July 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Intel: board: Fix missing sentinel for bxt_board_id
by Pradeep Tewani 26 Jul '17

26 Jul '17

Without a sentinel in the array of platform devices, we can get a panic so add a sentinel. Full stack dump below: [ 231.564705] BUG: KASAN: global-out-of-bounds in platform_match+0xb0/0x146 [ 231.565500] Read of size 1 at addr ffffffff8380a1a0 by task swapper/0/1 [ 231.566280] [ 231.566594] CPU: 0 PID: 1 Comm: swapper/0 Not tainted 4.12.0-00289-g152771f #1 [ 231.567526] Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.9.3-20161025_171302-gandalf 04/01/2014 [ 231.568730] Call Trace: [ 231.569128] dump_stack+0xf8/0x14a [ 231.569615] print_address_description+0x57/0x1e4 [ 231.570216] ? driver_probe_device+0x814/0x814 [ 231.570798] kasan_report+0x1cb/0x1eb [ 231.571302] ? platform_match+0xb0/0x146 [ 231.571833] __asan_load1+0x45/0x47 [ 231.572324] platform_match+0xb0/0x146 [ 231.572843] ? platform_uevent+0x73/0x73 [ 231.573370] __driver_attach+0x47/0x16a [ 231.573890] bus_for_each_dev+0x11a/0x15c [ 231.574422] ? bus_remove_file+0x57/0x57 [ 231.574951] ? do_raw_spin_unlock+0xc3/0xe0 [ 231.575501] driver_attach+0x2b/0x2e [ 231.576002] bus_add_driver+0x25e/0x3c5 [ 231.576520] ? set_debug_rodata+0x20/0x20 [ 231.584776] driver_register+0x13e/0x19d [ 231.585305] __platform_driver_register+0x6c/0x71 [ 231.585908] ? skl_driver_init+0x20/0x20 [ 231.586436] broxton_audio_init+0x17/0x19 [ 231.586976] do_one_initcall+0x11a/0x221 [ 231.587499] ? start_kernel+0x525/0x525 [ 231.588020] ? lock_downgrade+0x281/0x2ec [ 231.588557] ? trace_hardirqs_on_caller+0x223/0x267 [ 231.589169] ? _raw_spin_unlock_irqrestore+0x49/0x55 [ 231.589791] ? set_debug_rodata+0x20/0x20 [ 231.590324] kernel_init_freeable+0x1fd/0x2b0 [ 231.590891] ? rest_init+0x17c/0x17c [ 231.591384] kernel_init+0x11/0x157 [ 231.591876] ? rest_init+0x17c/0x17c [ 231.592371] ret_from_fork+0x2a/0x40 [ 231.592865] [ 231.593166] The buggy address belongs to the variable: [ 231.593806] bxt_board_ids+0x40/0x9e0 Fixes: 152771fbc3 ("ASoC: Intel: board: Add Geminilake platform support") Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Pradeep Tewani <pradeep.d.tewani(a)intel.com> --- sound/soc/intel/boards/bxt_rt298.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index a4b5bcb..7843104 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -595,6 +595,7 @@ static const struct platform_device_id bxt_board_ids[] = { (unsigned long)&broxton_rt298 }, { .name = "glk_alc298s_i2s", .driver_data = (unsigned long)&geminilake_rt298 }, + {} }; static struct platform_driver broxton_audio = { -- 2.7.4

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[alsa-devel] [PATCH 1/2] ASoC: rt5670: merge ADC L/R Mux
by Bard Liao 26 Jul '17

