December 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RESEND PATCH] ASoC: intel: Add Broadwell rt5650 machine driver
by Curtis Malainey 04 Dec '19

04 Dec '19

From: Ben Zhang <benzh(a)chromium.org> Add machine driver for Broadwell + rt5650. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Jon Flatley <jflat(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5650.c | 310 ++++++++++++++++++ .../common/soc-acpi-intel-hsw-bdw-match.c | 5 + 4 files changed, 328 insertions(+) create mode 100644 sound/soc/intel/boards/bdw-rt5650.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 6c9fd9ad566ed..5f31fadd2e1a1 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -31,6 +31,17 @@ endif ## SND_SOC_INTEL_HASWELL if SND_SOC_INTEL_HASWELL || SND_SOC_SOF_BROADWELL +config SND_SOC_INTEL_BDW_RT5650_MACH + tristate "Broadwell with RT5650 codec" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC + select SND_COMPRESS_OFFLOAD + select SND_SOC_RT5645 + help + This adds the ASoC machine driver for Intel Broadwell platforms with + the RT5650 codec. + Say Y if you have such a device + If unsure select "N". + config SND_SOC_INTEL_BDW_RT5677_MACH tristate "Broadwell with RT5677 codec" depends on I2C diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index ba1aa89db09d5..4efb3689503f8 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -2,6 +2,7 @@ snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-bdw-rt5650-mach-objs := bdw-rt5650.o snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o hda_dsp_common.o @@ -37,6 +38,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da721 obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c new file mode 100644 index 0000000000000..df8be71dfd72e --- /dev/null +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -0,0 +1,310 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * ASoC machine driver for Intel Broadwell platforms with RT5650 codec + * + * Copyright 2019, The Chromium OS Authors. All rights reserved. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/gpio/consumer.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> + +#include "../common/sst-dsp.h" +#include "../haswell/sst-haswell-ipc.h" + +#include "../../codecs/rt5645.h" + +struct bdw_rt5650_priv { + struct gpio_desc *gpio_hp_en; + struct snd_soc_component *component; +}; + +static const struct snd_soc_dapm_widget bdw_rt5650_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("DMIC Pair1", NULL), + SND_SOC_DAPM_MIC("DMIC Pair2", NULL), +}; + +static const struct snd_soc_dapm_route bdw_rt5650_map[] = { + /* Speakers */ + {"Speaker", NULL, "SPOL"}, + {"Speaker", NULL, "SPOR"}, + + /* Headset jack connectors */ + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + + /* Digital MICs + * DMIC Pair1 are the two DMICs connected on the DMICN1 connector. + * DMIC Pair2 are the two DMICs connected on the DMICN2 connector. + * Facing the camera, DMIC Pair1 are on the left side, DMIC Pair2 + * are on the right side. + */ + {"DMIC L1", NULL, "DMIC Pair1"}, + {"DMIC R1", NULL, "DMIC Pair1"}, + {"DMIC L2", NULL, "DMIC Pair2"}, + {"DMIC R2", NULL, "DMIC Pair2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static const struct snd_kcontrol_new bdw_rt5650_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("DMIC Pair1"), + SOC_DAPM_PIN_SWITCH("DMIC Pair2"), +}; + + +static struct snd_soc_jack headphone_jack; +static struct snd_soc_jack mic_jack; + +static struct snd_soc_jack_pin headphone_jack_pin = { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, +}; + +static struct snd_soc_jack_pin mic_jack_pin = { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, +}; + +static int broadwell_ssp0_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); + + /* The ADSP will covert the FE rate to 48k, max 4-channels */ + rate->min = rate->max = 48000; + channels->min = 2; + channels->max = 4; + + /* set SSP0 to 24 bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int bdw_rt5650_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 *codec_dai = rtd->codec_dai; + int ret; + + /* Workaround: set codec PLL to 19.2MHz that PLL source is + * from MCLK(24MHz) to conform 2.4MHz DMIC clock. + */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, + 24000000, 19200000); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + /* The actual MCLK freq is 24MHz. The codec is told that MCLK is + * 24.576MHz to satisfy the requirement of rl6231_get_clk_info. + * ASRC is enabled on AD and DA filters to ensure good audio quality. