; @@ -122,9 +122,9 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index ff2e0ca..cc41eb5 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -172,7 +172,7 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) for_each_card_prelinks(card, i, dai_link) { if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } codec_node = of_parse_phandle(pdev->dev.of_node, @@ -183,9 +183,9 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } ret = devm_snd_soc_register_card(&pdev->dev, card); -- 2.7.4

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[alsa-devel] Applied "ASoC: qcom: qdsp6: remove duplicated include from q6adm.c" to the asoc tree
by Mark Brown 21 Sep '18

21 Sep '18

The patch ASoC: qcom: qdsp6: remove duplicated include from q6adm.c has been applied to the asoc tree at https://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 c78d42c7fbd6f2a27a665bcd1ab60c9df617f7c9 Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Fri, 21 Sep 2018 18:24:58 +0800 Subject: [PATCH] ASoC: qcom: qdsp6: remove duplicated include from q6adm.c We include wait.h twice in q6adm.c. it is unnecessary. hence remove it. Further, order the include files as alphabet. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6adm.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 932c3ebfd252..da242515e146 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -2,25 +2,24 @@ // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved. // Copyright (c) 2018, Linaro Limited -#include <linux/slab.h> -#include <linux/wait.h> -#include <linux/kernel.h> #include <linux/device.h> -#include <linux/module.h> -#include <linux/sched.h> #include <linux/jiffies.h> +#include <linux/kernel.h> +#include <linux/kref.h> +#include <linux/module.h> #include <linux/of.h> #include <linux/of_platform.h> -#include <linux/kref.h> -#include <linux/wait.h> -#include <linux/soc/qcom/apr.h> #include <linux/platform_device.h> +#include <linux/sched.h> +#include <linux/slab.h> +#include <linux/soc/qcom/apr.h> +#include <linux/wait.h> #include <sound/asound.h> #include "q6adm.h" #include "q6afe.h" #include "q6core.h" -#include "q6dsp-errno.h" #include "q6dsp-common.h" +#include "q6dsp-errno.h" #define ADM_CMD_DEVICE_OPEN_V5 0x00010326 #define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329 -- 2.19.0

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[alsa-devel] [PATCH] ASoC: fix a use after free case
by Guennadi Liakhovetski 21 Sep '18

21 Sep '18

Unloading ASoC modules as used by the SOF driver leads to an object being used after it's been freed. Fix this be clearing a reference to it and making sure to check for its presence. This fixes issue #144. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/soc-core.c | 6 +++--- sound/soc/soc-topology.c | 5 +++++ 2 files changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9cfe10d8040c..dce9d53ff283 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -942,9 +942,9 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order) { int err; - if (dai && dai->probed && - dai->driver->remove_order == order) { - if (dai->driver->remove) { + if (dai && dai->probed) { + if (dai->driver && dai->driver->remove_order == order && + dai->driver->remove) { err = dai->driver->remove(dai); if (err < 0) dev_err(dai->dev, diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 66e77e020745..6d438ec8c0ec 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -502,6 +502,7 @@ static void remove_dai(struct snd_soc_component *comp, { struct snd_soc_dai_driver *dai_drv = container_of(dobj, struct snd_soc_dai_driver, dobj); + struct snd_soc_dai *dai; if (pass != SOC_TPLG_PASS_PCM_DAI) return; @@ -509,6 +510,10 @@ static void remove_dai(struct snd_soc_component *comp, if (dobj->ops && dobj->ops->dai_unload) dobj->ops->dai_unload(comp, dobj); + list_for_each_entry(dai, &comp->dai_list, list) + if (dai->driver == dai_drv) + dai->driver = NULL; + kfree(dai_drv->name); list_del(&dobj->list); kfree(dai_drv); -- 2.17.1

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[alsa-devel] [PATCH] ASoC:topology:bug fix:check return value avoid oops.
by Guennadi Liakhovetski 21 Sep '18

