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

[alsa-devel] [PATCH] ASoC: Intel: Boards: Add CNL RT274 I2S machine driver
by Guneshwor Singh 17 May '18

17 May '18

Add CNL I2S machine driver using Realtek RT274 codec in I2S mode configured to ssp0. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/intel/boards/Kconfig | 13 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cnl_rt274.c | 250 +++++++++++++++++++++++++++++++++++++ 3 files changed, 265 insertions(+) create mode 100644 sound/soc/intel/boards/cnl_rt274.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 6f754708a48c..29066f0b744f 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -262,4 +262,17 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_CNL_RT274_MACH + tristate "ASoC Audio driver for Cannonlake with RT274 I2S mode" + depends on X86 && ACPI && I2C + select SND_SOC_INTEL_SST + depends on SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT274 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Cannonlake platform + with RT274 I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 69d2dfaeb00c..a7a832c072ca 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -19,6 +19,7 @@ snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o snd-soc-skl_rt286-objs := skl_rt286.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o +snd-soc-cnl-rt274-objs := cnl_rt274.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -40,3 +41,4 @@ obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt566 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_CNL_RT274_MACH) += snd-soc-cnl-rt274.o diff --git a/sound/soc/intel/boards/cnl_rt274.c b/sound/soc/intel/boards/cnl_rt274.c new file mode 100644 index 000000000000..19458fb50f55 --- /dev/null +++ b/sound/soc/intel/boards/cnl_rt274.c @@ -0,0 +1,250 @@ +/* + * cnl_rt274.c - ASOC Machine driver for CNL + * + * Copyright (C) 2016-17 Intel Corp + * Author: Guneshwor Singh <guneshwor.o.singh(a)intel.com> + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program 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 + * General Public License for more details. + * + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#include <linux/module.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../../codecs/rt274.h" + +#define CNL_FREQ_OUT 24000000 +#define CNL_BE_FIXUP_RATE 48000 +#define RT274_CODEC_DAI "rt274-aif1" + +static int cnl_rt274_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai = + snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI); + int ret, ratio = 100; + + if (!codec_dai) + return -EINVAL; + + /* Codec needs clock for Jack detection and button press */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2, + CNL_FREQ_OUT, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret); + return ret; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_bclk_ratio(codec_dai, ratio); + if (ret) { + dev_err(codec_dai->dev, + "set bclk ratio failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK, + CNL_BE_FIXUP_RATE * ratio, + CNL_FREQ_OUT); + if (ret) { + dev_err(codec_dai->dev, + "enable PLL2 failed: %d\n", ret); + return ret; + } + } + + return 0; +} + +static struct snd_soc_jack cnl_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin cnl_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new cnl_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget cnl_rt274_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + cnl_rt274_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static int cnl_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; + + return 0; +} + +static const struct snd_soc_dapm_route cnl_map[] = { + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC", NULL, "Mic Jack"}, + {"DMic", NULL, "SoC DMIC"}, + {"DMIC01 Rx", NULL, "Capture"}, + {"dmic01_hifi", NULL, "DMIC01 Rx"}, + + {"AIF1 Playback", NULL, "ssp0 Tx"}, + {"ssp0 Tx", NULL, "codec1_out"}, + {"ssp0 Tx", NULL, "codec0_out"}, + + {"ssp0 Rx", NULL, "AIF1 Capture"}, + {"codec0_in", NULL, "ssp0 Rx"}, + + {"Headphone Jack", NULL, "Platform Clock"}, + {"Mic Jack", NULL, "Platform Clock"}, +}; + +static int cnl_rt274_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_card *card = runtime->card; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + int ret; + + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_HEADSET, &cnl_headset, + cnl_headset_pins, ARRAY_SIZE(cnl_headset_pins)); + if (ret) + return ret; + + ret = snd_soc_codec_set_jack(codec, &cnl_headset, NULL); + if (ret) + return ret; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0xf, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + return 0; +} + +static int cnl_be_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + rate->min = rate->max = CNL_BE_FIXUP_RATE; + channels->min = channels->max = 2; + snd_mask_none(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT)); + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), + SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static struct snd_soc_dai_link cnl_rt274_dailink[] = { + { + .name = "SSP0-Codec", + .cpu_dai_name = "SSP0 Pin", + .codec_name = "i2c-INT34C2:00", + .codec_dai_name = "rt274-aif1", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_be_fixup, + .no_pcm = 1, + .ignore_pmdown_time = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = cnl_rt274_init, + }, + { + .name = "dmic01", + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_dmic_fixup, + .no_pcm = 1, + .ignore_suspend = 1, + .dpcm_capture = 1, + }, +}; + +static int +cnl_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) +{ + link->platform_name = "0000:00:1f.3"; + link->nonatomic = 1; + + return 0; +} + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cnl = { + .name = "cnl-audio", + .dai_link = cnl_rt274_dailink, + .num_links = ARRAY_SIZE(cnl_rt274_dailink), + .dapm_widgets = cnl_rt274_widgets, + .num_dapm_widgets = ARRAY_SIZE(cnl_rt274_widgets), + .dapm_routes = cnl_map, + .num_dapm_routes = ARRAY_SIZE(cnl_map), + .controls = cnl_controls, + .num_controls = ARRAY_SIZE(cnl_controls), + .add_dai_link = cnl_add_dai_link, + .fully_routed = true, +}; + +static int snd_cnl_rt274_probe(struct platform_device *pdev) +{ + snd_soc_card_cnl.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cnl); +} + +static struct platform_driver snd_cnl_rt274_driver = { + .driver = { + .name = "cnl_rt274", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cnl_rt274_probe, +}; + +module_platform_driver(snd_cnl_rt274_driver); + +MODULE_AUTHOR("Guneshwor Singh <guneshwor.o.singh(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cnl_rt274"); -- 2.15.0