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, 24576000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops bdw_rt5650_ops = { + .hw_params = bdw_rt5650_hw_params, +}; + +static int bdw_rt5650_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct sst_pdata *pdata = dev_get_platdata(component->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings + * clock_divider = 4 means BCLK = MCLK/5 = 24MHz/5 = 4.8MHz + */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_TDM_CLOCK_MASTER, 4); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + return 0; +} + +static int bdw_rt5650_init(struct snd_soc_pcm_runtime *rtd) +{ + struct bdw_rt5650_priv *bdw_rt5650 = + snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Enable codec ASRC function for Stereo DAC/Stereo1 ADC/DMIC/I2S1. + * The ASRC clock source is clk_i2s1_asrc. + */ + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER | + RT5645_AD_STEREO_FILTER | + RT5645_AD_MONO_L_FILTER | + RT5645_AD_MONO_R_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + + /* 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(rtd->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + /* Create and initialize headphone jack */ + if (snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &headphone_jack, + &headphone_jack_pin, 1)) { + dev_err(component->dev, "Can't create headphone jack\n"); + } + + /* Create and initialize mic jack */ + if (snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &mic_jack, &mic_jack_pin, 1)) { + dev_err(component->dev, "Can't create mic jack\n"); + } + + rt5645_set_jack_detect(component, &headphone_jack, &mic_jack, NULL); + + bdw_rt5650->component = component; + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +SND_SOC_DAILINK_DEF(dummy, + DAILINK_COMP_ARRAY(COMP_DUMMY())); + +SND_SOC_DAILINK_DEF(fe, + DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); + +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); + +SND_SOC_DAILINK_DEF(be, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1"))); + +static struct snd_soc_dai_link bdw_rt5650_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback", + .dynamic = 1, + .init = bdw_rt5650_rtd_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(fe, dummy, platform), + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .id = 0, + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &bdw_rt5650_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = bdw_rt5650_init, + SND_SOC_DAILINK_REG(dummy, be, dummy), + }, +}; + +/* ASoC machine driver for Broadwell DSP + RT5650 */ +static struct snd_soc_card bdw_rt5650_card = { + .name = "bdw-rt5650", + .owner = THIS_MODULE, + .dai_link = bdw_rt5650_dais, + .num_links = ARRAY_SIZE(bdw_rt5650_dais), + .dapm_widgets = bdw_rt5650_widgets, + .num_dapm_widgets = ARRAY_SIZE(bdw_rt5650_widgets), + .dapm_routes = bdw_rt5650_map, + .num_dapm_routes = ARRAY_SIZE(bdw_rt5650_map), + .controls = bdw_rt5650_controls, + .num_controls = ARRAY_SIZE(bdw_rt5650_controls), + .fully_routed = true, +}; + +static int bdw_rt5650_probe(struct platform_device *pdev) +{ + struct bdw_rt5650_priv *bdw_rt5650; + + bdw_rt5650_card.dev = &pdev->dev; + + /* Allocate driver private struct */ + bdw_rt5650 = devm_kzalloc(&pdev->dev, sizeof(struct bdw_rt5650_priv), + GFP_KERNEL); + if (!bdw_rt5650) + return -ENOMEM; + snd_soc_card_set_drvdata(&bdw_rt5650_card, bdw_rt5650); + + return snd_soc_register_card(&bdw_rt5650_card); +} + +static int bdw_rt5650_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&bdw_rt5650_card); + return 0; +} + +static struct platform_driver bdw_rt5650_audio = { + .probe = bdw_rt5650_probe, + .remove = bdw_rt5650_remove, + .driver = { + .name = "bdw-rt5650", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(bdw_rt5650_audio) + +/* Module information */ +MODULE_AUTHOR("Ben Zhang <benzh(a)chromium.org>"); +MODULE_DESCRIPTION("Intel Broadwell RT5650 machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bdw-rt5650"); diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index 34eb0baaa9517..cae1e559ec79b 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -29,6 +29,11 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { .sof_fw_filename = "sof-bdw.ri", .sof_tplg_filename = "sof-bdw-rt286.tplg", }, + { + .id = "10EC5650", + .drv_name = "bdw-rt5650", + .fw_filename = "intel/IntcSST2.bin", + }, { .id = "RT5677CE", .drv_name = "bdw-rt5677", -- 2.24.0.393.g34dc348eaf-goog