21 Sep '18

From: Wu Zhigang <zhigang.wu(a)linux.intel.com> check the return value to free the kcontrols instance to avoid oops caused by the pointer dereference. Signed-off-by: Wu Zhigang <zhigang.wu(a)linux.intel.com> [guennadi.liakhovetski(a)linux.intel.com add handling of .autodisable=1 cases] Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/soc-dapm.c | 99 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 90 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e96793050c9..8dc31a7a42fd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2402,12 +2402,11 @@ static void dapm_free_path(struct snd_soc_dapm_path *path) kfree(path); } -void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) +static void snd_soc_dapm_free_widget_data(struct snd_soc_dapm_widget *w) { struct snd_soc_dapm_path *p, *next_p; enum snd_soc_dapm_direction dir; - list_del(&w->list); /* * remove source and sink paths associated to this widget. * While removing the path, remove reference to it from both @@ -2420,6 +2419,12 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) kfree(w->kcontrols); kfree_const(w->name); +} + +void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) +{ + list_del(&w->list); + snd_soc_dapm_free_widget_data(w); kfree(w); } @@ -3038,11 +3043,16 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget *w, *last; unsigned int val; + int ret = 0; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + /* + * widgets with the snd_soc_dapm_kcontrol ID and .num_controls = 0 can + * be appended to the list while scanning it, this is safe. + */ list_for_each_entry(w, &card->widgets, list) { if (w->new) @@ -3053,8 +3063,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) sizeof(struct snd_kcontrol *), GFP_KERNEL); if (!w->kcontrols) { - mutex_unlock(&card->dapm_mutex); - return -ENOMEM; + ret = -ENOMEM; + goto out_free; } } @@ -3062,23 +3072,29 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: - dapm_new_mixer(w); + ret = dapm_new_mixer(w); break; case snd_soc_dapm_mux: case snd_soc_dapm_demux: - dapm_new_mux(w); + ret = dapm_new_mux(w); break; case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: - dapm_new_pga(w); + ret = dapm_new_pga(w); break; case snd_soc_dapm_dai_link: - dapm_new_dai_link(w); + ret = dapm_new_dai_link(w); break; default: break; } + if (ret < 0) { + kfree(w->kcontrols); + w->kcontrols = NULL; + goto out_free; + } + /* Read the initial power state from the device */ if (w->reg >= 0) { soc_dapm_read(w->dapm, w->reg, &val); @@ -3087,6 +3103,11 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) if (val == w->on_val) w->power = 1; } + } + + list_for_each_entry(w, &card->widgets, list) { + if (w->new) + continue; w->new = 1; @@ -3097,6 +3118,66 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) dapm_power_widgets(card, SND_SOC_DAPM_STREAM_NOP); mutex_unlock(&card->dapm_mutex); return 0; + +out_free: + last = w; + + /* + * If any new widgets have been created above for .autodisable = 1 + * controls, they are also on this list, but at its very end. We're + * processing an error case, so the loop above was interrupted before + * reaching dynamically added widgets, since the latter cannot fail - + * their .num_controls = 0, so allocation cannot fail, and their type is + * snd_soc_dapm_kcontrol, those cannot fail either. Therefore "last" + * points to a widget before the first dynamic one. + */ + list_for_each_entry(w, &card->widgets, list) { + unsigned int i; + + if (w->new) + continue; + + if (w == last) + break; + + switch (w->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + for (i = 0; i < w->num_kcontrols; i++) { + struct snd_kcontrol *kcontrol = w->kcontrols[i]; + struct dapm_kcontrol_data *data = + kcontrol->private_data; + struct soc_mixer_control *mc = + (struct soc_mixer_control *) + kcontrol->private_value; + + if (mc->autodisable) + snd_soc_dapm_free_widget(data->widget); + } + break; + case snd_soc_dapm_demux: + case snd_soc_dapm_mux: + for (i = 0; i < w->num_kcontrols; i++) { + struct snd_kcontrol *kcontrol = w->kcontrols[i]; + struct dapm_kcontrol_data *data = + kcontrol->private_data; + struct soc_enum *e = (struct soc_enum *) + kcontrol->private_value; + + if (e->autodisable) + snd_soc_dapm_free_widget(data->widget); + } + break; + default: + break; + } + + snd_soc_dapm_free_widget_data(w); + } + + mutex_unlock(&card->dapm_mutex); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); -- 2.17.1

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[alsa-devel] [PATCH v2] ALSA: xen-front: Refine indentations and constify snd_pcm_ops
by Nick Simonov 21 Sep '18