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[alsa-devel] [PATCH 43/45] sound/soc/zte: remove duplicate includes
by Pravin Shedge 17 May '18

17 May '18

These duplicate includes have been found with scripts/checkincludes.pl but they have been removed manually to avoid removing false positives. Signed-off-by: Pravin Shedge <pravin.shedge4linux(a)gmail.com> --- sound/soc/zte/zx-i2s.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c index 9a05659..93428e1 100644 --- a/sound/soc/zte/zx-i2s.c +++ b/sound/soc/zte/zx-i2s.c @@ -20,9 +20,6 @@ #include <sound/core.h> #include <sound/dmaengine_pcm.h> #include <sound/initval.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> #define ZX_I2S_PROCESS_CTRL 0x04 #define ZX_I2S_TIMING_CTRL 0x08 -- 2.7.4

2 2

[alsa-devel] KASAN: null-ptr-deref Write in linear_transfer
by syzbot 13 May '18

13 May '18

Hello, syzkaller hit the following crash on 30a7acd573899fd8b8ac39236eff6468b195ac7d git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/master compiler: gcc (GCC) 7.1.1 20170620 .config is attached Raw console output is attached. C reproducer is attached syzkaller reproducer is attached. See https://goo.gl/kgGztJ for information about syzkaller reproducers IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+a8f5641f452c7e6abf03(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. ================================================================== BUG: KASAN: null-ptr-deref in memcpy include/linux/string.h:344 [inline] BUG: KASAN: null-ptr-deref in do_convert sound/core/oss/linear.c:52 [inline] BUG: KASAN: null-ptr-deref in convert sound/core/oss/linear.c:81 [inline] BUG: KASAN: null-ptr-deref in linear_transfer+0x634/0x900 sound/core/oss/linear.c:110 Write of size 2 at addr (null) by task syzkaller360172/7860 CPU: 0 PID: 7860 Comm: syzkaller360172 Not tainted 4.15.0-rc6+ #155 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x257 lib/dump_stack.c:53 kasan_report_error mm/kasan/report.c:349 [inline] kasan_report+0x13b/0x340 mm/kasan/report.c:409 check_memory_region_inline mm/kasan/kasan.c:260 [inline] check_memory_region+0x137/0x190 mm/kasan/kasan.c:267 memcpy+0x37/0x50 mm/kasan/kasan.c:303 memcpy include/linux/string.h:344 [inline] do_convert sound/core/oss/linear.c:52 [inline] convert sound/core/oss/linear.c:81 [inline] linear_transfer+0x634/0x900 sound/core/oss/linear.c:110 snd_pcm_plug_write_transfer+0x22d/0x420 sound/core/oss/pcm_plugin.c:611 snd_pcm_oss_write2+0x260/0x420 sound/core/oss/pcm_oss.c:1311 snd_pcm_oss_sync1+0x1cc/0x550 sound/core/oss/pcm_oss.c:1530 snd_pcm_oss_sync+0x5b6/0x830 sound/core/oss/pcm_oss.c:1604 snd_pcm_oss_release+0x20b/0x280 sound/core/oss/pcm_oss.c:2431 __fput+0x327/0x7e0 fs/file_table.c:210 ____fput+0x15/0x20 fs/file_table.c:244 task_work_run+0x199/0x270 kernel/task_work.c:113 exit_task_work include/linux/task_work.h:22 [inline] do_exit+0x9bb/0x1ad0 kernel/exit.c:865 do_group_exit+0x149/0x400 kernel/exit.c:968 get_signal+0x73f/0x16c0 kernel/signal.c:2335 do_signal+0x90/0x1eb0 