3 4

[alsa-devel] [PATCH resend] ASoC: fsl_audmix: add missed pm_runtime_disable
by Chuhong Yuan 04 Dec '19

04 Dec '19

The driver forgets to call pm_runtime_disable in probe failure and remove. Add the missed calls to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/fsl/fsl_audmix.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index a1db1bce330f..5faecbeb5497 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -505,15 +505,20 @@ static int fsl_audmix_probe(struct platform_device *pdev) ARRAY_SIZE(fsl_audmix_dai)); if (ret) { dev_err(dev, "failed to register ASoC DAI\n"); - return ret; + goto err_disable_pm; } priv->pdev = platform_device_register_data(dev, mdrv, 0, NULL, 0); if (IS_ERR(priv->pdev)) { ret = PTR_ERR(priv->pdev); dev_err(dev, "failed to register platform %s: %d\n", mdrv, ret); + goto err_disable_pm; } + return 0; + +err_disable_pm: + pm_runtime_disable(dev); return ret; } @@ -521,6 +526,8 @@ static int fsl_audmix_remove(struct platform_device *pdev) { struct fsl_audmix *priv = dev_get_drvdata(&pdev->dev); + pm_runtime_disable(&pdev->dev); + if (priv->pdev) platform_device_unregister(priv->pdev); -- 2.24.0

2 1

[alsa-devel] KASAN: slab-out-of-bounds Read in linear_transfer (3)
by syzbot 04 Dec '19