26 Jul '17

Some ADC L/R Muxes share the some control bit. So, merge them. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 42 ++++++++++++++++-------------------------- 1 file changed, 16 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0ec7985..2866f9b 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1151,20 +1151,15 @@ static const char * const rt5670_stereo_adc1_src[] = { static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc1_enum, RT5670_STO1_ADC_MIXER, RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); -static const struct snd_kcontrol_new rt5670_sto_adc_l1_mux = - SOC_DAPM_ENUM("Stereo1 ADC L1 source", rt5670_stereo1_adc1_enum); - -static const struct snd_kcontrol_new rt5670_sto_adc_r1_mux = - SOC_DAPM_ENUM("Stereo1 ADC R1 source", rt5670_stereo1_adc1_enum); +static const struct snd_kcontrol_new rt5670_sto_adc_1_mux = + SOC_DAPM_ENUM("Stereo1 ADC 1 Mux", rt5670_stereo1_adc1_enum); static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc1_enum, RT5670_STO2_ADC_MIXER, RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); -static const struct snd_kcontrol_new rt5670_sto2_adc_l1_mux = - SOC_DAPM_ENUM("Stereo2 ADC L1 source", rt5670_stereo2_adc1_enum); +static const struct snd_kcontrol_new rt5670_sto2_adc_1_mux = + SOC_DAPM_ENUM("Stereo2 ADC 1 Mux", rt5670_stereo2_adc1_enum); -static const struct snd_kcontrol_new rt5670_sto2_adc_r1_mux = - SOC_DAPM_ENUM("Stereo2 ADC R1 source", rt5670_stereo2_adc1_enum); /* MX-27 MX-26 [11] */ static const char * const rt5670_stereo_adc2_src[] = { @@ -1174,20 +1169,15 @@ static const char * const rt5670_stereo_adc2_src[] = { static SOC_ENUM_SINGLE_DECL(rt5670_stereo1_adc2_enum, RT5670_STO1_ADC_MIXER, RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); -static const struct snd_kcontrol_new rt5670_sto_adc_l2_mux = - SOC_DAPM_ENUM("Stereo1 ADC L2 source", rt5670_stereo1_adc2_enum); - -static const struct snd_kcontrol_new rt5670_sto_adc_r2_mux = - SOC_DAPM_ENUM("Stereo1 ADC R2 source", rt5670_stereo1_adc2_enum); +static const struct snd_kcontrol_new rt5670_sto_adc_2_mux = + SOC_DAPM_ENUM("Stereo1 ADC 2 Mux", rt5670_stereo1_adc2_enum); static SOC_ENUM_SINGLE_DECL(rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER, RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); -static const struct snd_kcontrol_new rt5670_sto2_adc_l2_mux = - SOC_DAPM_ENUM("Stereo2 ADC L2 source", rt5670_stereo2_adc2_enum); +static const struct snd_kcontrol_new rt5670_sto2_adc_2_mux = + SOC_DAPM_ENUM("Stereo2 ADC 2 Mux", rt5670_stereo2_adc2_enum); -static const struct snd_kcontrol_new rt5670_sto2_adc_r2_mux = - SOC_DAPM_ENUM("Stereo2 ADC R2 source", rt5670_stereo2_adc2_enum); /* MX-27 MX26 [10] */ static const char * const rt5670_stereo_adc_src[] = { @@ -1642,23 +1632,23 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, &rt5670_sto1_dmic_mux), SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto_adc_l2_mux), + &rt5670_sto_adc_2_mux), SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto_adc_r2_mux), + &rt5670_sto_adc_2_mux), SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto_adc_l1_mux), + &rt5670_sto_adc_1_mux), SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto_adc_r1_mux), + &rt5670_sto_adc_1_mux), SND_SOC_DAPM_MUX("Stereo2 DMIC Mux", SND_SOC_NOPM, 0, 0, &rt5670_sto2_dmic_mux), SND_SOC_DAPM_MUX("Stereo2 ADC L2 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto2_adc_l2_mux), + &rt5670_sto2_adc_2_mux), SND_SOC_DAPM_MUX("Stereo2 ADC R2 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto2_adc_r2_mux), + &rt5670_sto2_adc_2_mux), SND_SOC_DAPM_MUX("Stereo2 ADC L1 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto2_adc_l1_mux), + &rt5670_sto2_adc_1_mux), SND_SOC_DAPM_MUX("Stereo2 ADC R1 Mux", SND_SOC_NOPM, 0, 0, - &rt5670_sto2_adc_r1_mux), + &rt5670_sto2_adc_1_mux), SND_SOC_DAPM_MUX("Stereo2 ADC LR Mux", SND_SOC_NOPM, 0, 0, &rt5670_sto2_adc_lr_mux), SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0, -- 2.7.4