arch/x86/kernel/signal.c:809 exit_to_usermode_loop+0x214/0x310 arch/x86/entry/common.c:158 prepare_exit_to_usermode arch/x86/entry/common.c:195 [inline] syscall_return_slowpath arch/x86/entry/common.c:264 [inline] do_syscall_32_irqs_on arch/x86/entry/common.c:333 [inline] do_fast_syscall_32+0xbfd/0xf9d arch/x86/entry/common.c:389 entry_SYSENTER_compat+0x54/0x63 arch/x86/entry/entry_64_compat.S:129 RIP: 0023:0xf7fcac79 RSP: 002b:00000000f7f8412c EFLAGS: 00000292 ORIG_RAX: 00000000000000f0 RAX: fffffffffffffe00 RBX: 00000000080fd024 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 00000000080fd024 RBP: 0000000000000001 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 ================================================================== Kernel panic - not syncing: panic_on_warn set ... CPU: 0 PID: 7860 Comm: syzkaller360172 Tainted: G B 4.15.0-rc6+ #155 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x257 lib/dump_stack.c:53 panic+0x1e4/0x41c kernel/panic.c:183 kasan_end_report+0x50/0x50 mm/kasan/report.c:176 kasan_report_error mm/kasan/report.c:356 [inline] kasan_report+0x144/0x340 mm/kasan/report.c:409 check_memory_region_inline mm/kasan/kasan.c:260 [inline] check_memory_region+0x137/0x190 mm/kasan/kasan.c:267 memcpy+0x37/0x50 mm/kasan/kasan.c:303 memcpy include/linux/string.h:344 [inline] do_convert sound/core/oss/linear.c:52 [inline] convert sound/core/oss/linear.c:81 [inline] linear_transfer+0x634/0x900 sound/core/oss/linear.c:110 snd_pcm_plug_write_transfer+0x22d/0x420 sound/core/oss/pcm_plugin.c:611 snd_pcm_oss_write2+0x260/0x420 sound/core/oss/pcm_oss.c:1311 snd_pcm_oss_sync1+0x1cc/0x550 sound/core/oss/pcm_oss.c:1530 snd_pcm_oss_sync+0x5b6/0x830 sound/core/oss/pcm_oss.c:1604 snd_pcm_oss_release+0x20b/0x280 sound/core/oss/pcm_oss.c:2431 __fput+0x327/0x7e0 fs/file_table.c:210 ____fput+0x15/0x20 fs/file_table.c:244 task_work_run+0x199/0x270 kernel/task_work.c:113 exit_task_work include/linux/task_work.h:22 [inline] do_exit+0x9bb/0x1ad0 kernel/exit.c:865 do_group_exit+0x149/0x400 kernel/exit.c:968 get_signal+0x73f/0x16c0 kernel/signal.c:2335 do_signal+0x90/0x1eb0 arch/x86/kernel/signal.c:809 exit_to_usermode_loop+0x214/0x310 arch/x86/entry/common.c:158 prepare_exit_to_usermode arch/x86/entry/common.c:195 [inline] syscall_return_slowpath arch/x86/entry/common.c:264 [inline] do_syscall_32_irqs_on arch/x86/entry/common.c:333 [inline] do_fast_syscall_32+0xbfd/0xf9d arch/x86/entry/common.c:389 entry_SYSENTER_compat+0x54/0x63 arch/x86/entry/entry_64_compat.S:129 RIP: 0023:0xf7fcac79 RSP: 002b:00000000f7f8412c EFLAGS: 00000292 ORIG_RAX: 00000000000000f0 RAX: fffffffffffffe00 RBX: 00000000080fd024 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 00000000080fd024 RBP: 0000000000000001 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title If you want to test a patch for this bug, please reply with: #syz test: git://repo/address.git branch and provide the patch inline or as an attachment. To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