21 Sep '18

snd_pcm_ops are not supposed to change. So mark the non-const structs as const. Also, refine indentation to increase readability. Signed-off-by: Nick Simonov <nicksimonovv(a)gmail.com> --- Changes v2: - Fix typo in commit description, ncrease to increase. sound/xen/xen_snd_front_alsa.c | 46 +++++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index 129180e..2cbd967 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -637,31 +637,31 @@ static int alsa_pb_fill_silence(struct snd_pcm_substream *substream, * to know when the buffer can be transferred to the backend. */ -static struct snd_pcm_ops snd_drv_alsa_playback_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_pb_copy_user, - .copy_kernel = alsa_pb_copy_kernel, - .fill_silence = alsa_pb_fill_silence, +static const struct snd_pcm_ops snd_drv_alsa_playback_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_pb_copy_user, + .copy_kernel = alsa_pb_copy_kernel, + .fill_silence = alsa_pb_fill_silence, }; -static struct snd_pcm_ops snd_drv_alsa_capture_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_cap_copy_user, - .copy_kernel = alsa_cap_copy_kernel, +static const struct snd_pcm_ops snd_drv_alsa_capture_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_cap_copy_user, + .copy_kernel = alsa_cap_copy_kernel, }; static int new_pcm_instance(struct xen_snd_front_card_info *card_info, -- 2.7.4

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[alsa-devel] [PATCH 00/11] add add for_each_xx() macro
by Kuninori Morimoto 20 Sep '18

20 Sep '18

Hi Mark Current ALSA SoC is using many list_for_each_entry() and/or for() loop for each lists. But, these need to use same pattern, and it is not readable. Now, "codec list loop for rtd", and "codec list loop for dai_link" had been replaced to for_each_xxx() macro. These patches replace "xxx list loop for yyy" by using for_each_yyy_xxx() macro. # Maybe some of them are overkill replacement, and/or # using not good naming. I'm happy to get feedback 1) - 2) : fixup previous for_each_xxx() macro patch 3) - 11) : new for_each_xxx() macro Kuninori Morimoto (11): 1) ASoC: convert for_each_rtd_codec_dai() for missing part 2) ASoC: rename for_each_rtd_codec_dai_reverse to rollback 3) ASoC: add for_each_card_prelinks() macro 4) ASoC: add for_each_card_links() macro 5) ASoC: add for_each_card_rtds() macro 6) ASoC: add for_each_card_components() macro 7) ASoC: add for_each_component_all() macro 8) ASoC: add for_each_component_dais() macro 9) ASoC: add for_each_comp_order() macro 10) ASoC: add for_each_dpcm_fe() macro 11) ASoC: add for_each_dpcm_be() macro include/sound/soc-dpcm.h | 10 ++ include/sound/soc.h | 29 ++++- sound/soc/codecs/hdac_hdmi.c | 2 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 5 +- sound/soc/generic/simple-card-utils.c | 6 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 4 +- sound/soc/intel/boards/broadwell.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 4 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 5 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 13 +-- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 13 +-- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 13 +-- sound/soc/mediatek/mt8173/mt8173-max98090.c | 13 +-- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 7 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 7 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 7 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/qcom/apq8096.c | 7 +- sound/soc/qcom/sdm845.c | 7 +- sound/soc/samsung/tm2_wm5110.c | 13 +-- sound/soc/sh/rcar/ctu.c | 2 +- sound/soc/sh/rcar/src.c | 2 +- sound/soc/soc-compress.c | 4 +- sound/soc/soc-core.c | 130 +++++++++++------------ sound/soc/soc-dapm.c | 2 +- sound/soc/soc-pcm.c | 101 +++++++++--------- 29 files changed, 230 insertions(+), 193 deletions(-) -- 2.7.4

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[alsa-devel] Applied "ASoC: rt5663: Remove the boost volume in the beginning of playback" to the asoc tree
by Mark Brown 20 Sep '18

20 Sep '18

The patch ASoC: rt5663: Remove the boost volume in the beginning of playback has been applied to the asoc tree at https://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 fc795bf7224efda8c35e505966c2757856064247 Mon Sep 17 00:00:00 2001 From: Oder Chiou <oder_chiou(a)realtek.com> Date: Wed, 19 Sep 2018 09:56:47 +0800 Subject: [PATCH] ASoC: rt5663: Remove the boost volume in the beginning of playback The patch removes the boost volume in the beginning of playback while the DAC volume set to lower. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5663.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 9bd24ad42240..2444fad7c2df 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -72,6 +72,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8020 }, { 0x0086, 0x0028 }, + { 0x0100, 0xa020 }, { 0x0117, 0x0f28 }, { 0x02fb, 0x8089 }, }; @@ -580,7 +581,7 @@ static const struct reg_default rt5663_reg[] = { { 0x00fd, 0x0001 }, { 0x00fe, 0x10ec }, { 0x00ff, 0x6406 }, - { 0x0100, 0xa0a0 }, + { 0x0100, 0xa020 }, { 0x0108, 0x4444 }, { 0x0109, 0x4444 }, { 0x010a, 0xaaaa }, @@ -2337,6 +2338,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, 0x8000); snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000, 0x3000); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080); } break; @@ -2351,6 +2354,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN); snd_soc_component_update_bits(component, RT5663_DACREF_LDO, 0x3e0e, 0); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0); } break; -- 2.19.0