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[alsa-devel] [PATCH v4 0/9] add support for Sama5d2 audio PLLs and enable ClassD
by Quentin Schulz 26 Jul '17

26 Jul '17

This patch series adds support for the audio PLLs and enables ClassD that can be found in ATMEL Sama5d2 SoC. There are two audio PLLs (PMC and PAD) that shares the same parent (FRAC). FRAC can output between 620 and 700MHz and only multiply the rate of its parent. The two audio PLLs then divide the FRAC rate to best match the asked rate. I basically took an old patch series posted by Nicolas on December, 6th 2016[1][2][3] and the comments Boris did on the first version[4] Nicolas sent on July, 15th 2015. I also fixed the function used to compute the divisors, removed useless spinlocks and added a range to the audio frac PLL to stay within vendor's supported range. Clocks that are children of gclk (generated-clk) are now able to propagate rate to the audio PLL clocks when needed. However, there are multiple children clocks that could technically change the rate of audio_pll (via gck). With the rate locking introduced in Jerome Brunet's patch series[5], the first consumer to enable the clock will be the one definitely setting the rate of the clock. Without the rate locking, the last consumer to set the rate will be able to mess with the rate. Since audio IPs are most likely to request the same rate, we enforce that the only clks able to modify gck rate are those of audio IPs. To remain consistent, we deny other clocks to be children of audio_pll. v4: - initialized the clk_init_data structure, - switched to of_clk_parent_fill instead of manually setting parent_names, - removed include of clkdev.h in favor of slab.h, - removed unnecessary second cast in operation, - used clamp instead of two conditions, - removed dependency on clock-output-names, the name is gotten from DT node, - merged all three drivers for audio PLLs (FRAC, PMC, PAD) into one, v3: - added a flag per generated clock to know which one is allowed to modify audio_pll rate, - set the flag in setup function depending on the compatible and the clock ID, v2: - split big patch containing dt-binding, pll drivers and classd modifications, - removed unused AUDIO_PLL_*FOUT* defines from clk-audio-pll-pmc, - added conditions for audio pll rate setting restriction for SSC and I2S, - renamed all mentions of ACLK to GCLK in classD driver, Thanks, Quentin [1] https://patchwork.kernel.org/patch/9462351/ [2] https://patchwork.kernel.org/patch/9462347/ [3] https://patchwork.kernel.org/patch/9462349/ [4] https://www.spinics.net/lists/arm-kernel/msg436120.html [5] http://www.spinics.net/lists/linux-clk/msg17927.html Cyrille Pitchen (2): ARM: dts: at91: sama5d2: add classd nodes ARM: dts: at91: sama5d2_xplained: add pin muxing and enable classd Quentin Schulz (7): clk: at91: clk-generated: remove useless divisor loop dt-bindings: clk: at91: add audio plls to the compatible list clk: at91: add audio pll clock drivers clk: at91: clk-generated: create function to find best_diff clk: at91: clk-generated: make gclk determine audio_pll rate ASoC: atmel-classd: remove aclk clock from DT binding ASoC: atmel-classd: remove aclk clock Documentation/devicetree/bindings/clock/at91-clock.txt | 10 +- Documentation/devicetree/bindings/sound/atmel-classd.txt | 9 +- arch/arm/boot/dts/at91-sama5d2_xplained.dts | 16 +- arch/arm/boot/dts/sama5d2.dtsi | 39 +- arch/arm/mach-at91/Kconfig | 4 +- drivers/clk/at91/Makefile | 1 +- drivers/clk/at91/clk-audio-pll.c | 531 ++++++++- drivers/clk/at91/clk-generated.c | 101 +- include/linux/clk/at91_pmc.h | 25 +- sound/soc/atmel/atmel-classd.c | 47 +- 10 files changed, 724 insertions(+), 59 deletions(-) create mode 100644 drivers/clk/at91/clk-audio-pll.c base-commit: 047a0f692a9354d730dad30f25e1ebd8437b32b1 -- git-series 0.9.1