3 4

[alsa-devel] Prepare to covert Codec/Platform into Component
by Kuninori Morimoto 09 May '18

09 May '18

Hi Mark, Lars-Peter I will post all prepare patches to convert Codec/Platform into Component. It has 3 steps. 1) add new necessary functions 2) remove/don't use codec hw_write/read 3) remove/don't use codec reg_cache If these are accepted, all prepares were completed. Next, I can post "convert Codec/Platform into Component" patch bomb. These are too many convert patches, to avoid review issue, I will post few of them as sample. 4) sample of don't use rtd->codec 5) sample of convert Platform into Component 6) sample of convert Codec into Component I will post some of "remove" patches too for each steps. I will re-post all "remove" patches if all are accepted. For your review, please consider below. If these are not good, all covert patch doesn't work. 1. use devm_snd_soc_register_component() and remove snd_soc_unregister_component() My patches is using devm_ version function, and removed unregister function. This is not a big deal 2. New flags. Because "Codec" and "CPU/Platform" are using different features, we need to use some new flags. .idle_bias_off/on .ignore_pmdown_time/pmdown_time These are because "CPU/Platform" and "Codec" are using different default value. "Codec" side convert patches are using these. .endianness .non_legacy_dai_naming These are because "Codec" only needs fixup format, and not use "legacy" DAI name. "Codec" side convert patches are using these I added its flags info on each patch. 3. rtd->platform rtd->platform will be removed. Now we can find all rtd connected component by using rtdcom_lookup. rtd->platform will be replaced to snd_soc_rtdcom_lookup(rtd, DRV_NAME); Here, DRV_NAME is platform's component driver name. 4. rtd->codec rtd->codec will be removed. In soc_bind_dai_link(), It is rtd->codec = rtd->codec_dai->codec; And now, DAI has codec and component pointer. So, rtd->codec will be replaced into rtd->codec_dai->component. Best regards --- Kuninori Morimoto

2 104

[alsa-devel] [PATCH] ASoC: Intel: atom: fix ACPI/PCI Kconfig
by Pierre-Louis Bossart 12 Apr '18

12 Apr '18

The split between ACPI and PCI platforms generated issues with randconfig: with SND_SST_ATOM_HIFI2_PLATFORM_PCI=y and SND_SST_ATOM_HIFI2_PLATFORM=m, we get this module link failure: ERROR: "sst_context_init" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_context_cleanup" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_alloc_drv_context" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "intel_sst_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_configure_runtime_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! To keep things simple, let's expose two configs for SND_SST_ATOM_HIFI2_PLATFORM_PCI and SND_SST_ATOM_HIFI2_PLATFORM_ACPI, which select a common SND_SST_ATOM_HIFI2_PLATFORM option. To avoid breaking existing solutions with the semantics change, SND_SST_ATOM_HIFI2_PLATFORM_ACPI uses "default ACPI" so that "make oldnoconfig" and "make olddefconfig" still work as expected. Also remove mentions of Medfield while we are at it since it was removed recently. Reported-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 4772c16ede52 ("ASoC: Intel: Kconfig: Simplify-clarify ACPI/PCI dependencies") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f2c9e8c5970a..d7ac77b0197b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -72,24 +72,28 @@ config SND_SOC_INTEL_BAYTRAIL for Baytrail Chromebooks but this option is now deprecated and is not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead. +config SND_SST_ATOM_HIFI2_PLATFORM + tristate + select SND_SOC_COMPRESS + config SND_SST_ATOM_HIFI2_PLATFORM_PCI - tristate "PCI HiFi2 (Medfield, Merrifield) Platforms" + tristate "PCI HiFi2 (Merrifield) Platforms" depends on X86 && PCI select SND_SST_IPC_PCI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM help - If you have a Intel Medfield or Merrifield/Edison platform, then + If you have a Intel Merrifield/Edison platform, then enable this option by saying Y or m. Distros will typically not - enable this option: Medfield devices are not available to - developers and while Merrifield/Edison can run a mainline kernel with - limited functionality it will require a firmware file which - is not in the standard firmware tree + enable this option: while Merrifield/Edison can run a mainline + kernel with limited functionality it will require a firmware file + which is not in the standard firmware tree -config SND_SST_ATOM_HIFI2_PLATFORM +config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" + default ACPI depends on X86 && ACPI select SND_SST_IPC_ACPI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH select IOSF_MBI help -- 2.14.1