04 Dec '19

Hello, syzbot found the following crash on: HEAD commit: 76bb8b05 Merge tag 'kbuild-v5.5' of git://git.kernel.org/p.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=1416a90ee00000 kernel config: https://syzkaller.appspot.com/x/.config?x=dd226651cb0f364b dashboard link: https://syzkaller.appspot.com/bug?extid=f153bde47a62e0b05f83 compiler: gcc (GCC) 9.0.0 20181231 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=10fe9e41e00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=1522042ae00000 The bug was bisected to: commit 65766ee0bf7fe8b3be80e2e1c3ef54ad59b29476 Author: Takashi Iwai <tiwai(a)suse.de> Date: Fri Nov 9 10:59:45 2018 +0000 ALSA: oss: Use kvzalloc() for local buffer allocations bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=1727742ae00000 final crash: https://syzkaller.appspot.com/x/report.txt?x=14a7742ae00000 console output: https://syzkaller.appspot.com/x/log.txt?x=10a7742ae00000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+f153bde47a62e0b05f83(a)syzkaller.appspotmail.com Fixes: 65766ee0bf7f ("ALSA: oss: Use kvzalloc() for local buffer allocations") ================================================================== BUG: KASAN: slab-out-of-bounds in memcpy include/linux/string.h:380 [inline] BUG: KASAN: slab-out-of-bounds in do_convert sound/core/oss/linear.c:48 [inline] BUG: KASAN: slab-out-of-bounds in convert sound/core/oss/linear.c:81 [inline] BUG: KASAN: slab-out-of-bounds in linear_transfer sound/core/oss/linear.c:110 [inline] BUG: KASAN: slab-out-of-bounds in linear_transfer+0x6de/0x970 sound/core/oss/linear.c:88 Read of size 1 at addr ffff8880a3eb5540 by task syz-executor567/8984 CPU: 0 PID: 8984 Comm: syz-executor567 Not tainted 5.4.0-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x197/0x210 lib/dump_stack.c:118 print_address_description.constprop.0.cold+0xd4/0x30b mm/kasan/report.c:374 __kasan_report.cold+0x1b/0x41 mm/kasan/report.c:506 kasan_report+0x12/0x20 mm/kasan/common.c:638 check_memory_region_inline mm/kasan/generic.c:185 [inline] check_memory_region+0x134/0x1a0 mm/kasan/generic.c:192 memcpy+0x24/0x50 mm/kasan/common.c:124 memcpy include/linux/string.h:380 [inline] do_convert sound/core/oss/linear.c:48 [inline] convert sound/core/oss/linear.c:81 [inline] linear_transfer sound/core/oss/linear.c:110 [inline] linear_transfer+0x6de/0x970 sound/core/oss/linear.c:88 snd_pcm_plug_read_transfer+0x197/0x2e0 sound/core/oss/pcm_plugin.c:651 snd_pcm_oss_read2+0x1f0/0x3f0 sound/core/oss/pcm_oss.c:1460 snd_pcm_oss_read1 sound/core/oss/pcm_oss.c:1517 [inline] snd_pcm_oss_read+0x548/0x6a0 sound/core/oss/pcm_oss.c:2741 __vfs_read+0x8a/0x110 fs/read_write.c:425 vfs_read+0x1f0/0x440 fs/read_write.c:461 ksys_read+0x14f/0x290 fs/read_write.c:587 __do_sys_read fs/read_write.c:597 [inline] __se_sys_read fs/read_write.c:595 [inline] __x64_sys_read+0x73/0xb0 fs/read_write.c:595 do_syscall_64+0xfa/0x790 arch/x86/entry/common.c:294 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x445a99 Code: e8 dc bd 02 00 48 83 c4 18 c3 0f 1f 80 00 00 00 00 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 cb 11 fc ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007f2955880ce8 EFLAGS: 00000246 ORIG_RAX: 0000000000000000 RAX: ffffffffffffffda RBX: 00000000006dbc28 RCX: 0000000000445a99 RDX: 0000000000001000 RSI: 0000000020000380 RDI: 0000000000000003 RBP: 00000000006dbc20 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000006dbc2c R13: 00007ffd9643eb4f R14: 00007f29558819c0 R15: 20c49ba5e353f7cf Allocated by task 8984: save_stack+0x23/0x90 mm/kasan/common.c:71 set_track mm/kasan/common.c:79 [inline] __kasan_kmalloc mm/kasan/common.c:512 [inline] __kasan_kmalloc.constprop.0+0xcf/0xe0 mm/kasan/common.c:485 kasan_kmalloc+0x9/0x10 mm/kasan/common.c:526 __do_kmalloc_node mm/slab.c:3616 [inline] __kmalloc_node+0x4e/0x70 mm/slab.c:3623 kmalloc_node include/linux/slab.h:579 [inline] kvmalloc_node+0x68/0x100 mm/util.c:574 kvmalloc include/linux/mm.h:655 [inline] kvzalloc include/linux/mm.h:663 [inline] snd_pcm_plugin_alloc+0x585/0x770 sound/core/oss/pcm_plugin.c:70 snd_pcm_plug_alloc+0x146/0x330 sound/core/oss/pcm_plugin.c:129 snd_pcm_oss_change_params_locked+0x210f/0x3750 sound/core/oss/pcm_oss.c:1024 snd_pcm_oss_change_params+0x7b/0xd0 sound/core/oss/pcm_oss.c:1087 snd_pcm_oss_get_active_substream+0x136/0x190 sound/core/oss/pcm_oss.c:1104 snd_pcm_oss_get_rate sound/core/oss/pcm_oss.c:1754 [inline] snd_pcm_oss_set_rate sound/core/oss/pcm_oss.c:1746 [inline] snd_pcm_oss_ioctl+0x1794/0x33a0 sound/core/oss/pcm_oss.c:2593 vfs_ioctl fs/ioctl.c:47 [inline] file_ioctl fs/ioctl.c:545 [inline] do_vfs_ioctl+0x977/0x14e0 fs/ioctl.c:732 ksys_ioctl+0xab/0xd0 fs/ioctl.c:749 __do_sys_ioctl fs/ioctl.c:756 [inline] __se_sys_ioctl fs/ioctl.c:754 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:754 do_syscall_64+0xfa/0x790 arch/x86/entry/common.c:294 entry_SYSCALL_64_after_hwframe+0x49/0xbe Freed by task 0: (stack is not available) The buggy address belongs to the object at ffff8880a3eb5500 which belongs to the cache kmalloc-64 of size 64 The buggy address is located 0 bytes to the right of 64-byte region [ffff8880a3eb5500, ffff8880a3eb5540) The buggy address belongs to the page: page:ffffea00028fad40 refcount:1 mapcount:0 mapping:ffff8880aa400380 index:0x0 raw: 00fffe0000000200 ffffea00027da7c8 ffff8880aa401348 ffff8880aa400380 raw: 0000000000000000 ffff8880a3eb5000 0000000100000020 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8880a3eb5400: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff8880a3eb5480: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc > ffff8880a3eb5500: 00 00 00 00 00 00 00 00 fc fc fc fc fc fc fc fc ^ ffff8880a3eb5580: 00 00 00 00 00 fc fc fc fc fc fc fc fc fc fc fc ffff8880a3eb5600: 00 00 00 00 00 fc fc fc fc fc fc fc fc fc fc fc ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. For information about bisection process see: https://goo.gl/tpsmEJ#bisection syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