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[alsa-devel] [PATCH v2 00/12] Sound Open Firmware Core
by Liam Girdwood 20 Sep '18

20 Sep '18

Sound Open Firmware (SOF) is a host and DSP architecture agnostic audio DSP firmware. SOF is not tied to any specific host architecture or any specific physical IO communication type (it will work with on SoC DSPs, or DSP connected via SPI/I2C). SOF is also not coupled to any particular DSP architecture and has abstraction similar to Linux to allow porting to other DSP architectures. https://www.sofproject.org/ This patch series introduces the SOF core, later series will add host platform support for SOF. The SOF core manages all the core DSP services and ALSA/ASoC IO. The core is responsible for loading firmware, parsing topology, exposing PCMs and kcontrols, providing debug and trace mechanisms and performing IPC between host and DSP. The SOF core also has logic to allow reuse of existing machine drivers for other platforms/machines without any code modification. i.e. DAI links can be modified at runtime to bind with SOF and SOF topologies instead of existing hard coded DAI links and topology. Changes since V1 :- o Added PM source. o Updated header comments to C++ style o Used generic private data in snd_soc_pcm_runtime o remove sof_get_pages() o Added comments for compressed SG pages. o Check return values for runtime_pm calls. Liam Girdwood (12): ASoC: SOF: Add Sound Open Firmware driver core ASoC: SOF: Add Sound Open Firmware KControl support ASoC: SOF: Add driver debug support. ASoC: SOF: Add support for IPC IO between DSP and Host ASoC: SOF: Add PCM operations support ASoC: SOF: Add support for loading topologies ASoC: SOF: Add DSP firmware trace event support ASoC: SOF: Add DSP HW abstraction operations ASoC: SOF: Add firmware loader support ASoC: SOF: Add PM support ASoC: SOF: Add userspace ABI support ASoC: SOF: Add Build support for SOF core include/sound/soc.h | 3 + include/sound/sof.h | 82 ++ include/uapi/sound/sof-abi.h | 29 + include/uapi/sound/sof-eq.h | 102 ++ include/uapi/sound/sof-fw.h | 67 + include/uapi/sound/sof-ipc.h | 938 ++++++++++++ include/uapi/sound/sof-tone.h | 27 + include/uapi/sound/sof-topology.h | 94 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sof/Kconfig | 36 + sound/soc/sof/Makefile | 6 + sound/soc/sof/control.c | 270 ++++ sound/soc/sof/core.c | 381 +++++ sound/soc/sof/debug.c | 157 ++ sound/soc/sof/ipc.c | 788 ++++++++++ sound/soc/sof/loader.c | 287 ++++ sound/soc/sof/ops.c | 210 +++ sound/soc/sof/ops.h | 298 ++++ sound/soc/sof/pcm.c | 733 ++++++++++ sound/soc/sof/pm.c | 345 +++++ sound/soc/sof/sof-priv.h | 519 +++++++ sound/soc/sof/topology.c | 2222 +++++++++++++++++++++++++++++ sound/soc/sof/trace.c | 293 ++++ 24 files changed, 7889 insertions(+) create mode 100644 include/sound/sof.h create mode 100644 include/uapi/sound/sof-abi.h create mode 100644 include/uapi/sound/sof-eq.h create mode 100644 include/uapi/sound/sof-fw.h create mode 100644 include/uapi/sound/sof-ipc.h create mode 100644 include/uapi/sound/sof-tone.h create mode 100644 include/uapi/sound/sof-topology.h create mode 100644 sound/soc/sof/Kconfig create mode 100644 sound/soc/sof/Makefile create mode 100644 sound/soc/sof/control.c create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/debug.c create mode 100644 sound/soc/sof/ipc.c create mode 100644 sound/soc/sof/loader.c create mode 100644 sound/soc/sof/ops.c create mode 100644 sound/soc/sof/ops.h create mode 100644 sound/soc/sof/pcm.c create mode 100644 sound/soc/sof/pm.c create mode 100644 sound/soc/sof/sof-priv.h create mode 100644 sound/soc/sof/topology.c create mode 100644 sound/soc/sof/trace.c -- 2.17.1

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