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[alsa-devel] Disable DAPM for click avoidance ?
by Ricard Wanderlof 27 Mar '18

27 Mar '18

I'm having an issue with a codec where the normal DAPM procedure of disabling most of the codec when there is no stream ongoing results in large clicks when a playback stream starts and stops, because the output jumps from 0V when the output is disabled to over a volt when the output is enabled. There appears to be a 'slow stop/start' feature in the codec, however, it takes about one second to ramp up and down the output voltage, and that is too long a delay when starting a stream. In our usecase the power consumtion of the codec is negligable compared to the rest of the electronics, so leaving the codec running at all times is perfectly feasible. Is there any way from a machine driver to disable the DAPM switching, i.e. leave parts of the codec running at all times? In this specific case, there are two internal PM controls which enable the output. One of them is controlled by an output mixer element (SND_SOC_DAPM_MIXER_E), and the other one from set_bias_level(). I've had a look in the core code, but it seems that the whole business of changing the bias level when a stream is started is always performed and that there is no way to say "Set the bias level to ON and leave it there". An alternative would be to augment the codec driver with an 'always on' feature, which effectively always enables PM for certain parts of the chip. But it seems like the wrong place to do this type of thing. Another use for this feature is that we have had a lot of problems over the years with buggy I2S interfaces misbehaving when streams are started and stopped, and there would be an advantage to letting the AIF run the whole time after startup. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH - JACK PCM plugin] jack: Do not Xrun the ALSA buffer
by twischer＠de.adit-jv.com 21 Mar '18

