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

[alsa-devel] [PATCH v4] ASoC: intel: Add Broadwell rt5650 machine driver
by Curtis Malainey 18 Dec '19

18 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> --- v2: * Addressed Pierre's comments * Added SOF dependencies * Added platform override * Fixed Kconfig * Moved to devm register v3: * Removed THIS_MODULE * Added PM Ops v4: * Fixed sparse warnings sound/soc/intel/boards/Kconfig | 12 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5650.c | 319 ++++++++++++++++++ .../common/soc-acpi-intel-hsw-bdw-match.c | 7 + 4 files changed, 340 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..de613ae5052bf 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -31,6 +31,18 @@ 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 I2C + depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST + depends on X86_INTEL_LPSS || COMPILE_TEST + 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..ba3fc1ef900ab --- /dev/null +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -0,0 +1,319 @@ +// 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/delay.h> +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.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_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), + 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, +}; + +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) +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; +} +#endif + +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, +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) + .init = bdw_rt5650_rtd_init, +#endif + .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; + struct snd_soc_acpi_mach *mach; + int ret; + + 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; + + /* override plaform name, if required */ + mach = (&pdev->dev)->platform_data; + ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5650_card, + mach->mach_params.platform); + + if (ret) + return ret; + + snd_soc_card_set_drvdata(&bdw_rt5650_card, bdw_rt5650); + + return devm_snd_soc_register_card(&pdev->dev, &bdw_rt5650_card); +} + +static struct platform_driver bdw_rt5650_audio = { + .probe = bdw_rt5650_probe, + .driver = { + .name = "bdw-rt5650", + .pm = &snd_soc_pm_ops, + }, +}; + +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..35958553652ec 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 +29,13 @@ 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", + .sof_fw_filename = "sof-bdw.ri", + .sof_tplg_filename = "sof-bdw-rt5650.tplg", + }, { .id = "RT5677CE", .drv_name = "bdw-rt5677", -- 2.24.1.735.g03f4e72817-goog

3 2

[alsa-devel] [PATCH] ALSA: ice1724: Fix sleep-in-atomic in Infrasonic Quartet support code
by Takashi Iwai 18 Dec '19

18 Dec '19

Jia-Ju Bai reported a possible sleep-in-atomic scenario in the ice1724 driver with Infrasonic Quartet support code: namely, ice->set_rate callback gets called inside ice->reg_lock spinlock, while the callback in quartet.c holds ice->gpio_mutex. This patch fixes the invalid call: it simply moves the calls of ice->set_rate and ice->set_mclk callbacks outside the spinlock. Reported-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/5d43135e-73b9-a46a-2155-9e91d0dcdf83@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/ice1712/ice1724.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index c80a16ee6e76..242542e23d28 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -647,6 +647,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, unsigned long flags; unsigned char mclk_change; unsigned int i, old_rate; + bool call_set_rate = false; if (rate > ice->hw_rates->list[ice->hw_rates->count - 1]) return -EINVAL; @@ -670,7 +671,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, * setting clock rate for internal clock mode */ old_rate = ice->get_rate(ice); if (force || (old_rate != rate)) - ice->set_rate(ice, rate); + call_set_rate = true; else if (rate == ice->cur_rate) { spin_unlock_irqrestore(&ice->reg_lock, flags); return 0; @@ -678,12 +679,14 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, } ice->cur_rate = rate; + spin_unlock_irqrestore(&ice->reg_lock, flags); + + if (call_set_rate) + ice->set_rate(ice, rate); /* setting master clock */ mclk_change = ice->set_mclk(ice, rate); - spin_unlock_irqrestore(&ice->reg_lock, flags); - if (mclk_change && ice->gpio.i2s_mclk_changed) ice->gpio.i2s_mclk_changed(ice); if (ice->gpio.set_pro_rate) -- 2.16.4

1 0

[alsa-devel] [BUG] ALSA: ice1712: a possible sleep-in-atomic-context bug in snd_vt1724_set_pro_rate()
by Jia-Ju Bai 18 Dec '19