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[alsa-devel] [PATCH v3 0/9] add support for Sama5d2 audio PLLs and enable ClassD
by Quentin Schulz 26 Jul '17

26 Jul '17

This patch series adds support for the audio PLLs and enables ClassD that can be found in ATMEL Sama5d2 SoC. There are two audio PLLs (PMC and PAD) that shares the same parent (FRAC). FRAC can output between 620 and 700MHz and only multiply the rate of its parent. The two audio PLLs then divide the FRAC rate to best match the asked rate. I basically took an old patch series posted by Nicolas on December, 6th 2016[1][2][3] and the comments Boris did on the first version[4] Nicolas sent on July, 15th 2015. I also fixed the function used to compute the divisors, removed useless spinlocks and added a range to the audio frac PLL to stay within vendor's supported range. Clocks that are children of gclk (generated-clk) are now able to propagate rate to the audio PLL clocks when needed. However, there are multiple children clocks that could technically change the rate of audio_pll (via gck). With the rate locking introduced in Jerome Brunet's patch series[5], the first consumer to enable the clock will be the one definitely setting the rate of the clock. Without the rate locking, the last consumer to set the rate will be able to mess with the rate. Since audio IPs are most likely to request the same rate, we enforce that the only clks able to modify gck rate are those of audio IPs. To remain consistent, we deny other clocks to be children of audio_pll. Thanks, Quentin [1] https://patchwork.kernel.org/patch/9462351/ [2] https://patchwork.kernel.org/patch/9462347/ [3] https://patchwork.kernel.org/patch/9462349/ [4] https://www.spinics.net/lists/arm-kernel/msg436120.html [5] http://www.spinics.net/lists/linux-clk/msg17927.html Cyrille Pitchen (2): ARM: dts: at91: sama5d2: add classd nodes ARM: dts: at91: sama5d2_xplained: add pin muxing and enable classd Quentin Schulz (7): clk: at91: clk-generated: remove useless divisor loop clk: at91: add audio plls to the compatible list in DT binding clk: at91: add audio pll clock drivers clk: at91: clk-generated: create function to find best_diff clk: at91: clk-generated: make gclk determine audio_pll rate ASoC: atmel-classd: remove aclk clock from DT binding ASoC: atmel-classd: remove aclk clock .../devicetree/bindings/clock/at91-clock.txt | 10 + .../devicetree/bindings/sound/atmel-classd.txt | 9 +- arch/arm/boot/dts/at91-sama5d2_xplained.dts | 16 ++ arch/arm/boot/dts/sama5d2.dtsi | 39 +++- arch/arm/mach-at91/Kconfig | 4 + drivers/clk/at91/Makefile | 2 + drivers/clk/at91/clk-audio-pll-pad.c | 206 ++++++++++++++++++ drivers/clk/at91/clk-audio-pll-pmc.c | 174 +++++++++++++++ drivers/clk/at91/clk-audio-pll.c | 239 +++++++++++++++++++++ drivers/clk/at91/clk-generated.c | 101 +++++++-- include/linux/clk/at91_pmc.h | 25 +++ sound/soc/atmel/atmel-classd.c | 47 ++-- 12 files changed, 813 insertions(+), 59 deletions(-) create mode 100644 drivers/clk/at91/clk-audio-pll-pad.c create mode 100644 drivers/clk/at91/clk-audio-pll-pmc.c create mode 100644 drivers/clk/at91/clk-audio-pll.c -- 2.11.0

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[alsa-devel] [PATCH] ALSA: hda - Add mute led support for HP ProBook 440 G4
by Kai-Heng Feng 25 Jul '17