21 Mar '18

From: Timo Wischer <twischer(a)de.adit-jv.com> when the JACK thread is requesting too many audio frames Playback: Without this commit the ALSA audio buffer will be played with endless repeats as long as the user application has not provided new audio data. Therefore this garbage will be played as long as the user application has not called snd_pcm_stop() after an Xrun. With this fix the rest of the JACK buffer will be filled with silence. Capture: Without this commit the audio data in the ALSA buffer would be overwritten. With this commit the new data from the JACK buffer will not be copied. Therefore the existing data in the ALSA buffer will not be overwritten. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- This patch is depending on http://mailman.alsa-project.org/pipermail/alsa-devel/2018-January/130986.ht… diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index c22a5d0..a4f35f4 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -54,6 +54,68 @@ typedef struct { static int snd_pcm_jack_stop(snd_pcm_ioplug_t *io); + +static inline snd_pcm_uframes_t snd_pcm_jack_playback_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + const snd_pcm_jack_t* const jack = io->private_data; + + /* cannot use io->hw_ptr without calling snd_pcm_avail_update() + * because it is not guranteed that snd_pcm_jack_pointer() was already + * called + */ + snd_pcm_sframes_t avail; + avail = hw_ptr + io->buffer_size - appl_ptr; + if (avail < 0) + avail += jack->boundary; + else if ((snd_pcm_uframes_t) avail >= jack->boundary) + avail -= jack->boundary; + return avail; +} + +static inline snd_pcm_uframes_t snd_pcm_jack_capture_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + const snd_pcm_jack_t* const jack = io->private_data; + + /* cannot use io->hw_ptr without calling snd_pcm_avail_update() + * because it is not guranteed that snd_pcm_jack_pointer() was already + * called + */ + snd_pcm_sframes_t avail; + avail = hw_ptr - appl_ptr; + if (avail < 0) + avail += jack->boundary; + return avail; +} + +static inline snd_pcm_uframes_t snd_pcm_jack_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + return (io->stream == SND_PCM_STREAM_PLAYBACK) ? + snd_pcm_jack_playback_avail(io, hw_ptr, appl_ptr) : + snd_pcm_jack_capture_avail(io, hw_ptr, appl_ptr); +} + +static inline snd_pcm_uframes_t snd_pcm_jack_hw_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + /* available data/space which can be transfered by the user application */ + const snd_pcm_uframes_t user_avail = snd_pcm_jack_avail(io, hw_ptr, + appl_ptr); + + if (user_avail > io->buffer_size) { + /* there was an Xrun */ + return 0; + } + /* available data/space which can be transfered by the DMA */ + return io->buffer_size - user_avail; +} + static int pcm_poll_block_check(snd_pcm_ioplug_t *io) { static char buf[32]; @@ -154,7 +216,6 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) { snd_pcm_jack_t *jack = io->private_data; snd_pcm_uframes_t hw_ptr; - const snd_pcm_channel_area_t *areas; snd_pcm_uframes_t xfer = 0; unsigned int channel; @@ -164,40 +225,57 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) jack->areas[channel].first = 0; jack->areas[channel].step = jack->sample_bits; } - - if (io->state != SND_PCM_STATE_RUNNING) { - if (io->stream == SND_PCM_STREAM_PLAYBACK) { - for (channel = 0; channel < io->channels; channel++) - snd_pcm_area_silence(&jack->areas[channel], 0, nframes, io->format); - return 0; - } - } hw_ptr = jack->hw_ptr; - areas = snd_pcm_ioplug_mmap_areas(io); - - while (xfer < nframes) { - snd_pcm_uframes_t frames = nframes - xfer; - snd_pcm_uframes_t offset = hw_ptr % io->buffer_size; - snd_pcm_uframes_t cont = io->buffer_size - offset; - - if (cont < frames) - frames = cont; + if (io->state == SND_PCM_STATE_RUNNING) { + const snd_pcm_channel_area_t *areas = snd_pcm_ioplug_mmap_areas(io); + + while (xfer < nframes) { + snd_pcm_uframes_t frames = nframes - xfer; + snd_pcm_uframes_t offset = hw_ptr % io->buffer_size; + snd_pcm_uframes_t cont = io->buffer_size - offset; + snd_pcm_uframes_t hw_avail = + snd_pcm_jack_hw_avail(io, hw_ptr, + io->appl_ptr); + + /* stop copying if there is no more data available */ + if (hw_avail <= 0) + break; + + /* split the snd_pcm_area_copy() function into two parts + * if the data to copy passes the buffer wrap around + */ + if (cont < frames) + frames = cont; + + if (hw_avail < frames) + frames = hw_avail; + + for (channel = 0; channel < io->channels; channel++) { + if (io->stream == SND_PCM_STREAM_PLAYBACK) + snd_pcm_area_copy(&jack->areas[channel], xfer, &areas[channel], offset, frames, io->format); + else + snd_pcm_area_copy(&areas[channel], offset, &jack->areas[channel], xfer, frames, io->format); + } - for (channel = 0; channel < io->channels; channel++) { - if (io->stream == SND_PCM_STREAM_PLAYBACK) - snd_pcm_area_copy(&jack->areas[channel], xfer, &areas[channel], offset, frames, io->format); - else - snd_pcm_area_copy(&areas[channel], offset, &jack->areas[channel], xfer, frames, io->format); + hw_ptr += frames; + if (hw_ptr >= jack->boundary) + hw_ptr -= jack->boundary; + xfer += frames; } - - hw_ptr += frames; - if (hw_ptr >= jack->boundary) - hw_ptr -= jack->boundary; - xfer += frames; } jack->hw_ptr = hw_ptr; + /* check if requested frames were copied */ + if (xfer < nframes) { + /* always fill the not yet written JACK buffer with silence */ + if (io->stream == SND_PCM_STREAM_PLAYBACK) { + const snd_pcm_uframes_t samples = nframes - xfer; + for (channel = 0; channel < io->channels; channel++) + snd_pcm_area_silence(&jack->areas[channel], xfer, samples, io->format); + } + } + pcm_poll_unblock_check(io); /* unblock socket for polling if needed */ return 0; -- 2.7.4