2 1

[alsa-devel] [PATCH] ALSA: pcm: oss: Avoid potential buffer overflows
by Takashi Iwai 04 Dec '19

04 Dec '19

syzkaller reported an invalid access in PCM OSS read, and this seems to be an overflow of the internal buffer allocated for a plugin. Since the rate plugin adjusts its transfer size dynamically, the calculation for the chained plugin might be bigger than the given buffer size in some extreme cases, which lead to such an buffer overflow as caught by KASAN. Fix it by limiting the max transfer size properly by checking against the destination size in each plugin transfer callback. Reported-by: syzbot+f153bde47a62e0b05f83(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/linear.c | 2 ++ sound/core/oss/mulaw.c | 2 ++ sound/core/oss/route.c | 2 ++ 3 files changed, 6 insertions(+) diff --git a/sound/core/oss/linear.c b/sound/core/oss/linear.c index 2045697f449d..797d838a2f9e 100644 --- a/sound/core/oss/linear.c +++ b/sound/core/oss/linear.c @@ -107,6 +107,8 @@ static snd_pcm_sframes_t linear_transfer(struct snd_pcm_plugin *plugin, } } #endif + if (frames > dst_channels[0].frames) + frames = dst_channels[0].frames; convert(plugin, src_channels, dst_channels, frames); return frames; } diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c index 7915564bd394..3788906421a7 100644 --- a/sound/core/oss/mulaw.c +++ b/sound/core/oss/mulaw.c @@ -269,6 +269,8 @@ static snd_pcm_sframes_t mulaw_transfer(struct snd_pcm_plugin *plugin, } } #endif + if (frames > dst_channels[0].frames) + frames = dst_channels[0].frames; data = (struct mulaw_priv *)plugin->extra_data; data->func(plugin, src_channels, dst_channels, frames); return frames; diff --git a/sound/core/oss/route.c b/sound/core/oss/route.c index c8171f5783c8..72dea04197ef 100644 --- a/sound/core/oss/route.c +++ b/sound/core/oss/route.c @@ -57,6 +57,8 @@ static snd_pcm_sframes_t route_transfer(struct snd_pcm_plugin *plugin, return -ENXIO; if (frames == 0) return 0; + if (frames > dst_channels[0].frames) + frames = dst_channels[0].frames; nsrcs = plugin->src_format.channels; ndsts = plugin->dst_format.channels; -- 2.16.4

1 0

[alsa-devel] [PATCH v2] ASoC: core: Init pcm runtime work early to avoid warnings
by Curtis Malainey 04 Dec '19