18 Dec '19

The driver may sleep while holding a read lock. The function call path (from bottom to top) in Linux 4.19 is: sound/pci/ice1712/quartet.c, 414: mutex_lock in reg_write sound/pci/ice1712/quartet.c, 485: reg_write in set_cpld sound/pci/ice1712/quartet.c, 876: set_cpld in qtet_set_rate sound/pci/ice1712/ice1724.c, 687: (FUNC_PTR) qtet_set_rate in snd_vt1724_set_pro_rate sound/pci/ice1712/ice1724.c, 668: _raw_spin_lock_irqsave in snd_vt1724_set_pro_rate (FUNC_PTR) means a function pointer is called. mutex_lock() can sleep at runtime. I am not sure how to properly fix this possible bug, so I only report it. This bug is found by a static analysis tool STCheck written by myself. Best wishes, Jia-Ju Bai

2 1

[alsa-devel] Playback issue while recording simultaneously
by Mohan Prasath Ramamoorthy 18 Dec '19

18 Dec '19

Dear All, We are working on external modules of microphone and speaker and we are using the kernel version 3.10.96 We are recording with commands like arecord and aplay for recording and playback respectively. While doing the below experiments we could able to observe an issue as below, 1. With only recording and not playback we haven't faced any issues. 2. With only playback and not recording we haven't faced any issues. 3. By starting an infinite recording in the background and by doing a playback of the audio file(30seconds, the sampling rate of 22050, the channel is stereo), after some iterations, we are facing a below error and we could not able to listen to the audio. Error: aplay: pcm_write:1940: write error: Input/output error Below are the respective commands for recording and playback, 1. arecord -D hw:0,0 -c 2 -f s16_le -r 8000 /home/ubuntu/mic_testing.wav & 2. aplay -D hw:0,3 audio_22k_c2.wav But when the issues come, we just stopped recording and restarted the recording in the background. After this, the playback is working fine. Also, during the issue came, we have checked for the below properties, 1. I2S has been enabled. Checked under the file: /sys/kernel/debug/clock/clock_tree 2. Driver callbacks have been called. (Checked with the kernel logs[dmesg]) We would like to know the reason behind this error.? -- With Best Regards, Mohan Prasath VVDN Technologies Pvt Ltd Cell: +91 9629768919 Web: www.vvdntech.com -- _Disclaimer: _© 2019 VVDN Technologies Pvt. Ltd. This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful._ _ _ _ __

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[alsa-devel] [INTERNAL REVIEW] asoc: sti: fix possible sleep-in-atomic
by Arnaud Pouliquen 18 Dec '19

18 Dec '19

Change mutex and spinlock management to avoid sleep in atomic issue. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- sound/soc/sti/uniperif_player.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 48ea915b24ba..2ed92c990b97 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -226,7 +226,6 @@ static void uni_player_set_channel_status(struct uniperif *player, * sampling frequency. If no sample rate is already specified, then * set one. */ - mutex_lock(&player->ctrl_lock); if (runtime) { switch (runtime->rate) { case 22050: @@ -303,7 +302,6 @@ static void uni_player_set_channel_status(struct uniperif *player, player->stream_settings.iec958.status[3 + (n * 4)] << 24; SET_UNIPERIF_CHANNEL_STA_REGN(player, n, status); } - mutex_unlock(&player->ctrl_lock); /* Update the channel status */ if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) @@ -365,8 +363,10 @@ static int uni_player_prepare_iec958(struct uniperif *player, SET_UNIPERIF_CTRL_ZERO_STUFF_HW(player); + mutex_lock(&player->ctrl_lock); /* Update the channel status */ uni_player_set_channel_status(player, runtime); + mutex_unlock(&player->ctrl_lock); /* Clear the user validity user bits */ SET_UNIPERIF_USER_VALIDITY_VALIDITY_LR(player, 0); @@ -598,7 +598,6 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, iec958->status[1] = ucontrol->value.iec958.status[1]; iec958->status[2] = ucontrol->value.iec958.status[2]; iec958->status[3] = ucontrol->value.iec958.status[3]; - mutex_unlock(&player->ctrl_lock); spin_lock_irqsave(&player->irq_lock, flags); if (player->substream && player->substream->runtime) @@ -608,6 +607,8 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol, uni_player_set_channel_status(player, NULL); spin_unlock_irqrestore(&player->irq_lock, flags); + mutex_unlock(&player->ctrl_lock); + return 0; } -- 2.17.1

1 0

[alsa-devel] [BUG] ALSA: soc: sti: a possible sleep-in-atomic-context bug in uni_player_ctl_iec958_put()
by Jia-Ju Bai 18 Dec '19