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[alsa-devel] [PATCH v1 0/6] ALSA: usb: UAC3. Add support for Basic Audio Device (BADD)
by Jorge Sanjuan 19 Mar '18

19 Mar '18

This adds functionality for the Basic Audio Device (BADD) subset that is defined in the USB Audio Class 3 (UAC3). In a BADD compliant driver, all Class Specific descriptors are inferred and so they should all be generated in the *host* as the device wont expose those. This also applies to the new HIGH CAPABILITY descriptor as the GET request is not to be supported in a BADD compliant driver. The new class requires the device to have 3 usb configurations as follows: 1: Legacy Mode: UAC1 or UAC2. 2: BADD device with a prefined topology. (Minimum). 3: UAC3 device for more detailed description or more complex devices that can't be covered by the BADD profile. This patch series also includes some minor fixes to sound/usb. It has been tested with an actual UAC3 compliant device in different host machines and all configs (UAC1/2, BADD and UAC3) work. Also, this has been implemented on top of the the patch which adds UAC3 support to the usb sound card driver: commit ddd452d7b04b86fb5f9285a19ac54deca9264ac1 Author: Ruslan Bilovol <ruslan.bilovol(a)gmail.com> Date: Tue Nov 7 04:01:20 2017 +0200 Jorge Sanjuan (6): usb: audio: Add bi-directional terminal types. usb: audio: Fix variable length field to be variable. ALSA: usb: UAC3. Add initial mixer support. ALSA: usb: Only get AudioControl header for UAC1 class. ALSA: usb: Use Class Specific EP for UAC3 devices. ALSA: usb: ADC3: Add initial BADD spec support include/linux/usb/audio-v3.h | 2 +- include/uapi/linux/usb/audio.h | 21 +- sound/usb/Makefile | 3 +- sound/usb/badd.c | 518 +++++++++++++++++++++++++++++++++++++++++ sound/usb/badd.h | 14 ++ sound/usb/card.c | 25 +- sound/usb/mixer.c | 24 ++ sound/usb/stream.c | 165 ++++++++----- sound/usb/usbaudio.h | 1 + 9 files changed, 702 insertions(+), 71 deletions(-) create mode 100644 sound/usb/badd.c create mode 100644 sound/usb/badd.h -- 2.11.0

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[alsa-devel] [PATCH] ASoC: mt8173-rt5650: fix child-node lookup
by Johan Hovold 12 Mar '18

12 Mar '18

This driver used the wrong OF-helper when looking up the optional capture-codec child node during probe. Instead of searching just children of the sound node, a tree-wide depth-first search starting at the unrelated platform node was done. Not only could this end up matching an unrelated node or no node at all; the platform node could also be prematurely freed since of_find_node_by_name() drops a reference to its first argument. This particular pattern has been observed leading to crashes after probe deferrals in other drivers. Fix this by dropping the broken call to of_find_node_by_name() and keeping only the second, correct lookup using of_get_child_by_name() while taking care not to bail out if the optional node is missing. Note that this also addresses two capture-codec node-reference leaks (one for each of the original helper calls). Compile tested only. Fixes: d349caeb0510 ("ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver") Cc: PC Liao <pc.liao(a)mediatek.com> Signed-off-by: Johan Hovold <johan(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 40ebefd625c1..679fc8bea0a3 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -272,15 +272,10 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) } mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; - if (of_find_node_by_name(platform_node, "codec-capture")) { - np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); - if (!np) { - dev_err(&pdev->dev, - "%s: Can't find codec-capture DT node\n", - __func__); - return -EINVAL; - } + np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); + if (np) { ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); + of_node_put(np); if (ret < 0) { dev_err(&pdev->dev, "%s codec_capture_dai name fail %d\n", -- 2.13.6

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[alsa-devel] [PATCH] sound/tlv320dac33: Add device tree support
by Pavel Machek 05 Mar '18