04 Dec '19

There are cases where we fail before we reach soc_new_pcm which would init the workqueue. When we fail we attempt to flush the queue which generates warnings from the workqueue subsystem when we have not inited the queue. Solution is to use a proxy function to get around this issue. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- include/sound/soc.h | 1 + sound/soc/soc-compress.c | 6 ++---- sound/soc/soc-core.c | 10 ++++++++++ sound/soc/soc-pcm.c | 11 ++++------- 4 files changed, 17 insertions(+), 11 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index c28a1ed5e8df9..2628967998264 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1150,6 +1150,7 @@ struct snd_soc_pcm_runtime { unsigned int num_codecs; struct delayed_work delayed_work; + void (*close_delayed_work_func)(struct snd_soc_pcm_runtime *rtd); #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dpcm_root; #endif diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 61f230324164d..6615ef64c7f55 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -214,10 +214,8 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) * This is to ensure there are no pops or clicks in between any music tracks * due to DAPM power cycling. */ -static void close_delayed_work(struct work_struct *work) +static void close_delayed_work(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = - container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); struct snd_soc_dai *codec_dai = rtd->codec_dai; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -929,7 +927,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* DAPM dai link stream work */ - INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + rtd->close_delayed_work_func = close_delayed_work; rtd->compr = compr; compr->private_data = rtd; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 062653ab03a37..0e2e628302f1d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -435,6 +435,15 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd) device_unregister(rtd->dev); } +static void close_delayed_work(struct work_struct *work) { + struct snd_soc_pcm_runtime *rtd = + container_of(work, struct snd_soc_pcm_runtime, + delayed_work.work); + + if (rtd->close_delayed_work_func) + rtd->close_delayed_work_func(rtd); +} + static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -470,6 +479,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( rtd->dev = dev; dev_set_drvdata(dev, rtd); + INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); /* * for rtd->codec_dais diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 01eb8700c3de5..b78f6ff2b1d3f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -637,10 +637,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) * This is to ensure there are no pops or clicks in between any music tracks * due to DAPM power cycling. */ -static void close_delayed_work(struct work_struct *work) +static void close_delayed_work(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = - container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); struct snd_soc_dai *codec_dai = rtd->codec_dais[0]; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -660,7 +658,7 @@ static void close_delayed_work(struct work_struct *work) mutex_unlock(&rtd->card->pcm_mutex); } -static void codec2codec_close_delayed_work(struct work_struct *work) +static void codec2codec_close_delayed_work(struct snd_soc_pcm_runtime *rtd) { /* * Currently nothing to do for c2c links @@ -2974,10 +2972,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) /* DAPM dai link stream work */ if (rtd->dai_link->params) - INIT_DELAYED_WORK(&rtd->delayed_work, - codec2codec_close_delayed_work); + rtd->close_delayed_work_func = codec2codec_close_delayed_work; else - INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + rtd->close_delayed_work_func = close_delayed_work; pcm->nonatomic = rtd->dai_link->nonatomic; rtd->pcm = pcm; -- 2.24.0.393.g34dc348eaf-goog

3 2

[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt5640: Update quirk for Teclast X89
by Hans de Goede 04 Dec '19

04 Dec '19

When the Teclast X89 quirk was added we did not have jack-detection support yet. Note the over-current detection limit is set to 2mA instead of the usual 1.5mA because this tablet tends to give false-positive button-presses when it is set to 1.5mA. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index dd2b5ad08659..243f683bc02a 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -707,13 +707,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MCLK_EN), }, { + /* Teclast X89 */ .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "TECLAST"), DMI_MATCH(DMI_BOARD_NAME, "tPAD"), }, .driver_data = (void *)(BYT_RT5640_IN3_MAP | - BYT_RT5640_MCLK_EN | - BYT_RT5640_SSP0_AIF1), + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_1P0 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), }, { /* Toshiba Satellite Click Mini L9W-B */ .matches = { -- 2.23.0

3 2

[alsa-devel] [PATCH v4 1/2] ASoC: dt-bindings: fsl_asrc: add compatible string for imx8qm & imx8qxp
by Shengjiu Wang 04 Dec '19

04 Dec '19

Add compatible string "fsl,imx8qm-asrc" for imx8qm platform, "fsl,imx8qxp-asrc" for imx8qxp platform. There are two asrc modules in imx8qm & imx8qxp, the clock mapping is different for each other, so add new property "fsl,asrc-clk-map" to distinguish them. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 -none changes in v3 -use only one compatible string "fsl,imx8qm-asrc", -add new property "fsl,asrc-clk-map". changes in v4 -add "fsl,imx8qxp-asrc" Documentation/devicetree/bindings/sound/fsl,asrc.txt | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt index 1d4d9f938689..f4481779e8f2 100644 --- a/Documentation/devicetree/bindings/sound/fsl,asrc.txt +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -8,7 +8,8 @@ three substreams within totally 10 channels. Required properties: - - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + - compatible : Contains "fsl,imx35-asrc", "fsl,imx53-asrc", + "fsl,imx8qm-asrc", "fsl,imx8qxp-asrc" - reg : Offset and length of the register set for the device. @@ -35,6 +36,11 @@ Required properties: - fsl,asrc-width : Defines a mutual sample width used by DPCM Back Ends. + - fsl,asrc-clk-map : Defines clock map used in driver. which is required + by imx8qm/imx8qxp platform + <0> - select the map for asrc0 in imx8qm/imx8qxp + <1> - select the map for asrc1 in imx8qm/imx8qxp + Optional properties: - big-endian : If this property is absent, the little endian mode -- 2.21.0