18 Dec '19

The driver may sleep while holding a spinlock. The function call path (from bottom to top) in Linux 4.19 is: sound/soc/sti/uniperif_player.c, 229: mutex_lock in uni_player_set_channel_status sound/soc/sti/uniperif_player.c, 608: uni_player_set_channel_status in uni_player_ctl_iec958_put sound/soc/sti/uniperif_player.c, 603: _raw_spin_lock_irqsave in uni_player_ctl_iec958_put mutex_lock() can sleep at runtime. I am not sure how to properly fix this possible bug, so I only report it. This bug is found by a static analysis tool STCheck written by myself. Best wishes, Jia-Ju Bai

2 1

[alsa-devel] [PATCH 0/6] ASoC: soc-core cleanup step8
by Kuninori Morimoto 18 Dec '19

18 Dec '19

Hi Mark These are soc-core cleanup step8. There is no relationship between each patches. Kuninori Morimoto (6): ASoC: soc-core: remove snd_soc_rtdcom_list ASoC: soc-core: care .ignore_suspend for Component suspend ASoC: soc-core: remove duplicate pinctrl operation when suspend ASoC: soc-core: do pinctrl_pm_select_xxx() as component ASoC: soc-core: add snd_soc_close_delayed_work() ASoC: soc-core: add snd_soc_stream_stop() include/sound/soc.h | 20 +++--- sound/soc/soc-component.c | 33 +++++---- sound/soc/soc-compress.c | 110 ++++++++-------------------- sound/soc/soc-core.c | 180 +++++++++++++++++++++++++--------------------- sound/soc/soc-pcm.c | 113 +++++++---------------------- 5 files changed, 179 insertions(+), 277 deletions(-) -- 2.7.4

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[alsa-devel] Undecalared Variable "p" when compile alsa-lib with TPLG_DEBUG defined
by GitHub issues - opened 18 Dec '19

18 Dec '19

alsa-project/alsa-lib issue #18 was opened from aiChaoSONG: Undecalared Variable "p" when compile alsa-lib with TPLG_DEBUG defined, the pointer 'p' in line 152 of src/topology/data.c seem not declared. see code below. ``` static void dump_priv_data(struct tplg_elem *elem) { struct snd_soc_tplg_private *priv = elem->data; unsigned int i, j = 0; tplg_dbg(" elem size = %d, priv data size = %d\n", elem->size, priv->size); for (i = 0; i < priv->size; i++) { if (j++ % 8 == 0) tplg_dbg("\n"); *******tplg_dbg(" 0x%x", *p++);******* } tplg_dbg("\n\n"); } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/18 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] TI McASP: Independent RX & TX
by Sven Schwermer 18 Dec '19

18 Dec '19

Hi, I’m looking at hooking up two codecs to a single AM3358’s McASP instance. One codec will be hooked up to the receive section while the other one will be hooked up to the transmit section. BCLK, MCLK and format will not necessarily be shared and may differ. As far as I understand, the McASP allows to operate receive and transmit pretty much independently. Looking at the ASoC driver (davinci-mcasp.c), however, it seems to me that it is not really possible to use receive and transmit independently. Is that correct? Is that particular use case not within the scope of the current driver or is my initial assumption wrong that transmit and receive are basically independent? Any pointers are appreciated. Best regards, Sven

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[alsa-devel] [PATCH] ALSA: usx2y: Adjust indentation in snd_usX2Y_hwdep_dsp_status
by Nathan Chancellor 18 Dec '19

18 Dec '19

Clang warns: ../sound/usb/usx2y/usX2Yhwdep.c:122:3: warning: misleading indentation; statement is not part of the previous 'if' [-Wmisleading-indentation] info->version = USX2Y_DRIVER_VERSION; ^ ../sound/usb/usx2y/usX2Yhwdep.c:120:2: note: previous statement is here if (us428->chip_status & USX2Y_STAT_CHIP_INIT) ^ 1 warning generated. This warning occurs because there is a space before the tab on this line. Remove it so that the indentation is consistent with the Linux kernel coding style and clang no longer warns. This was introduced before the beginning of git history so no fixes tag. Link: https://github.com/ClangBuiltLinux/linux/issues/831 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> --- sound/usb/usx2y/usX2Yhwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index d1caa8ed9e68..9985fc139487 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -119,7 +119,7 @@ static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, info->num_dsps = 2; // 0: Prepad Data, 1: FPGA Code if (us428->chip_status & USX2Y_STAT_CHIP_INIT) info->chip_ready = 1; - info->version = USX2Y_DRIVER_VERSION; + info->version = USX2Y_DRIVER_VERSION; return 0; } -- 2.24.1

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