05 Mar '18

This adds device tree support to tlv320dac33.c. Signed-off-by: Pavel Machek <pavel(a)ucw.cz> diff --git a/Documentation/devicetree/bindings/sound/tlv320dac33.txt b/Documentation/devicetree/bindings/sound/tlv320dac33.txt new file mode 100644 index 0000000..6cbd311 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320dac33.txt @@ -0,0 +1,32 @@ +Texas Instruments - tlv320dac33 Codec module + +The tlv320dac33 serial control bus communicates through I2C protocols. + +Required properties: + +- compatible - "ti,tlv320dac33" +- reg - I2C slave address + +Optional properties: + +- power-gpios - gpio pin to power the device, active high + +- interrupts - The interrupt output from the device. +- interrupt-parent - The parent interrupt controller. + +- ti,keep-bclk - Keep the BCLK running in FIFO modes +- ti,burst-bclkdiv - BCLK divider value in burst mode + +Example: + +tlv320dac33: audio-codec@19 { + compatible = "ti,tlv320dac33"; + reg = <0x19>; + + interrupt-parent = <&gpio2>; + interrupts = <21 1>; /* gpio_53, IRQF_TRIGGER_RISING */ + power-gpio = <&gpio2 28 0>; /* gpio_60 */ + + ti,keep-bclk; + ti,burst-bclkdiv = <3>; +}; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 5b94a15..2a0fe06 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -30,6 +32,7 @@ #include <linux/interrupt.h> #include <linux/gpio.h> #include <linux/regulator/consumer.h> +#include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -105,7 +108,7 @@ struct tlv320dac33_priv { unsigned int nsample; /* burst read amount from host */ int mode1_latency; /* latency caused by the i2c writes in * us */ - u8 burst_bclkdiv; /* BCLK divider value in burst mode */ + u32 burst_bclkdiv; /* BCLK divider value in burst mode */ unsigned int burst_rate; /* Interface speed in Burst modes */ int keep_bclk; /* Keep the BCLK continuously running @@ -1484,16 +1487,11 @@ static struct snd_soc_dai_driver dac33_dai = { static int dac33_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - struct tlv320dac33_platform_data *pdata; + struct tlv320dac33_platform_data *pdata = client->dev.platform_data; struct tlv320dac33_priv *dac33; + struct device_node *np = client->dev.of_node; int ret, i; - if (client->dev.platform_data == NULL) { - dev_err(&client->dev, "Platform data not set\n"); - return -ENODEV; - } - pdata = client->dev.platform_data; - dac33 = devm_kzalloc(&client->dev, sizeof(struct tlv320dac33_priv), GFP_KERNEL); if (dac33 == NULL) @@ -1505,10 +1503,26 @@ static int dac33_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, dac33); - dac33->power_gpio = pdata->power_gpio; - dac33->burst_bclkdiv = pdata->burst_bclkdiv; - dac33->keep_bclk = pdata->keep_bclk; - dac33->mode1_latency = pdata->mode1_latency; + if (pdata) { + dac33->power_gpio = pdata->power_gpio; + dac33->burst_bclkdiv = pdata->burst_bclkdiv; + dac33->keep_bclk = pdata->keep_bclk; + dac33->mode1_latency = pdata->mode1_latency; + } else if (np) { + ret = of_get_named_gpio(np, "power-gpios", 0); + if (ret >= 0) + dac33->power_gpio = ret; + else + dac33->power_gpio = -1; + + if (of_property_read_bool(np, "ti,keep-bclk")) + dac33->keep_bclk = true; + + of_property_read_u32(np, "ti,burst-bclkdiv", &dac33->burst_bclkdiv); + } else { + dev_err(&client->dev, "Platform data not set\n"); + return -ENODEV; + } if (!dac33->mode1_latency) dac33->mode1_latency = 10000; /* 10ms */ dac33->irq = client->irq; @@ -1574,9 +1588,16 @@ static const struct i2c_device_id tlv320dac33_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id); +static const struct of_device_id tlv320dac33_of_match[] = { + { .compatible = "ti,tlv320dac33", }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, tlv320dac33_of_match); + static struct i2c_driver tlv320dac33_i2c_driver = { .driver = { .name = "tlv320dac33-codec", + .of_match_table = of_match_ptr(tlv320dac33_of_match), }, .probe = dac33_i2c_probe, .remove = dac33_i2c_remove, -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

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