25 Jul '17

Mic mute led does not work on HP ProBook 440 G4. We can use CXT_FIXUP_MUTE_LED_GPIO fixup to support it. BugLink: https://bugs.launchpad.net/bugs/1705586 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e8253737c47a..615a69a32970 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -931,6 +931,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), -- 2.13.3

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[alsa-devel] [PATCH v2 0/2] ASoC: Add I2S support for Allwinner H3 SoCs
by codekipper＠gmail.com 25 Jul '17

25 Jul '17

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, please find attached a series of patches to bring i2s support to the Allwinner H3 SoC. This has been tested with the following setups: A20 Olimex EVB connected to a pcm5102 Orange Pi 2 connected to a uda1380 Orange Pi 2 hdmi audio playback Pine 64 connected to the audio DAC board To get i2s working some additional patches are required which will be delivered later. For now they have been pushed here https://github.com/codekipper/linux-sunxi/commits/sunxi-audio-h3 I don't own a A33 device which uses the i2s block for the audio codec so if someone could test against that it would be much appreciated. I'm also wondering if there is a preferred way of setting the lrclk size in the dts?..currently it is set to the sample width but for example the pcm5102a wants it to be 32 bits whatever the sample rate. Thanks in advance, CK v2 changes compared to v1 are: - massive refactoring to remove duplicate code making use of regmap_fields. - extending the clock divisors. - removed code that should be delivered when we support 20/24bits --- Marcus Cooper (2): ASoC: sun4i-i2s: Add more quirks for newer SoCs ASoC: sun4i-i2s: Add support for H3 .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + sound/soc/sunxi/sun4i-i2s.c | 469 ++++++++++++++++++--- 2 files changed, 407 insertions(+), 64 deletions(-) -- 2.13.3

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[alsa-devel] [ANNOUNCE] Audio Mini Summit 2017 at Prague, Oct 27
by Takashi Iwai 25 Jul '17

25 Jul '17

Hi, this is an announcement of Audio Mini Summit 2017. It's an annual meeting we regularly have along with major Linux conferences, and this year, we'll hold the meeting at Prague (Czech Republic) on Friday, October 27, right after GStreamer Summit, ELCE and Open Source Summit Europe. We planned a whole-day meeting, and will likely have an evening dinner event, too. Please note that the meeting will be held in a conference room of SUSE Prague office, not in Hilton hotel where ELCE/OSSE is held. It's located near Hilton, just a couple of blocks away. The address of SUSE office is: Corso II, Křižíkova 148/34, 18600 Praha 8 https://goo.gl/maps/anET1yh3XXu Also, since the conference room is hosted by SUSE, it's an open event at this time; that is, you can attend only this meeting without registration for other Linux Foundation events in that week, too. If you are interested in the event, please fill up the registration form in Google doc spreadsheet below: https://docs.google.com/spreadsheets/d/1a-RaYtTVVtgh_cyRN5GwATiixppGCMEwqw6… Just add yourself in the attendee list from the web page. And, now we are gathering the topics to discuss, too. If you have any interesting topics to be discussed in the meeting, feel free to post to alsa-devel ML. I'm going to write up later. thanks, Takashi

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Re: [alsa-devel] [PATCH] ASoC: rt5514: Use the local variable to get the data from SPI device
by Hsin-yu Chao 25 Jul '17

25 Jul '17

The original cause seems to be in spi driver. I tested setting enable_dma flag to false in spi-pxa2xx driver and that fixes the problem. Will investigate in spi driver side. Thanks! Hsin-yu On Mon, Jul 24, 2017 at 11:06 AM, Oder Chiou <oder_chiou(a)realtek.com> wrote: > > On Fri, Jul 21, 2017 at 04:00:47PM +0800, Oder Chiou wrote: > > > The patch uses the local variable to get the data from SPI device to > > > prevent to get all zero data. > > > > How do you end up with all zero data from reading directly into rxbuf? > > > While the "runtime->dma_area" put into the rxbuf directly on the Intel > platform, it seemed that had the limitation to let the memory cannot be > written the data using the driver of the SPI controller. We verified the > original code on the platforms of Nvidia and Rockchip, and these are > working fine. >