3 3

[alsa-devel] utils/alsa.m4: fix atopology detection without alsatest
by GitHub pull_request - opened 04 Dec '19

04 Dec '19

alsa-project/alsa-lib pull request #16 was opened from ffontaine: Since commit 75d393a563efb578c79364a277087c6326267f52, alsa-utils fails to build with --disable-alsatest because atopology is not detected Fixes: - http://autobuild.buildroot.org/results/d0fb760669b02b813115af04adcf24530d35… Signed-off-by: Fabrice Fontaine <fontaine.fabrice(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/16 Patch URL : https://github.com/alsa-project/alsa-lib/pull/16.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH] ASoC: snd-sof-pci - add model parameter and pass it to HDA codec driver
by Jaroslav Kysela 03 Dec '19

03 Dec '19

It may be useful to pass the specific model to the generic HDA codec routines like the legacy HDA driver (snd-hda-intel) allows. The model name "sofbus" is tricky anyway. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- include/sound/sof.h | 1 + sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/sof-pci-dev.c | 5 +++++ 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/sof.h b/include/sound/sof.h index 479101736ee0..a62686baa95d 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -25,6 +25,7 @@ struct snd_sof_pdata { const char *drv_name; const char *name; const char *platform; + const char *modelname; struct device *dev; diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 91bd88fddac7..ccb640bacc99 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -271,7 +271,7 @@ static int hda_init(struct snd_sof_dev *sdev) mutex_init(&hbus->prepare_mutex); hbus->pci = pci; hbus->mixer_assigned = -1; - hbus->modelname = "sofbus"; + hbus->modelname = sdev->pdata->modelname; /* initialise hdac bus */ bus->addr = pci_resource_start(pci, 0); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bbeffd932de7..c01ad85aad2a 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -34,6 +34,10 @@ static int sof_pci_debug; module_param_named(sof_pci_debug, sof_pci_debug, int, 0444); MODULE_PARM_DESC(sof_pci_debug, "SOF PCI debug options (0x0 all off)"); +static char *model; +module_param(model, charp, 0444); +MODULE_PARM_DESC(model, "Use the given HDA board model."); + #define SOF_PCI_DISABLE_PM_RUNTIME BIT(0) #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) @@ -326,6 +330,7 @@ static int sof_pci_probe(struct pci_dev *pci, sof_pdata->desc = (struct sof_dev_desc *)pci_id->driver_data; sof_pdata->dev = dev; sof_pdata->platform = dev_name(dev); + sof_pdata->modelname = model; /* alternate fw and tplg filenames ? */ if (fw_path) -- 2.20.1

3 4

[alsa-devel] Integrating Texes Instruments PCM1860 ADC
by Umar Dockrat 03 Dec '19

03 Dec '19

Good Day, I have been struggling with the integration of a PCM1860 ADC for audio recording. I have created a "sound card" board that contains both a PCM5102A (DAC) and PCM1860 (ADC). The main processor is an Omega Onion 2 which is based on MediaTek MT7688 SoC device. I am struggling with creating the device tree as the Kernel currently has reference to the PCM186X where X is 2,3,4 or 5 but not the more simple PCM1860. The difference between the PCM1860 and the PCM186X is that the PCM1860 does not have I2C control and is HW configured using pull up/down. https://www.kernel.org/doc/Documentation/devicetree/bindings/sound/pcm186x.… I have tried multiple device tree setup's and have only got the DAC working when configured in isolation, as soon as I add any reference to the ADC then both DAC and ADC are not recognised. This DTSI allows the DAC to function https://pastebin.com/z9H67DHd There are my attempts at including the DAC and ADC: https://pastebin.com/RLsKNsiR https://pastebin.com/rmJVcCz1 Please can someone assist with the correct way to integrate this ADC device? Any suggestions on how to properly debug this further is also really appreciated. Thanks, UFD

1 0