1 0

[alsa-devel] [PATCH 0/3] Motorola Droid 4 Audio Support
by Sebastian Reichel 25 Jul '17

25 Jul '17

Hi, I got working sound on Droid 4 with mainline \o/. The codec is currently missing support for detecting if something has been plugged into the 3.5mm connector, since that seems to require some closed source firmware and needs further investigation. I think this can be added later. Tested: - Playing via [HiFi] using Speaker, Earpiece, Headphone - Playing via [Voice] using Speaker, Earpiece, Headphone - Recording via [Voice] using Mic - Volume Controls - Checking that DAPM graph look correct -- Sebastian Sebastian Reichel (3): ASoC: codec: cpcap: new codec ARM: dts: motorola-cpcap-mapphone: add audio-codec ARM: dts: omap4-droid4: add soundcard .../bindings/sound/motorola,cpcap-audio-codec.txt | 19 + arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 6 + arch/arm/boot/dts/omap4-droid4-xt894.dts | 76 ++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cpcap.c | 1422 ++++++++++++++++++++ 6 files changed, 1528 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt create mode 100644 sound/soc/codecs/cpcap.c -- 2.13.2

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[alsa-devel] [PATCH] linux/kernel.h: move DIV_ROUND_DOWN_ULL() macro
by Masahiro Yamada 25 Jul '17

25 Jul '17

This macro is useful to avoid link error on 32-bit systems. We have the same definition in two drivers, so move it to include/linux/kernel.h While we are here, refactor DIV_ROUND_UP_ULL() by using DIV_ROUND_DOWN_ULL(). Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- drivers/mtd/nand/denali.c | 3 --- include/linux/kernel.h | 7 +++++-- sound/soc/codecs/pcm512x.c | 3 --- 3 files changed, 5 insertions(+), 8 deletions(-) diff --git a/drivers/mtd/nand/denali.c b/drivers/mtd/nand/denali.c index d723be352148..3087b0ba7b7f 100644 --- a/drivers/mtd/nand/denali.c +++ b/drivers/mtd/nand/denali.c @@ -980,9 +980,6 @@ static int denali_erase(struct mtd_info *mtd, int page) return irq_status & INTR__ERASE_COMP ? 0 : NAND_STATUS_FAIL; } -#define DIV_ROUND_DOWN_ULL(ll, d) \ - ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; }) - static int denali_setup_data_interface(struct mtd_info *mtd, int chipnr, const struct nand_data_interface *conf) { diff --git a/include/linux/kernel.h b/include/linux/kernel.h index 96381f2b0e09..2a254ae6f7f7 100644 --- a/include/linux/kernel.h +++ b/include/linux/kernel.h @@ -78,8 +78,11 @@ #define FIELD_SIZEOF(t, f) (sizeof(((t*)0)->f)) #define DIV_ROUND_UP __KERNEL_DIV_ROUND_UP -#define DIV_ROUND_UP_ULL(ll,d) \ - ({ unsigned long long _tmp = (ll)+(d)-1; do_div(_tmp, d); _tmp; }) + +#define DIV_ROUND_DOWN_ULL(ll, d) \ + ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; }) + +#define DIV_ROUND_UP_ULL(ll, d) DIV_ROUND_DOWN_ULL((ll) + (d) - 1, (d)) #if BITS_PER_LONG == 32 # define DIV_ROUND_UP_SECTOR_T(ll,d) DIV_ROUND_UP_ULL(ll, d) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 72b19e62f626..ad3bfb894ee0 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -30,9 +30,6 @@ #include "pcm512x.h" -#define DIV_ROUND_DOWN_ULL(ll, d) \ - ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; }) - #define PCM512x_NUM_SUPPLIES 3 static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { "AVDD", -- 2.7.4

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